From clive at lansink.co.nz Mon Apr 1 01:06:42 2013 From: clive at lansink.co.nz (Clive Lansink) Date: Mon, 1 Apr 2013 10:06:42 +1300 Subject: [Freeswitch-users] Recording incoming calls Message-ID: <20130331211539.4DDCBDA021@jlo.kiwilink.co.nz> An embedded and charset-unspecified text was scrubbed... Name: not available Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/331cbdfa/attachment.pl From cal.leeming at simplicitymedialtd.co.uk Mon Apr 1 01:52:03 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 31 Mar 2013 22:52:03 +0100 Subject: [Freeswitch-users] Internal VoIP network + PSTN In-Reply-To: References: Message-ID: Hello, Have you started off looking over the wiki? If not, have a look at this first; http://wiki.freeswitch.org/wiki/Getting_Started_Guide There is a learning curve with FreeSWITCH, as with any technology, and it's not just a case of point and click. You'll need to take some time to learn how FreeSWITCH works and figure out the approach you want to use.. Either that or buy a cudatel :) Cal On Sat, Mar 30, 2013 at 5:40 PM, Ebrahim Bararian < ebrahim.bararian at gmail.com> wrote: > Hello all, > > I'm new to freeswitch and want to do two jobs with it. > > First, I want to make an Inernal IP network(two soft IP phones) connected > to the PSTN network. > > Second I want to use the freeswitch to connect to the PSTN phone line via > the voice modem. > > Can anyone help me with these problems? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130331/0d44d701/attachment.html From jsun at junsun.net Mon Apr 1 04:19:21 2013 From: jsun at junsun.net (Jun Sun) Date: Sun, 31 Mar 2013 17:19:21 -0700 Subject: [Freeswitch-users] bridging two outbound calls In-Reply-To: <1FFF97C269757C458224B7C895F35F15257F06@cantor.std.visionutv.se> References: <515635C3.6040509@junsun.net> , <5156845C.4070302@junsun.net> <374BC379-8DAA-4CEA-AB0D-2DB313E31209@visionutveckling.se> <51570684.2010203@junsun.net>, <5157D80B.6070709@junsun.net> <1FFF97C269757C458224B7C895F35F15257F06@cantor.std.visionutv.se> Message-ID: <5158D289.9090101@junsun.net> Thanks for the pastebin suggestion. I have attached pastebin URL for the output from fs_cli console (debug level 7), from initiating the command to both parties picking up the phone. http://pastebin.com/tZewdvri I did not touch bypass_media in any way. I assume it should be off. (And this is a fairly standard system, other than that it is hosted on AWS EC2 machine) Since I"m using the same SIP trunk, even if bypass_media is set to true, it probably should still work. Thanks in advance. I can get a pcap file if that will be helpful. Jun On 3/31/2013 12:18 AM, Peter Olsson wrote: > Yes, it's there to separate if the second parameter is for en extension (no '&' prefix) or an application (using '&' prefix). Read the full description here: http://wiki.freeswitch.org/wiki/Mod_commands#originate. If you would allow a space in between it would make '&' the second parameter, and the actual string the third - which is not correct. This is just normal syntax rules for how to parse arguments. > > About the audio issue - make sure to get debug logs, and a pcap when making the call, post to pastebin. This is usually a NAT or codec issue. SInce you said it was working if calling a conference, another thing to check might be if you're settings bypass_media or not. > > /Peter > ________________________________________ > Fr?n: Jun Sun [jsun at junsun.net] > Skickat: den 31 mars 2013 08:30 > Till: FreeSWITCH Users Help > Cc: Peter Olsson > ?mne: Re: [Freeswitch-users] bridging two outbound calls > > Is there any specific reason why "&" must be immediately followed by the > application name? I found this restriction pretty annoying (at least for > newbies ;0) > > A simple patch would easily fix this. See one attached. Any takers? > > I'm still searching for the solution for no sound after bridging. Would > appreciate any pointers. (Again, this really has to be one of the > simplest cases ... why it has been so hard?) > > Cheers. > > Jun > > On 3/30/2013 8:36 AM, Jun Sun wrote: >> >> Oh, my god. That is it! After removing the extra space between "&" and >> "bridge", the second number now gets dialed. >> >> However, I cannot hear each other between these two phones. I think the >> signaling part is working, because hanging up one end will cause the >> other end hung up. However, the media is not flowing through. >> >> This must be a simple mistake. Any pointers? >> >> BTW, using conference() app works, i.e., both ends can connect and talk. >> So my system should be in general healthy state. >> >> Cheers. >> >> Jun >> >> On 3/29/2013 11:38 PM, Peter Olsson wrote: >>> It looks like you have a space between & and bridge? It might be my >>> email reader though. Anyway, it must be set like this: &bridge(). >>> >>> Also, I'm not sure about the tel: stuff, if you can set it that way, >>> especially since there is a whitespace in between as well. >>> >>> /Peter >>> >>> 30 mar 2013 kl. 07:27 skrev "Jun Sun" : >>> >>>> >>>> Yes, I tried. I can reverse the positions of those two numbers and >>>> always the first number gets called and the second number gets nothing. >>>> >>>> I start to wonder whether I'm using bridge() application wrongly. Can it >>>> dial out directly to a PSTN number via sofia? >>>> >>>> I was also fumbling with two orignate commands (followed by park() >>>> application) and uuid_bridge to connect. No failures on console, but the >>>> two lines are not talking. >>>> >>>> Thanks. >>>> >>>> Jun >>>> >>>> On 3/29/2013 6:25 PM, Zoltan Medveczky wrote: >>>>> You probably already tried this, but are you able to place a call to >>>>> your party B endpoint at all (i.e. originate(sofia/internal/18005551212 >>>>> @X.X.X.X:5060)? >>>>> >>>>> If that's failing, I'd say there's probably something wrong with your >>>>> dial string. Perhaps your SIP UA is not bound to port 5060? >>>>> >>>>> On Fri, Mar 29, 2013 at 5:45 PM, Jun Sun >>>> > wrote: >>>>> >>>>> I feel really stupid. This has to be one of the simplest cases in >>>>> freeswitch, but I can't seem to get it work. >>>>> >>>>> My goal is to originate two outbound calls and bridge them >>>>> together, a >>>>> typical callback use case. I like to to do it from socket >>>>> api/fs_cli. >>>>> >>>>> Here is what I typed in fs_cli: >>>>> >>>>> originate sofia/internal/15102991912 >>>>> @X.X.X.X:5060 & >>>>> bridge(sofia/internal/18005551212 @X.X.X.X:5060) >>>>> >>>>> The first leg is initiated and answered successfully, but the >>>>> second leg >>>>> never happens. From the console I don't see any action done by >>>>> FS to do >>>>> the bridging part. >>>>> >>>>> Any idea? Thanks in advance. >>>>> >>>>> Cheers. >>>>> >>>>> Jun >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >>> >> >> > > > > !DSPAM:5157d35132761492122807! > From jsun at junsun.net Mon Apr 1 04:25:58 2013 From: jsun at junsun.net (Jun Sun) Date: Sun, 31 Mar 2013 17:25:58 -0700 Subject: [Freeswitch-users] bridging two outbound calls In-Reply-To: References: <515635C3.6040509@junsun.net> <5156845C.4070302@junsun.net> <374BC379-8DAA-4CEA-AB0D-2DB313E31209@visionutveckling.se> <51570684.2010203@junsun.net> <5157D80B.6070709@junsun.net> Message-ID: <5158D416.3050101@junsun.net> I think I understand the logic now. It seems "&" is used as some kind of escape symbol while logically it implies "and consequently ...". I think it is the later logical meaning which lures newbies into using spaces. A simple patch as I attached offers the syntactical sugar without complicating the matter. The patch simply moves all spaces following the "&" to before it. Unless "&" can mean something else in arg list, the patch should be safe and practical. Anyway, I don't think I will forget about the "&" space issue anymore. :) Cheers. Jun On 3/31/2013 10:40 AM, Steven Ayre wrote: > If your question is why can't FS allow a space, its a) specifically a prefix when evaluating that argument and b) there can be further arguments to the originate command after that one, additional spaces would make it problematic figuring out why arg is which. > > Steve > > > > On 31 Mar 2013, at 07:30, Jun Sun wrote: > >> >> Is there any specific reason why "&" must be immediately followed by the application name? I found this restriction pretty annoying (at least for newbies ;0) >> >> A simple patch would easily fix this. See one attached. Any takers? >> >> I'm still searching for the solution for no sound after bridging. Would appreciate any pointers. (Again, this really has to be one of the simplest cases ... why it has been so hard?) >> >> Cheers. >> >> Jun >> >> On 3/30/2013 8:36 AM, Jun Sun wrote: >>> >>> Oh, my god. That is it! After removing the extra space between "&" and >>> "bridge", the second number now gets dialed. >>> >>> However, I cannot hear each other between these two phones. I think the >>> signaling part is working, because hanging up one end will cause the >>> other end hung up. However, the media is not flowing through. >>> >>> This must be a simple mistake. Any pointers? >>> >>> BTW, using conference() app works, i.e., both ends can connect and talk. >>> So my system should be in general healthy state. >>> >>> Cheers. >>> >>> Jun >>> >>> On 3/29/2013 11:38 PM, Peter Olsson wrote: >>>> It looks like you have a space between & and bridge? It might be my >>>> email reader though. Anyway, it must be set like this: &bridge(). >>>> >>>> Also, I'm not sure about the tel: stuff, if you can set it that way, >>>> especially since there is a whitespace in between as well. >>>> >>>> /Peter >>>> >>>> 30 mar 2013 kl. 07:27 skrev "Jun Sun" : >>>> >>>>> >>>>> Yes, I tried. I can reverse the positions of those two numbers and >>>>> always the first number gets called and the second number gets nothing. >>>>> >>>>> I start to wonder whether I'm using bridge() application wrongly. Can it >>>>> dial out directly to a PSTN number via sofia? >>>>> >>>>> I was also fumbling with two orignate commands (followed by park() >>>>> application) and uuid_bridge to connect. No failures on console, but the >>>>> two lines are not talking. >>>>> >>>>> Thanks. >>>>> >>>>> Jun >>>>> >>>>> On 3/29/2013 6:25 PM, Zoltan Medveczky wrote: >>>>>> You probably already tried this, but are you able to place a call to >>>>>> your party B endpoint at all (i.e. originate(sofia/internal/18005551212 >>>>>> @X.X.X.X:5060)? >>>>>> >>>>>> If that's failing, I'd say there's probably something wrong with your >>>>>> dial string. Perhaps your SIP UA is not bound to port 5060? >>>>>> >>>>>> On Fri, Mar 29, 2013 at 5:45 PM, Jun Sun >>>>> > wrote: >>>>>> >>>>>> I feel really stupid. This has to be one of the simplest cases in >>>>>> freeswitch, but I can't seem to get it work. >>>>>> >>>>>> My goal is to originate two outbound calls and bridge them >>>>>> together, a >>>>>> typical callback use case. I like to to do it from socket >>>>>> api/fs_cli. >>>>>> >>>>>> Here is what I typed in fs_cli: >>>>>> >>>>>> originate sofia/internal/15102991912 >>>>>> @X.X.X.X:5060 & >>>>>> bridge(sofia/internal/18005551212 @X.X.X.X:5060) >>>>>> >>>>>> The first leg is initiated and answered successfully, but the >>>>>> second leg >>>>>> never happens. From the console I don't see any action done by >>>>>> FS to do >>>>>> the bridging part. >>>>>> >>>>>> Any idea? Thanks in advance. >>>>>> >>>>>> Cheers. >>>>>> >>>>>> Jun >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 >>>>> >>>>> !DSPAM:5156810f32761697518053! >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 vbvbrj at gmail.com Mon Apr 1 13:19:49 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 01 Apr 2013 12:19:49 +0300 Subject: [Freeswitch-users] FusionPBX or other on working FS Message-ID: <51595135.1010904@gmail.com> Hello. Recently I checked FusionPBX to see what it may help in administer FS. First I downloaded the web files and pushed to web server. Running for configuration ended to nothing. It created its tables in DB but the web page does not work. Then I deployed FusionPBX from precompiled iso. It seems intuitive for basic setups. Now I read that when setting up FusionPBX it will break any configuration FS had. So I am worried if I will be able to restore the configs from web GUI of FusionPBX to what it was, as I understood it is no good to change config files of FS directly when FusionPBX is on them. Is it true? Can't I combine manual and via GUI configuring of FS? May be other GUI will allow me more freedom? -- Mimiko desu. From ibrahimghaznavi at gmail.com Mon Apr 1 13:40:16 2013 From: ibrahimghaznavi at gmail.com (Syed Ibrahim Ghaznavi) Date: Mon, 1 Apr 2013 14:40:16 +0500 Subject: [Freeswitch-users] Problem configuring OpenBTS2.8 with Freeswitch 1.0.6 In-Reply-To: References: Message-ID: i felt neglected, can anyone still help me with the below mentioned problem. Gratitude, Ibrahim On Fri, Mar 29, 2013 at 3:49 PM, Syed Ibrahim Ghaznavi < ibrahimghaznavi at gmail.com> wrote: > Thanks steve for the prompt help ! > > I am pasting a line below from the log, i guess the context is public ? : > > mod_dialplan_xml.c:557 Processing IMSI410071190004419 > ->2222 in *context public > > * > You are right that i have no extension 2222. But i am confused as to how > should i make an extension using IMSI. Like the sample extensions > (1000-1019) looks like: > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > Whereas i found one example of the extension using IMSI: > > > > > > > > > > > > > > > > > > > > > > > Can anyone validate if the above syntax is perfect for adding an extension > using the above syntax? > > Another confusion related to the workflow: > Assuming the syntax is correct, should i follow the following steps to > establish a call between the 2 users: > > - Add the extensions of the users in: > /usr/local/freeswitch/conf/directory/default > - Then reference that entry in the dialplan > - public.xml or > - default.xml ? > - In the new extension added in pubic or default, i should bridge the > call by adding the following extension right? > > > > - > > > > > > > > > > > > > After this i should be able to make calls right? > > Steve- i just check my version of freeswitch it is : FreeSWITCH > Version 1.3.13. Thanks for the suggestion though. > > Any help will be greatly appreciated !!! Thanks much ! > > > > > Gratitude, > > Ibrahim > > > > > On Thu, Mar 28, 2013 at 8:11 PM, Steven Ayre wrote: > >> Your log shows that you're entering the dialplan with destination number >> 2222. However there are no extensions that match this number. You need to >> create a dialplan extension match this number and bridge the call to the >> registered user. >> >> Your log shows that you're using the default configuration. This is only >> intended as an example. I recommend you replace or modify it to only do >> what you need. >> >> FreeSWITCH 1.0.6 is also a very old unsupported release that contains >> known problems. I suggest you upgrade to either 1.2.7 or Git master. >> >> -Steve >> >> >> >> >> On 28 March 2013 14:19, Syed Ibrahim Ghaznavi wrote: >> >>> Hi, >>> I have configured OpenBTS with Freeswitch and registered 2 users using >>> VBTS_New_User found here: >>> http://wush.net/trac/rangepublic/wiki/freeswitchConfig >>> >>> I can see the 2 tuples in the sqlite3.db, however when i attempt to make >>> a call between the 2 registered users, the log on Freeswitch is as follows: >>> >>> 2013-03-27 21:18:07.495839 [NOTICE] switch_channel.c:976 New Channel >>> sofia/internal/IMSI410071190004419 at 127.0.0.1 [e9051208-96f9-11e2-8f70- >>> 5979f626837d] >>> 2013-03-27 21:18:07.495839 [DEBUG] switch_core_session.c:975 Send signal >>> sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >>> 2013-03-27 21:18:07.495839 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change >>> CS_NEW >>> 2013-03-27 21:18:07.495839 [DEBUG] switch_core_session.c:975 Send signal >>> sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >>> 2013-03-27 21:18:07.495839 [DEBUG] switch_core_state_machine.c:433 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State NEW >>> 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:7697 IP 127.0.0.1 Approved by >>> acl "domains[]". Access Granted. >>> 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:5597 Channel sofia/internal/ >>> IMSI410071190004419 at 127.0.0.1 entering state [received][100] >>> 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:5608 Remote SDP: >>> v=0 >>> o=IMSI410071190004419 0 0 IN IP4 127.0.0.1 >>> s=Talk Time >>> t=0 0 >>> m=audio 16502 RTP/AVP 3 >>> c=IN IP4 127.0.0.1 >>> a=rtpmap:3 GSM/8000 >>> >>> 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:5821 (sofia/internal/ >>> IMSI410071190004419 at 127.0.0.1) State Change CS_NEW -> CS_INIT >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send >>> signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change >>> CS_INIT >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:454 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State INIT >>> 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:86 sofia/internal/ >>> IMSI410071190004419 at 127.0.0.1 SOFIA INIT >>> 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:126 (sofia/internal/ >>> IMSI410071190004419 at 127.0.0.1) State Change CS_INIT -> CS_ROUTING >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send >>> signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:454 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State INIT going to sleep >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change >>> CS_ROUTING >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:2012 (sofia/internal/ >>> IMSI410071190004419 at 127.0.0.1) Callstate Change DOWN -> RINGING >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:470 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State ROUTING >>> 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:149 sofia/internal/ >>> IMSI410071190004419 at 127.0.0.1 SOFIA ROUTING >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:117 >>> sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard ROUTING >>> 2013-03-27 21:18:07.515818 [INFO] mod_dialplan_xml.c:557 Processing >>> IMSI410071190004419 ->2222 in context public >>> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing >>> [public->unloop] continue=false >>> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (PASS) >>> [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false >>> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) >>> [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false >>> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing >>> [public->outside_call] continue=true >>> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Absolute >>> Condition [outside_call] >>> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Action >>> set(outside_call=true) >>> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Action >>> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >>> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing >>> [public->call_debug] continue=true >>> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) >>> [call_debug] ${call_debug}(false) =~ /^true$/ break=never >>> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing >>> [public->public_extensions] continue=false >>> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) >>> [public_extensions] destination_number(2222) =~ /^(10[01][0-9])$/ >>> break=on-false >>> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing >>> [public->public_did] continue=false >>> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) >>> [public_did] destination_number(2222) =~ /^(5551212)$/ break=on-false >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:167 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State Change CS_ROUTING >>> -> CS_EXECUTE >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send >>> signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:470 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State ROUTING going to >>> sleep >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change >>> CS_EXECUTE >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:477 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State EXECUTE >>> 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:242 sofia/internal/ >>> IMSI410071190004419 at 127.0.0.1 SOFIA EXECUTE >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:209 >>> sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard EXECUTE >>> EXECUTE sofia/internal/IMSI410071190004419 at 127.0.0.1set(outside_call=true) >>> 2013-03-27 21:18:07.515818 [DEBUG] mod_dptools.c:1367 sofia/internal/ >>> IMSI410071190004419 at 127.0.0.1 SET [outside_call]=[true] >>> EXECUTE sofia/internal/IMSI410071190004419 at 127.0.0.1export(RFC2822_DATE=Wed, 27 Mar 2013 21:18:07 +0500) >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:1143 EXPORT >>> (export_vars) [RFC2822_DATE]=[Wed, 27 Mar 2013 21:18:07 +0500] >>> 2013-03-27 21:18:07.515818 [NOTICE] switch_core_state_machine.c:262 >>> sofia/internal/IMSI410071190004419 at 127.0.0.1 has executed the last >>> dialplan instruction, hanging up. >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:3011 (sofia/internal/ >>> IMSI410071190004419 at 127.0.0.1) Callstate Change RINGING -> HANGUP >>> 2013-03-27 21:18:07.515818 [NOTICE] switch_core_state_machine.c:264 >>> Hangup sofia/internal/IMSI410071190004419 at 127.0.0.1 [CS_EXECUTE] >>> [NORMAL_CLEARING] >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:3034 Send signal >>> sofia/internal/IMSI410071190004419 at 127.0.0.1 [KILL] >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send >>> signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:477 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State EXECUTE going to >>> sleep >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change >>> CS_HANGUP >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:676 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State HANGUP >>> 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:503 Channel >>> sofia/internal/IMSI410071190004419 at 127.0.0.1 hanging up, cause: >>> NORMAL_CLEARING >>> 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:633 Responding to INVITE >>> with: 480 >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:48 >>> sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard HANGUP, cause: >>> NORMAL_CLEARING >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:676 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State HANGUP going to >>> sleep >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:446 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State Change CS_HANGUP >>> -> CS_REPORTING >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send >>> signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change >>> CS_REPORTING >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:758 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State REPORTING >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:92 >>> sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard REPORTING, cause: >>> NORMAL_CLEARING >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:758 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State REPORTING going to >>> sleep >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:440 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State Change >>> CS_REPORTING -> CS_DESTROY >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send >>> signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >>> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1518 Session 44 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Locked, Waiting on >>> external entities >>> 2013-03-27 21:18:07.515818 [NOTICE] switch_core_session.c:1536 Session >>> 44 (sofia/internal/IMSI410071190004419 at 127.0.0.1) Ended >>> 2013-03-27 21:18:07.515818 [NOTICE] switch_core_session.c:1540 Close >>> Channel sofia/internal/IMSI410071190004419 at 127.0.0.1 [CS_DESTROY] >>> 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:565 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Callstate Change HANGUP >>> -> DOWN >>> 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:568 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change >>> CS_DESTROY >>> 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:578 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State DESTROY >>> 2013-03-27 21:18:07.525838 [DEBUG] mod_sofia.c:396 sofia/internal/ >>> IMSI410071190004419 at 127.0.0.1 SOFIA DESTROY >>> 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:99 >>> sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard DESTROY >>> 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:578 >>> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State DESTROY going to >>> sleep >>> >>> Gratitude, >>> Ibrahim >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20130401/e5ef6dba/attachment-0001.html From kennedy4260 at gmail.com Mon Apr 1 06:17:14 2013 From: kennedy4260 at gmail.com (Kevin Kennedy) Date: Sun, 31 Mar 2013 19:17:14 -0700 Subject: [Freeswitch-users] RFC4904 Trunkgroup Identity Message-ID: I am new to freeswitch as far as testing it, but have been on the user-list for a long time. I have searched through my archive of emails as well as searched on google for any answer that I can find on this. I am looking for the configuration options for RFC4904, trunk-group identity. This is where you can set Freeswitch up to send unscreened calls through your trunkgroup using the tgrp and trunk-context that is in the registration for every outbound call. Any help would be appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130331/c00e01f3/attachment-0001.html From ashish at nms.co.in Mon Apr 1 09:07:29 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 1 Apr 2013 10:37:29 +0530 Subject: [Freeswitch-users] mod_perl hangupCause() returning 'NONE' and hangup_cause is null Message-ID: Hi Michael, I am getting nothing out of that. On Fri, Mar 29, 2013 at 1:55 AM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Access to rxfax channel variables (Michael Collins) > 2. Re: callee id inbound (Steven Schoch) > 3. Re: mod_perl hangupCause() returning 'NONE' and hangup_cause > is null (Michael Collins) > 4. Re: Calling extension using ruby (Michael Collins) > 5. Re: callee id inbound (Hermouet Erwan) > > > ---------- Forwarded message ---------- > From: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Thu, 28 Mar 2013 13:17:20 -0700 > Subject: Re: [Freeswitch-users] Access to rxfax channel variables > http://wiki.freeswitch.org/wiki/Variable_session_in_hangup_hook > > -MC > > On Thu, Mar 28, 2013 at 2:33 AM, Eugene Prokopiev wrote: > >> Hi, >> >> How to read rxfax channel variables described in >> http://wiki.freeswitch.org/wiki/Mod_spandsp#Controlling_the_app? >> >> My dialplan: >> >> >> >> >> > >> data="/var/spool/fax/$1/${strftime(%Y%m%d%H%M%S)}-${caller_id_number}.tif"/> >> >> >> But I see only: >> >> 2013-03-28 13:28:40.256497 [INFO] mod_commands.c:5899 FAX received >> from 2440663() : >> >> -- >> Regards, >> Eugene Prokopiev >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > ---------- Forwarded message ---------- > From: Steven Schoch > To: FreeSWITCH Users Help > Cc: > Date: Thu, 28 Mar 2013 13:18:18 -0700 > Subject: Re: [Freeswitch-users] callee id inbound > I understand. I'm sure there must be a parameter you can put in the > "user/12345@${domain} action, but I don't know what that is. Someone > more knowledgeable will have to answer that. > > -- > Steve > > > ---------- Forwarded message ---------- > From: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Thu, 28 Mar 2013 13:18:58 -0700 > Subject: Re: [Freeswitch-users] mod_perl hangupCause() returning 'NONE' > and hangup_cause is null > In a case like this you are better off doing a uuid_dump to a temp file > and sifting through the output to see what is actually available. My guess > is that there will be something in there that you can use. > > -MC > > On Thu, Mar 28, 2013 at 4:50 AM, Ashish gautam wrote: > >> Hi, >> >> I am running a perl script after the call lands on my dialplan. I want to >> get the status that whether the call was originated properly and and if yes >> what was the hangup cause. I am getting the hangup status using >> hangupCause() and it returns 'NONE' everytime no matter what the reason >> is. Also the hangup_cause variable is nill or not set everytime. >> >> Any help is appreciated. >> >> Thanks in advance!! >> >> -- >> Regards >> >> Ashish Gautam >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > ---------- Forwarded message ---------- > From: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Thu, 28 Mar 2013 13:22:51 -0700 > Subject: Re: [Freeswitch-users] Calling extension using ruby > something like this: > > originate {ignore_early_media=true}sofia/gateway/gwname/${phone_num} > &speak(flite|kal|${text_to_speak}) > > -MC > > On Thu, Mar 28, 2013 at 1:54 AM, Amit Kumar wrote: > >> I am trying to make an application that can initiate a call from the Web >> app to a PSTN line. So when a user clicks on call, the PSTN line will ring, >> and FS will play the text entered on the web app to the PSTN. >> >> I am able to dial the PSTN from a SIP phone(Zoiper). I am also able to >> call the PSTN line from the CLI, but I have no clue as to how to eliminate >> the SIP Phone all together. >> >> I can connect to FS using ESL, and send the command to initiate a call, >> but since no user is registered without the SIP phone registering them, I >> am not sure how to go ahead. >> >> Any 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > ---------- Forwarded message ---------- > From: Hermouet Erwan > To: FreeSWITCH Users Help > Cc: > Date: Thu, 28 Mar 2013 21:25:25 +0100 > Subject: Re: [Freeswitch-users] callee id inbound > Yes it s that > > > Steven Schoch a ?crit : >> >> I understand. I'm sure there must be a parameter you can put in the >> "user/12345@${domain} action, but I don't know what that is. Someone >> more knowledgeable will have to answer that. >> >> -- >> Steve >> >> ------------------------------ >> >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > Hermouet Erwan > Responsable technique > Bluetel > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/2dc54eba/attachment-0001.html From asaad2 at gmail.com Mon Apr 1 14:46:29 2013 From: asaad2 at gmail.com (BookBag) Date: Mon, 1 Apr 2013 06:46:29 -0400 Subject: [Freeswitch-users] FusionPBX or other on working FS In-Reply-To: <51595135.1010904@gmail.com> References: <51595135.1010904@gmail.com> Message-ID: You will have to re-enter your manual settings On Apr 1, 2013 5:30 AM, "Mimiko" wrote: > Hello. > > Recently I checked FusionPBX to see what it may help in administer FS. > First I downloaded the web files and pushed to web server. Running for > configuration ended to nothing. It created its tables in DB but the web > page does not work. Then I deployed FusionPBX from precompiled iso. It > seems intuitive for basic setups. > > Now I read that when setting up FusionPBX it will break any > configuration FS had. So I am worried if I will be able to restore the > configs from web GUI of FusionPBX to what it was, as I understood it is > no good to change config files of FS directly when FusionPBX is on them. > > Is it true? Can't I combine manual and via GUI configuring of FS? May be > other GUI will allow me more freedom? > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/50efcb0c/attachment.html From mehroz.ashraf85 at gmail.com Mon Apr 1 16:16:13 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Mon, 1 Apr 2013 05:16:13 -0700 (PDT) Subject: [Freeswitch-users] SSL/TLS customized encryption. In-Reply-To: References: <1364394004803-7589146.post@n2.nabble.com> <1364476282723-7589184.post@n2.nabble.com> <1364539539133-7589238.post@n2.nabble.com> Message-ID: <1364818573173-7589290.post@n2.nabble.com> Tried to change gentls_cert script where openssl creates keys and certificates , with Elliptic curve parameters, but it always generates error with the command. Is there any way around other than that. How can we generate certs/keys without using FS script? It might help too see what is really happening ! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SSL-TLS-customized-encryption-tp7589146p7589290.html Sent from the freeswitch-users mailing list archive at Nabble.com. From itsusama at gmail.com Mon Apr 1 17:17:10 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Mon, 1 Apr 2013 18:17:10 +0500 Subject: [Freeswitch-users] Mod Nibblebill and PGSQL in Core Message-ID: <009101ce2edb$3ba556d0$b2f00470$@gmail.com> Hi, if I enable pgsql in core how do I limit the database size, what would FS be logging in the database and why does mod nibblebill need it? If I understand correctly, mod_nibblebill uses a different database to bill the users. Regards. From shaheryarkh at gmail.com Mon Apr 1 17:29:29 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Mon, 1 Apr 2013 14:29:29 +0100 Subject: [Freeswitch-users] Mod Nibblebill and PGSQL in Core In-Reply-To: <009101ce2edb$3ba556d0$b2f00470$@gmail.com> References: <009101ce2edb$3ba556d0$b2f00470$@gmail.com> Message-ID: Yes, the billing database is separated but mod_nibble still needs to query core db, e.g. to check online users, call states, heartbeat etc. etc. Thank you. On Mon, Apr 1, 2013 at 2:17 PM, Usama Zaidi wrote: > Hi, > > if I enable pgsql in core how do I limit the database size, what would FS > be > logging in the database and why does mod nibblebill need it? If I > understand > correctly, mod_nibblebill uses a different database to bill the users. > > 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 > -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/5aca4515/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Apr 1 17:54:03 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 1 Apr 2013 14:54:03 +0100 Subject: [Freeswitch-users] 3 for 1 - core devs Message-ID: Hello all, Not sure if the core devs are still open to the idea of the 3 for 1 deal, but here goes; I have reviewed and commented the following tickets; http://jira.freeswitch.org/browse/FS-3964#comment-40390 http://jira.freeswitch.org/browse/FS-4208#comment-40391 http://jira.freeswitch.org/browse/FS-3899#comment-40392 If any core dev has a spare few moments, would you mind throwing your two cents on this ticket; http://jira.freeswitch.org/browse/FS-5245 I would have done 5 for 1, but I couldn't find any more tickets I was able to offer much help on. Thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/df6e7218/attachment.html From derkan at gmail.com Mon Apr 1 14:10:31 2013 From: derkan at gmail.com (=?UTF-8?B?RXJrYW4gRHVybXXFnw==?=) Date: Mon, 01 Apr 2013 13:10:31 +0300 Subject: [Freeswitch-users] Callcenter DB-Lock Problem(DELETE FROM members) Message-ID: <51595D17.7010403@gmail.com> Hi, When active calls increases(300 calls/min, cps:4-9) in the sytem we get errors like: 2013-03-30 19:29:40.913442 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] DELETE FROM members WHERE system = 'single_box' AND uuid = 'f8b239d0-995e-11e2-b825-b9a5cd559726' AND (abandoned_epoch = '1364664489' OR joined_epoch = '1364664395') We are using local PostgreSQL instance over UnixODBC for callcenter and core db(not default sqllite) Restarting DB does not resolve problem, same errors continues. Also no errors in postgre's log file. During problem, although free agents available; no calls are routed to agents; all calls were waiting on the queue. All calls seem to hang at queue(but ghost calls), only cleaned after clean FS restart. At http://jira.freeswitch.org/browse/FS-3127 says it is fixed. FS VERSION: 1.2.7 Compiled from stable tar release SYSTEM: CentOS release 6.3 x64 DB: Postgresql 9.2(on the same machine, only used by FreeSWITCH) HARDWARE: Xen Virtual Machine; 4x Xeon E5530 @ 2.40GHz, 8GB Memory Has anyone faced same problem? How can I resolve it? Regards, Erkan D. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/c64fc13e/attachment.html From avi at avimarcus.net Mon Apr 1 21:01:28 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 1 Apr 2013 20:01:28 +0300 Subject: [Freeswitch-users] Mod Nibblebill and PGSQL in Core In-Reply-To: <009101ce2edb$3ba556d0$b2f00470$@gmail.com> References: <009101ce2edb$3ba556d0$b2f00470$@gmail.com> Message-ID: mod_nibblebill used ODBC to connect to whichever DB you want it to. It doesn't take up any space, since it only decrements the account cash value (with no audit trail). -Avi Marcus BestFone On Mon, Apr 1, 2013 at 4:17 PM, Usama Zaidi wrote: > Hi, > > if I enable pgsql in core how do I limit the database size, what would FS > be > logging in the database and why does mod nibblebill need it? If I > understand > correctly, mod_nibblebill uses a different database to bill the users. > > 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/20130401/7800c3a3/attachment-0001.html From victor.chukalovskiy at gmail.com Mon Apr 1 22:38:06 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Mon, 01 Apr 2013 14:38:06 -0400 Subject: [Freeswitch-users] Updating FS to a particular version In-Reply-To: References: <5154F2D0.5010509@gmail.com> <51564CF0.6060008@gmail.com> Message-ID: <5159D40E.8090505@gmail.com> Great, thanks to everyone who helped. Is it worth adding to the WiKi? It's similar to http://wiki.freeswitch.org/wiki/Installation_Guide#Reverting_to_an_Earlier_Commit_in_Git but not exactly the same. The final version is: git fetch git checkout git clean -d -f -x rm -rf /usr/local/freeswitch/{lib,mod,bin}/* ./bootstrap.sh ./configure make make install -Victor On 13-03-30 01:13 PM, Avi Marcus wrote: > The code to delete the installed binaries is here: > > http://wiki.freeswitch.org/wiki/Installation_Guide#Reverting_to_an_Earlier_Commit_in_Git > rm -rf /usr/local/freeswitch/{lib,mod,bin}/* > > Also, you don't need to do a git pull -- just a git fetch. fetch grabs > the updates. Pull fetches the updates and then checks out the most > recent revision on the selected branch. So do fetch then checkout the > version you want. > > -Avi Marcus > BestFone > > On Sat, Mar 30, 2013 at 5:24 AM, Victor Chukalovskiy > > > wrote: > > Am I missing something? Any unnecessary steps? Based on Michael's > input, the following should be done in the source directory: > > git pull > git checkout > git clean -d -f -x > ./bootstrap.sh > ./configure > > make > make install > > Thank you, > Victor > > On 13-03-29 11:29 AM, Michael Collins wrote: >> FS will never overwrite existing configs in the conf directory. >> However, you can be extra safe by backing up your conf directory, >> performing your update, then restoring your conf directory. >> >> As far as the non-standard "configure" script that will need to >> be backed up as well. You'll need to restore that script after >> you run or re-run the bootstrap.sh script. Same goes for >> modules.conf - back it up and restore it after you run the >> modified configure script. In cases like this I recommend that >> you write a simple shell script and add a few comments to it so >> that the next time you do this in a few months you'll know not >> only what is supposed to happen (shell script commands) but why >> (comments). >> >> -MC >> >> On Thu, Mar 28, 2013 at 6:48 PM, Victor Chukalovskiy >> > > wrote: >> >> Hello, >> >> What would be the right way to update existing system to a >> particular >> version? >> >> Under normal conditions I'd do "make current" Here I want >> exactly the >> same, but: >> -specify the version to install >> Making sure to: >> -preserve non-standard ./configure string used during initial >> install >> -make sure conf directory is not overwritten by stock config. >> -make sure modules.conf is not overwritten / changed >> >> It should be simple...but want to avoid trial and error. WiKi is >> somewhat confusing. >> >> Thank you, >> -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 >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.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/20130401/07dec905/attachment.html From jsun at junsun.net Mon Apr 1 23:27:34 2013 From: jsun at junsun.net (Jun Sun) Date: Mon, 01 Apr 2013 12:27:34 -0700 Subject: [Freeswitch-users] bridging two outbound calls In-Reply-To: <5158D289.9090101@junsun.net> References: <515635C3.6040509@junsun.net> , <5156845C.4070302@junsun.net> <374BC379-8DAA-4CEA-AB0D-2DB313E31209@visionutveckling.se> <51570684.2010203@junsun.net>, <5157D80B.6070709@junsun.net> <1FFF97C269757C458224B7C895F35F15257F06@cantor.std.visionutv.se> <5158D289.9090101@junsun.net> Message-ID: <5159DFA6.30007@junsun.net> Hmm, crazy! After enabling "bypass_media" I can start to hear the conversation from both ends. originate {bypass_media=true}sofia/internal/15107079642 at 216.xxx.xxx.11:5060 &bridge(sofia/internal/15102991921 at 216.xxx.xxx.11:5060) This is even better for my callback purpose. However, I really thought without bypassing media would be easier to get work first. Still feel strange why it would not work ... Cheers. Jun On 3/31/2013 5:19 PM, Jun Sun wrote: > > Thanks for the pastebin suggestion. I have attached pastebin URL for the > output from fs_cli console (debug level 7), from initiating the command > to both parties picking up the phone. > > http://pastebin.com/tZewdvri > > I did not touch bypass_media in any way. I assume it should be off. (And > this is a fairly standard system, other than that it is hosted on AWS > EC2 machine) > > Since I"m using the same SIP trunk, even if bypass_media is set to true, > it probably should still work. > > Thanks in advance. I can get a pcap file if that will be helpful. > > Jun > > On 3/31/2013 12:18 AM, Peter Olsson wrote: >> Yes, it's there to separate if the second parameter is for en >> extension (no '&' prefix) or an application (using '&' prefix). Read >> the full description here: >> http://wiki.freeswitch.org/wiki/Mod_commands#originate. If you would >> allow a space in between it would make '&' the second parameter, and >> the actual string the third - which is not correct. This is just >> normal syntax rules for how to parse arguments. >> >> About the audio issue - make sure to get debug logs, and a pcap when >> making the call, post to pastebin. This is usually a NAT or codec >> issue. SInce you said it was working if calling a conference, another >> thing to check might be if you're settings bypass_media or not. >> >> /Peter >> ________________________________________ >> Fr?n: Jun Sun [jsun at junsun.net] >> Skickat: den 31 mars 2013 08:30 >> Till: FreeSWITCH Users Help >> Cc: Peter Olsson >> ?mne: Re: [Freeswitch-users] bridging two outbound calls >> >> Is there any specific reason why "&" must be immediately followed by the >> application name? I found this restriction pretty annoying (at least for >> newbies ;0) >> >> A simple patch would easily fix this. See one attached. Any takers? >> >> I'm still searching for the solution for no sound after bridging. Would >> appreciate any pointers. (Again, this really has to be one of the >> simplest cases ... why it has been so hard?) >> >> Cheers. >> >> Jun >> >> On 3/30/2013 8:36 AM, Jun Sun wrote: >>> >>> Oh, my god. That is it! After removing the extra space between "&" and >>> "bridge", the second number now gets dialed. >>> >>> However, I cannot hear each other between these two phones. I think the >>> signaling part is working, because hanging up one end will cause the >>> other end hung up. However, the media is not flowing through. >>> >>> This must be a simple mistake. Any pointers? >>> >>> BTW, using conference() app works, i.e., both ends can connect and talk. >>> So my system should be in general healthy state. >>> >>> Cheers. >>> >>> Jun >>> >>> On 3/29/2013 11:38 PM, Peter Olsson wrote: >>>> It looks like you have a space between & and bridge? It might be my >>>> email reader though. Anyway, it must be set like this: &bridge(). >>>> >>>> Also, I'm not sure about the tel: stuff, if you can set it that way, >>>> especially since there is a whitespace in between as well. >>>> >>>> /Peter >>>> >>>> 30 mar 2013 kl. 07:27 skrev "Jun Sun" : >>>> >>>>> >>>>> Yes, I tried. I can reverse the positions of those two numbers and >>>>> always the first number gets called and the second number gets >>>>> nothing. >>>>> >>>>> I start to wonder whether I'm using bridge() application wrongly. >>>>> Can it >>>>> dial out directly to a PSTN number via sofia? >>>>> >>>>> I was also fumbling with two orignate commands (followed by park() >>>>> application) and uuid_bridge to connect. No failures on console, >>>>> but the >>>>> two lines are not talking. >>>>> >>>>> Thanks. >>>>> >>>>> Jun >>>>> >>>>> On 3/29/2013 6:25 PM, Zoltan Medveczky wrote: >>>>>> You probably already tried this, but are you able to place a call to >>>>>> your party B endpoint at all (i.e. >>>>>> originate(sofia/internal/18005551212 >>>>>> @X.X.X.X:5060)? >>>>>> >>>>>> If that's failing, I'd say there's probably something wrong with your >>>>>> dial string. Perhaps your SIP UA is not bound to port 5060? >>>>>> >>>>>> On Fri, Mar 29, 2013 at 5:45 PM, Jun Sun >>>>> > wrote: >>>>>> >>>>>> I feel really stupid. This has to be one of the simplest >>>>>> cases in >>>>>> freeswitch, but I can't seem to get it work. >>>>>> >>>>>> My goal is to originate two outbound calls and bridge them >>>>>> together, a >>>>>> typical callback use case. I like to to do it from socket >>>>>> api/fs_cli. >>>>>> >>>>>> Here is what I typed in fs_cli: >>>>>> >>>>>> originate sofia/internal/15102991912 >>>>>> @X.X.X.X:5060 & >>>>>> bridge(sofia/internal/18005551212 >>>>>> @X.X.X.X:5060) >>>>>> >>>>>> The first leg is initiated and answered successfully, but the >>>>>> second leg >>>>>> never happens. From the console I don't see any action done by >>>>>> FS to do >>>>>> the bridging part. >>>>>> >>>>>> Any idea? Thanks in advance. >>>>>> >>>>>> Cheers. >>>>>> >>>>>> Jun >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 >>>> >>> >>> >> >> >> >> !DSPAM:5157d35132761492122807! >> > > From steveayre at gmail.com Mon Apr 1 23:46:49 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 1 Apr 2013 20:46:49 +0100 Subject: [Freeswitch-users] bridging two outbound calls In-Reply-To: <5159DFA6.30007@junsun.net> References: <515635C3.6040509@junsun.net> <5156845C.4070302@junsun.net> <374BC379-8DAA-4CEA-AB0D-2DB313E31209@visionutveckling.se> <51570684.2010203@junsun.net> <5157D80B.6070709@junsun.net> <1FFF97C269757C458224B7C895F35F15257F06@cantor.std.visionutv.se> <5158D289.9090101@junsun.net> <5159DFA6.30007@junsun.net> Message-ID: AFAIK on AWS EC2 the server runs behind NAT. My guess is your FS install isn't configured correctly to be behind NAT, so the server wasn't able to send media to FS correctly. But it can send correctly to itself (bypass_media). On 1 April 2013 20:27, Jun Sun wrote: > > Hmm, crazy! After enabling "bypass_media" I can start to hear the > conversation from both ends. > > originate > {bypass_media=true}sofia/internal/15107079642 at 216.xxx.xxx.11:5060 > &bridge(sofia/internal/15102991921 at 216.xxx.xxx.11:5060) > > This is even better for my callback purpose. However, I really thought > without bypassing media would be easier to get work first. Still feel > strange why it would not work ... > > Cheers. > > Jun > > On 3/31/2013 5:19 PM, Jun Sun wrote: > > > > Thanks for the pastebin suggestion. I have attached pastebin URL for the > > output from fs_cli console (debug level 7), from initiating the command > > to both parties picking up the phone. > > > > http://pastebin.com/tZewdvri > > > > I did not touch bypass_media in any way. I assume it should be off. (And > > this is a fairly standard system, other than that it is hosted on AWS > > EC2 machine) > > > > Since I"m using the same SIP trunk, even if bypass_media is set to true, > > it probably should still work. > > > > Thanks in advance. I can get a pcap file if that will be helpful. > > > > Jun > > > > On 3/31/2013 12:18 AM, Peter Olsson wrote: > >> Yes, it's there to separate if the second parameter is for en > >> extension (no '&' prefix) or an application (using '&' prefix). Read > >> the full description here: > >> http://wiki.freeswitch.org/wiki/Mod_commands#originate. If you would > >> allow a space in between it would make '&' the second parameter, and > >> the actual string the third - which is not correct. This is just > >> normal syntax rules for how to parse arguments. > >> > >> About the audio issue - make sure to get debug logs, and a pcap when > >> making the call, post to pastebin. This is usually a NAT or codec > >> issue. SInce you said it was working if calling a conference, another > >> thing to check might be if you're settings bypass_media or not. > >> > >> /Peter > >> ________________________________________ > >> Fr?n: Jun Sun [jsun at junsun.net] > >> Skickat: den 31 mars 2013 08:30 > >> Till: FreeSWITCH Users Help > >> Cc: Peter Olsson > >> ?mne: Re: [Freeswitch-users] bridging two outbound calls > >> > >> Is there any specific reason why "&" must be immediately followed by the > >> application name? I found this restriction pretty annoying (at least for > >> newbies ;0) > >> > >> A simple patch would easily fix this. See one attached. Any takers? > >> > >> I'm still searching for the solution for no sound after bridging. Would > >> appreciate any pointers. (Again, this really has to be one of the > >> simplest cases ... why it has been so hard?) > >> > >> Cheers. > >> > >> Jun > >> > >> On 3/30/2013 8:36 AM, Jun Sun wrote: > >>> > >>> Oh, my god. That is it! After removing the extra space between "&" and > >>> "bridge", the second number now gets dialed. > >>> > >>> However, I cannot hear each other between these two phones. I think the > >>> signaling part is working, because hanging up one end will cause the > >>> other end hung up. However, the media is not flowing through. > >>> > >>> This must be a simple mistake. Any pointers? > >>> > >>> BTW, using conference() app works, i.e., both ends can connect and > talk. > >>> So my system should be in general healthy state. > >>> > >>> Cheers. > >>> > >>> Jun > >>> > >>> On 3/29/2013 11:38 PM, Peter Olsson wrote: > >>>> It looks like you have a space between & and bridge? It might be my > >>>> email reader though. Anyway, it must be set like this: &bridge(). > >>>> > >>>> Also, I'm not sure about the tel: stuff, if you can set it that way, > >>>> especially since there is a whitespace in between as well. > >>>> > >>>> /Peter > >>>> > >>>> 30 mar 2013 kl. 07:27 skrev "Jun Sun" : > >>>> > >>>>> > >>>>> Yes, I tried. I can reverse the positions of those two numbers and > >>>>> always the first number gets called and the second number gets > >>>>> nothing. > >>>>> > >>>>> I start to wonder whether I'm using bridge() application wrongly. > >>>>> Can it > >>>>> dial out directly to a PSTN number via sofia? > >>>>> > >>>>> I was also fumbling with two orignate commands (followed by park() > >>>>> application) and uuid_bridge to connect. No failures on console, > >>>>> but the > >>>>> two lines are not talking. > >>>>> > >>>>> Thanks. > >>>>> > >>>>> Jun > >>>>> > >>>>> On 3/29/2013 6:25 PM, Zoltan Medveczky wrote: > >>>>>> You probably already tried this, but are you able to place a call to > >>>>>> your party B endpoint at all (i.e. > >>>>>> originate(sofia/internal/18005551212 > >>>>>> @X.X.X.X:5060)? > >>>>>> > >>>>>> If that's failing, I'd say there's probably something wrong with > your > >>>>>> dial string. Perhaps your SIP UA is not bound to port 5060? > >>>>>> > >>>>>> On Fri, Mar 29, 2013 at 5:45 PM, Jun Sun >>>>>> > wrote: > >>>>>> > >>>>>> I feel really stupid. This has to be one of the simplest > >>>>>> cases in > >>>>>> freeswitch, but I can't seem to get it work. > >>>>>> > >>>>>> My goal is to originate two outbound calls and bridge them > >>>>>> together, a > >>>>>> typical callback use case. I like to to do it from socket > >>>>>> api/fs_cli. > >>>>>> > >>>>>> Here is what I typed in fs_cli: > >>>>>> > >>>>>> originate sofia/internal/15102991912 > >>>>>> @X.X.X.X:5060 & > >>>>>> bridge(sofia/internal/18005551212 > >>>>>> @X.X.X.X:5060) > >>>>>> > >>>>>> The first leg is initiated and answered successfully, but the > >>>>>> second leg > >>>>>> never happens. From the console I don't see any action done by > >>>>>> FS to do > >>>>>> the bridging part. > >>>>>> > >>>>>> Any idea? Thanks in advance. > >>>>>> > >>>>>> Cheers. > >>>>>> > >>>>>> Jun > >>>>>> > >>>>>> > >>>>>> > _________________________________________________________________________ > >>>>>> > >>>>>> > >>>>>> Professional FreeSWITCH Consulting Services: > >>>>>> consulting at freeswitch.org > >>>>>> http://www.freeswitchsolutions.com > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> Official FreeSWITCH Sites > >>>>>> http://www.freeswitch.org > >>>>>> http://wiki.freeswitch.org > >>>>>> http://www.cluecon.com > >>>>>> > >>>>>> FreeSWITCH-users mailing list > >>>>>> FreeSWITCH-users at 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 > >>>> > >>> > >>> > >> > >> > >> > >> !DSPAM:5157d35132761492122807! > >> > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/02ef4564/attachment-0001.html From ebrahim.bararian at gmail.com Mon Apr 1 19:25:59 2013 From: ebrahim.bararian at gmail.com (Ebrahim Bararian) Date: Mon, 1 Apr 2013 19:55:59 +0430 Subject: [Freeswitch-users] Internal VoIP network + PSTN In-Reply-To: References: Message-ID: Thank you for your reply. I've read it all. I must state the problem in more details. As it is stated in the wiki we use the the internal configs for configuring internal calls and external configs for outbound calls. We also should state SIP provider for external calls. I want to run the freeswitch and two soft phones in the local net. Does it need any SIP provider for internal calls? I don't want to have internet connection. On Mon, Apr 1, 2013 at 2:22 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello, > > Have you started off looking over the wiki? > > If not, have a look at this first; > http://wiki.freeswitch.org/wiki/Getting_Started_Guide > > There is a learning curve with FreeSWITCH, as with any technology, and > it's not just a case of point and click. > > You'll need to take some time to learn how FreeSWITCH works and figure out > the approach you want to use.. Either that or buy a cudatel :) > > Cal > > On Sat, Mar 30, 2013 at 5:40 PM, Ebrahim Bararian < > ebrahim.bararian at gmail.com> wrote: > >> Hello all, >> >> I'm new to freeswitch and want to do two jobs with it. >> >> First, I want to make an Inernal IP network(two soft IP phones) connected >> to the PSTN network. >> >> Second I want to use the freeswitch to connect to the PSTN phone line via >> the voice modem. >> >> Can anyone help me with these problems? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20130401/1724df91/attachment.html From msc at freeswitch.org Mon Apr 1 23:57:02 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Apr 2013 12:57:02 -0700 Subject: [Freeswitch-users] Recording incoming calls In-Reply-To: <20130331211539.4DDCBDA021@jlo.kiwilink.co.nz> References: <20130331211539.4DDCBDA021@jlo.kiwilink.co.nz> Message-ID: Clive, I think the execute_on_answer is the way to go. However I believe you need to set that on the b-leg, so either include it as part of the bridge data or use the export app instead of set. Let us know how it goes. -MC On Sun, Mar 31, 2013 at 2:06 PM, Clive Lansink wrote: > Hello everyone > > I want to be able to record incoming calls. I had this all working using > record_session, until someone told me that when they phone in they no > longer hear any ringing before the phone is answered. Now I'm trying to set > things up so record_session works but without interfeering with the ringing > sound heard by the incoming caller. > > Incoming calls are received in the public context and then transferred to > extension 2000 in the default xml context. Then we have a local extension > in the default.xml dial plan that bridges to the incoming call group. If > the call is not answered within a certain timeframe, it transfers to > extension 1999 for voice mail - ie we don't have voice mail on each > extension. The definition of this extension is as follows: > > > > > data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> > data="{leg_timeout=18,ignore_early_media=true}${group_call(IncomingCall@ > ${domain_name})}"/> > > > > > My first attempt to add record_session was as follows: > > data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> > > > data="p:/media/phonecalls/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > This works except that the incoming caller hears no ringing sound while > our phones are ringing. In the log I see something like "Pre-Answer > sofia/external/021663999 at 27.111.14.67!" That is only there when the > record_session stuff is included so I presume the problem is that > record_session is somehow engaging with the call and the incoming caller > gets what amounts to an answered call, even though we haven't actually > answered it. > > Then I added > > but although the recording now starts at the time we answer the call, the > pre-answer thing is stil there and the incoming caller hears no ringing > sound. > > So then instead of executing record_session then and there as an > application, I tried > > I thought this might execute the record_session application when we answer > the call but it hasn't worked. > > Does anyone have any suggestions? > > > Clive Lansink > Email: Clive at Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/f23d67d6/attachment.html From RThodime at fuzebox.com Tue Apr 2 00:01:24 2013 From: RThodime at fuzebox.com (Raghavendra Thodime) Date: Mon, 1 Apr 2013 20:01:24 +0000 Subject: [Freeswitch-users] Jitter Buffer in Freeswitch Message-ID: Hi, I want to know if jitter buffer is enabled by default in the freeswitch. >From the code, it looks like it is not. But I am surprised that I couldn't see its effect when I start a conference and the quality is very clear. So wondering if there is any such buffering happens by default. Thanks Raghu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/734d64f1/attachment.html From msc at freeswitch.org Tue Apr 2 00:02:47 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Apr 2013 13:02:47 -0700 Subject: [Freeswitch-users] Internal VoIP network + PSTN In-Reply-To: References: Message-ID: Do you have your users set up in the directory? Do your phones register properly for those users? Can you make a call from one user to another user? If you have all three of those working then your "internal" stuff is good. Did you great a configuration file for your SIP provider? Is it located in sip_profiles/external/ subdirectory? What is the output of 'sofia status' from fs_cli? All of this information is in the wiki, but if you're having trouble with the way the wiki laid out then I might suggest that you get the FreeSWITCH Cookbook . It has simple "recipes" for doing all this stuff. (Disclosure: I am co-author of the FreeSWITCH Cookbook.) -MC On Mon, Apr 1, 2013 at 8:25 AM, Ebrahim Bararian wrote: > Thank you for your reply. I've read it all. > I must state the problem in more details. > As it is stated in the wiki we use the the internal configs for > configuring internal calls and external configs for outbound calls. We also > should state SIP provider for external calls. > I want to run the freeswitch and two soft phones in the local net. Does it > need any SIP provider for internal calls? I don't want to have internet > connection. > > > On Mon, Apr 1, 2013 at 2:22 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hello, >> >> Have you started off looking over the wiki? >> >> If not, have a look at this first; >> http://wiki.freeswitch.org/wiki/Getting_Started_Guide >> >> There is a learning curve with FreeSWITCH, as with any technology, and >> it's not just a case of point and click. >> >> You'll need to take some time to learn how FreeSWITCH works and figure >> out the approach you want to use.. Either that or buy a cudatel :) >> >> Cal >> >> On Sat, Mar 30, 2013 at 5:40 PM, Ebrahim Bararian < >> ebrahim.bararian at gmail.com> wrote: >> >>> Hello all, >>> >>> I'm new to freeswitch and want to do two jobs with it. >>> >>> First, I want to make an Inernal IP network(two soft IP phones) >>> connected to the PSTN network. >>> >>> Second I want to use the freeswitch to connect to the PSTN phone line >>> via the voice modem. >>> >>> Can anyone help me with these problems? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/2c4252b7/attachment-0001.html From shaheryarkh at gmail.com Tue Apr 2 00:43:32 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Mon, 1 Apr 2013 21:43:32 +0100 Subject: [Freeswitch-users] Change order of voicemail message retrival Message-ID: Hi, Is there any variable / parameter to change the order of messages played back while checking voicemail? Currently its FIFO, i am trying to play them in LIFO mode. Thank you. -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/d9661395/attachment.html From dvl36.ripe.nick at gmail.com Tue Apr 2 00:57:51 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Mon, 1 Apr 2013 23:57:51 +0300 Subject: [Freeswitch-users] Jitter Buffer in Freeswitch In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Jitterbuffer first line: The jitter buffer is implemented in the Sort Transportable Framed Utterances (STFU) library. The jitter buffer is *not enabled by default*. 2013/4/1 Raghavendra Thodime > Hi, > I want to know if jitter buffer is enabled by default in the freeswitch. > From the code, it looks like it is not. But I am surprised that I couldn't > see its effect when I start a conference and the quality is very clear. So > wondering if there is any such buffering happens by default. > > Thanks > Raghu > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/398cd264/attachment.html From jsun at junsun.net Tue Apr 2 01:08:10 2013 From: jsun at junsun.net (Jun Sun) Date: Mon, 1 Apr 2013 14:08:10 -0700 Subject: [Freeswitch-users] bridging two outbound calls In-Reply-To: References: <515635C3.6040509@junsun.net> <5156845C.4070302@junsun.net> <374BC379-8DAA-4CEA-AB0D-2DB313E31209@visionutveckling.se> <51570684.2010203@junsun.net> <5157D80B.6070709@junsun.net> <1FFF97C269757C458224B7C895F35F15257F06@cantor.std.visionutv.se> <5158D289.9090101@junsun.net> <5159DFA6.30007@junsun.net> Message-ID: Thanks, Steven. I think it's a very reasonable suspect, although it is puzzling to see conferencing works with two outbound calls. I've set freeswitch/ec2 up according to the wiki page, http://wiki.freeswitch.org/wiki/Amazon_ec2. I will double check. Cheers. Jun On Apr 1, 2013 12:54 PM, "Steven Ayre" wrote: > AFAIK on AWS EC2 the server runs behind NAT. My guess is your FS install > isn't configured correctly to be behind NAT, so the server wasn't able to > send media to FS correctly. But it can send correctly to itself > (bypass_media). > > > On 1 April 2013 20:27, Jun Sun wrote: > >> >> Hmm, crazy! After enabling "bypass_media" I can start to hear the >> conversation from both ends. >> >> originate >> {bypass_media=true}sofia/internal/15107079642 at 216.xxx.xxx.11:5060 >> &bridge(sofia/internal/15102991921 at 216.xxx.xxx.11:5060) >> >> This is even better for my callback purpose. However, I really thought >> without bypassing media would be easier to get work first. Still feel >> strange why it would not work ... >> >> Cheers. >> >> Jun >> >> On 3/31/2013 5:19 PM, Jun Sun wrote: >> > >> > Thanks for the pastebin suggestion. I have attached pastebin URL for the >> > output from fs_cli console (debug level 7), from initiating the command >> > to both parties picking up the phone. >> > >> > http://pastebin.com/tZewdvri >> > >> > I did not touch bypass_media in any way. I assume it should be off. (And >> > this is a fairly standard system, other than that it is hosted on AWS >> > EC2 machine) >> > >> > Since I"m using the same SIP trunk, even if bypass_media is set to true, >> > it probably should still work. >> > >> > Thanks in advance. I can get a pcap file if that will be helpful. >> > >> > Jun >> > >> > On 3/31/2013 12:18 AM, Peter Olsson wrote: >> >> Yes, it's there to separate if the second parameter is for en >> >> extension (no '&' prefix) or an application (using '&' prefix). Read >> >> the full description here: >> >> http://wiki.freeswitch.org/wiki/Mod_commands#originate. If you would >> >> allow a space in between it would make '&' the second parameter, and >> >> the actual string the third - which is not correct. This is just >> >> normal syntax rules for how to parse arguments. >> >> >> >> About the audio issue - make sure to get debug logs, and a pcap when >> >> making the call, post to pastebin. This is usually a NAT or codec >> >> issue. SInce you said it was working if calling a conference, another >> >> thing to check might be if you're settings bypass_media or not. >> >> >> >> /Peter >> >> ________________________________________ >> >> Fr?n: Jun Sun [jsun at junsun.net] >> >> Skickat: den 31 mars 2013 08:30 >> >> Till: FreeSWITCH Users Help >> >> Cc: Peter Olsson >> >> ?mne: Re: [Freeswitch-users] bridging two outbound calls >> >> >> >> Is there any specific reason why "&" must be immediately followed by >> the >> >> application name? I found this restriction pretty annoying (at least >> for >> >> newbies ;0) >> >> >> >> A simple patch would easily fix this. See one attached. Any takers? >> >> >> >> I'm still searching for the solution for no sound after bridging. Would >> >> appreciate any pointers. (Again, this really has to be one of the >> >> simplest cases ... why it has been so hard?) >> >> >> >> Cheers. >> >> >> >> Jun >> >> >> >> On 3/30/2013 8:36 AM, Jun Sun wrote: >> >>> >> >>> Oh, my god. That is it! After removing the extra space between "&" and >> >>> "bridge", the second number now gets dialed. >> >>> >> >>> However, I cannot hear each other between these two phones. I think >> the >> >>> signaling part is working, because hanging up one end will cause the >> >>> other end hung up. However, the media is not flowing through. >> >>> >> >>> This must be a simple mistake. Any pointers? >> >>> >> >>> BTW, using conference() app works, i.e., both ends can connect and >> talk. >> >>> So my system should be in general healthy state. >> >>> >> >>> Cheers. >> >>> >> >>> Jun >> >>> >> >>> On 3/29/2013 11:38 PM, Peter Olsson wrote: >> >>>> It looks like you have a space between & and bridge? It might be my >> >>>> email reader though. Anyway, it must be set like this: &bridge(). >> >>>> >> >>>> Also, I'm not sure about the tel: stuff, if you can set it that way, >> >>>> especially since there is a whitespace in between as well. >> >>>> >> >>>> /Peter >> >>>> >> >>>> 30 mar 2013 kl. 07:27 skrev "Jun Sun" : >> >>>> >> >>>>> >> >>>>> Yes, I tried. I can reverse the positions of those two numbers and >> >>>>> always the first number gets called and the second number gets >> >>>>> nothing. >> >>>>> >> >>>>> I start to wonder whether I'm using bridge() application wrongly. >> >>>>> Can it >> >>>>> dial out directly to a PSTN number via sofia? >> >>>>> >> >>>>> I was also fumbling with two orignate commands (followed by park() >> >>>>> application) and uuid_bridge to connect. No failures on console, >> >>>>> but the >> >>>>> two lines are not talking. >> >>>>> >> >>>>> Thanks. >> >>>>> >> >>>>> Jun >> >>>>> >> >>>>> On 3/29/2013 6:25 PM, Zoltan Medveczky wrote: >> >>>>>> You probably already tried this, but are you able to place a call >> to >> >>>>>> your party B endpoint at all (i.e. >> >>>>>> originate(sofia/internal/18005551212 >> >>>>>> @X.X.X.X:5060)? >> >>>>>> >> >>>>>> If that's failing, I'd say there's probably something wrong with >> your >> >>>>>> dial string. Perhaps your SIP UA is not bound to port 5060? >> >>>>>> >> >>>>>> On Fri, Mar 29, 2013 at 5:45 PM, Jun Sun > >>>>>> > wrote: >> >>>>>> >> >>>>>> I feel really stupid. This has to be one of the simplest >> >>>>>> cases in >> >>>>>> freeswitch, but I can't seem to get it work. >> >>>>>> >> >>>>>> My goal is to originate two outbound calls and bridge them >> >>>>>> together, a >> >>>>>> typical callback use case. I like to to do it from socket >> >>>>>> api/fs_cli. >> >>>>>> >> >>>>>> Here is what I typed in fs_cli: >> >>>>>> >> >>>>>> originate sofia/internal/15102991912 >> >>>>>> @X.X.X.X:5060 & >> >>>>>> bridge(sofia/internal/18005551212 >> >>>>>> @X.X.X.X:5060) >> >>>>>> >> >>>>>> The first leg is initiated and answered successfully, but the >> >>>>>> second leg >> >>>>>> never happens. From the console I don't see any action done by >> >>>>>> FS to do >> >>>>>> the bridging part. >> >>>>>> >> >>>>>> Any idea? Thanks in advance. >> >>>>>> >> >>>>>> Cheers. >> >>>>>> >> >>>>>> Jun >> >>>>>> >> >>>>>> >> >>>>>> >> _________________________________________________________________________ >> >>>>>> >> >>>>>> >> >>>>>> Professional FreeSWITCH Consulting Services: >> >>>>>> consulting at freeswitch.org >> >>>>>> http://www.freeswitchsolutions.com >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> Official FreeSWITCH Sites >> >>>>>> http://www.freeswitch.org >> >>>>>> http://wiki.freeswitch.org >> >>>>>> http://www.cluecon.com >> >>>>>> >> >>>>>> FreeSWITCH-users mailing list >> >>>>>> FreeSWITCH-users at 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 >> >>>> >> >>> >> >>> >> >> >> >> >> >> >> >> !DSPAM:5157d35132761492122807! >> >> >> > >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20130401/3c65ec0a/attachment-0001.html From kristin.king at quentustech.com Tue Apr 2 01:19:31 2013 From: kristin.king at quentustech.com (Kristin King) Date: Mon, 01 Apr 2013 14:19:31 -0700 Subject: [Freeswitch-users] Change order of voicemail message retrival In-Reply-To: References: Message-ID: <5159F9E3.9090204@quentustech.com> There wasn't an option for this, but I just finished coding one this morning. I'm filing the feature request and attaching the patch and hopefully it'll be going in shortly. Kristin King Quentus Technologies, INC 1037 NE 65th St, Ste 273 Seattle, WA 98115 Main: 877-211-9337 Office: 206-388-4778 Fax: 206-462-1861 Cell: 206-755-7329 Email: kristin.king at quentustech.com On 04/01/2013 01:43 PM, Muhammad Shahzad wrote: > Hi, > > Is there any variable / parameter to change the order of messages played > back while checking voicemail? Currently its FIFO, i am trying to play > them in LIFO mode. > > Thank you. > > > -- > Mit freundlichen Gr??en > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.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 info at shishir.com.np Tue Apr 2 01:20:18 2013 From: info at shishir.com.np (info at shishir.com.np) Date: Mon, 01 Apr 2013 14:20:18 -0700 Subject: [Freeswitch-users] Build Error Message-ID: Hi I Came across the following error while performing make current this morning. Please suggest. quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -I/usr/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -Ilibs/sofia-sip/libsofia-sip-ua/sdp -Ilibs/sofia-sip/libsofia-sip-ua/su -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/src/freeswitch/libs/apr/include -I/usr/src/freeswitch/libs/apr-util/include -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/src/freeswitch/libs/libtpl-1.5/src -I/usr/src/freeswitch/libs/stfu -I/usr/src/freeswitch/libs/sqlite -I/usr/src/freeswitch/libs/pcre -I/usr/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/src/freeswitch/libs/srtp/include -I/usr/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -I/usr/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_rtp.lo -MD -MP -MF .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c -fPIC -DPIC -o .libs/libfreeswitch_la-switch_rtp.o cc1: warnings being treated as errors src/switch_rtp.c: In function ?handle_ice?: src/switch_rtp.c:797: error: format ?%ld? expects type ?long int?, but argument 8 has type ?switch_size_t? src/switch_rtp.c: In function ?read_rtp_packet?: src/switch_rtp.c:3734: error: format ?%ld? expects type ?long int?, but argument 9 has type ?switch_size_t? src/switch_rtp.c:3906: error: format ?%ld? expects type ?long int?, but argument 11 has type ?switch_size_t? src/switch_rtp.c: In function ?rtp_common_write?: src/switch_rtp.c:5289: error: format ?%ld? expects type ?long int?, but argument 9 has type ?switch_size_t? src/switch_rtp.c:5293: error: format ?%ld? expects type ?long int?, but argument 9 has type ?switch_size_t? make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 make[2]: Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 Thanks From msc at freeswitch.org Tue Apr 2 01:49:24 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Apr 2013 14:49:24 -0700 Subject: [Freeswitch-users] RFC4904 Trunkgroup Identity In-Reply-To: References: Message-ID: I don't believe we support this as such. However, I've never seen a situation where FS can't route a call based on available criteria. Are these trunkgroup values in specific headers? If so it's possible to extract that information and use it for routing in the dialplan. If you have some call examples that you can put on pastebin.freeswitch.orgthat would be helpful. Thanks, MC On Sun, Mar 31, 2013 at 7:17 PM, Kevin Kennedy wrote: > I am new to freeswitch as far as testing it, but have been on the > user-list for a long time. I have searched through my archive of emails as > well as searched on google for any answer that I can find on this. I am > looking for the configuration options for RFC4904, trunk-group identity. > This is where you can set Freeswitch up to send unscreened calls through > your trunkgroup using the tgrp and trunk-context that is in the > registration for every outbound call. Any help would be 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/c42416c6/attachment.html From krice at freeswitch.org Tue Apr 2 01:54:41 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 01 Apr 2013 16:54:41 -0500 Subject: [Freeswitch-users] Build Error In-Reply-To: Message-ID: Update those should be fixed already... Major updates were rolled in this morning... On 4/1/13 4:20 PM, "info at shishir.com.np" wrote: > Hi I Came across the following error while performing make current this > morning. Please suggest. quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. > -I./src/include -I./libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -I/usr/src/freeswitch/libs/stfu > -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 > -Ilibs/sofia-sip/libsofia-sip-ua/sdp -Ilibs/sofia-sip/libsofia-sip-ua/su -g > -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE > -I/usr/src/freeswitch/libs/apr/include > -I/usr/src/freeswitch/libs/apr-util/include > -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib > -I/usr/src/freeswitch/libs/libtpl-1.5/src -I/usr/src/freeswitch/libs/stfu > -I/usr/src/freeswitch/libs/sqlite -I/usr/src/freeswitch/libs/pcre > -I/usr/src/freeswitch/libs/speex/include -Ilibs/speex/include > -I/usr/src/freeswitch/libs/srtp/include > -I/usr/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include > -I/usr/src/freeswitch/libs/spandsp/src > -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP > -I/usr/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src > -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall > -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT > libfreeswitch_la-switch_rtp.lo -MD -MP -MF > .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c -fPIC -DPIC -o > .libs/libfreeswitch_la-switch_rtp.o cc1: warnings being treated as > errors src/switch_rtp.c: In function ?handle_ice?: src/switch_rtp.c:797: > error: format ?%ld? expects type ?long int?, but argument 8 has type > ?switch_size_t? src/switch_rtp.c: In function > ?read_rtp_packet?: src/switch_rtp.c:3734: error: format ?%ld? expects type > ?long int?, but argument 9 has type ?switch_size_t? src/switch_rtp.c:3906: > error: format ?%ld? expects type ?long int?, but argument 11 has type > ?switch_size_t? src/switch_rtp.c: In function > ?rtp_common_write?: src/switch_rtp.c:5289: error: format ?%ld? expects type > ?long int?, but argument 9 has type ?switch_size_t? src/switch_rtp.c:5293: > error: format ?%ld? expects type ?long int?, but argument 9 has type > ?switch_size_t? make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 make[2]: > Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: > Leaving directory `/usr/src/freeswitch' make: *** [current] Error > 2 Thanks __________________________________________________________________ > _______ Professional FreeSWITCH Consulting > Services: consulting at freeswitch.org http://www.freeswitchsolutions.com FreeSW > ITCH-powered IP PBX: The CudaTel Communication > Server Official FreeSWITCH > Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon. > com FreeSWITCH-users mailing > list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From kristin.king at quentustech.com Tue Apr 2 02:02:26 2013 From: kristin.king at quentustech.com (Kristin King) Date: Mon, 01 Apr 2013 15:02:26 -0700 Subject: [Freeswitch-users] Change order of voicemail message retrival In-Reply-To: <5159F9E3.9090204@quentustech.com> References: <5159F9E3.9090204@quentustech.com> Message-ID: <515A03F2.3050506@quentustech.com> http://jira.freeswitch.org/browse/FS-5249 Kristin King Quentus Technologies, INC 1037 NE 65th St, Ste 273 Seattle, WA 98115 Main: 877-211-9337 Office: 206-388-4778 Fax: 206-462-1861 Cell: 206-755-7329 Email: kristin.king at quentustech.com On 04/01/2013 02:19 PM, Kristin King wrote: > There wasn't an option for this, but I just finished coding one this > morning. I'm filing the feature request and attaching the patch and > hopefully it'll be going in shortly. > > Kristin King > Quentus Technologies, INC > 1037 NE 65th St, Ste 273 > Seattle, WA 98115 > Main: 877-211-9337 > Office: 206-388-4778 > Fax: 206-462-1861 > Cell: 206-755-7329 > Email: kristin.king at quentustech.com > > On 04/01/2013 01:43 PM, Muhammad Shahzad wrote: >> Hi, >> >> Is there any variable / parameter to change the order of messages played >> back while checking voicemail? Currently its FIFO, i am trying to play >> them in LIFO mode. >> >> Thank you. >> >> >> -- >> Mit freundlichen Gr??en >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +49 176 99 83 10 85 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.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 From djbinter at gmail.com Tue Apr 2 02:09:16 2013 From: djbinter at gmail.com (DJB International) Date: Mon, 1 Apr 2013 15:09:16 -0700 Subject: [Freeswitch-users] Build Error In-Reply-To: References: Message-ID: Ken, I also experienced the built error today, but not sure whether it's related. http://jira.freeswitch.org/browse/FS-5248 On Mon, Apr 1, 2013 at 2:54 PM, Ken Rice wrote: > Update those should be fixed already... > > Major updates were rolled in this morning... > > > On 4/1/13 4:20 PM, "info at shishir.com.np" wrote: > > > Hi > > I Came across the following error while performing make current this > > > morning. Please suggest. > > > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. > > -I./src/include > -I./libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include > > > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include > > > -I/usr/src/freeswitch/libs/libteletone/src > -I/usr/src/freeswitch/libs/stfu > > -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 > > > -Ilibs/sofia-sip/libsofia-sip-ua/sdp -Ilibs/sofia-sip/libsofia-sip-ua/su > -g > > -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE > -D_LARGEFILE64_SOURCE > > -I/usr/src/freeswitch/libs/apr/include > > > -I/usr/src/freeswitch/libs/apr-util/include > > > -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib > > > -I/usr/src/freeswitch/libs/libtpl-1.5/src > -I/usr/src/freeswitch/libs/stfu > > -I/usr/src/freeswitch/libs/sqlite > -I/usr/src/freeswitch/libs/pcre > > -I/usr/src/freeswitch/libs/speex/include > -Ilibs/speex/include > > -I/usr/src/freeswitch/libs/srtp/include > > > -I/usr/src/freeswitch/libs/srtp/crypto/include > -Ilibs/srtp/crypto/include > > -I/usr/src/freeswitch/libs/spandsp/src > > > -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP > > > -I/usr/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > > > -I/usr/src/freeswitch/libs/curl/include > -I/usr/src/freeswitch/src/include > > -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src > > > -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden > > > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL > -Wall > > -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT > > > libfreeswitch_la-switch_rtp.lo -MD -MP -MF > > > .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c -fPIC -DPIC > -o > > .libs/libfreeswitch_la-switch_rtp.o > cc1: warnings being treated as > > errors > src/switch_rtp.c: In function ?handle_ice?: > src/switch_rtp.c:797: > > error: format ?%ld? expects type ?long int?, but > argument 8 has type > > ?switch_size_t? > src/switch_rtp.c: In function > > ?read_rtp_packet?: > src/switch_rtp.c:3734: error: format ?%ld? expects type > > ?long int?, but > argument 9 has type ?switch_size_t? > src/switch_rtp.c:3906: > > error: format ?%ld? expects type ?long int?, but > argument 11 has type > > ?switch_size_t? > src/switch_rtp.c: In function > > ?rtp_common_write?: > src/switch_rtp.c:5289: error: format ?%ld? expects type > > ?long int?, but > argument 9 has type ?switch_size_t? > src/switch_rtp.c:5293: > > error: format ?%ld? expects type ?long int?, but > argument 9 has type > > ?switch_size_t? > make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 > make[2]: > > Leaving directory `/usr/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: > > Leaving directory `/usr/src/freeswitch' > make: *** [current] Error > > 2 > > Thanks > > > __________________________________________________________________ > > _______ > Professional FreeSWITCH Consulting > > Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > FreeSW > > ITCH-powered IP PBX: The CudaTel Communication > > Server > > > Official FreeSWITCH > > Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon. > > com > > FreeSWITCH-users mailing > > list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman > > /listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > > ions/freeswitch-users > http://www.freeswitch.org > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #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/20130401/dfdb6867/attachment-0001.html From sertys at gmail.com Tue Apr 2 02:34:46 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Tue, 2 Apr 2013 00:34:46 +0200 Subject: [Freeswitch-users] Dynamic codec switching Message-ID: I remember fairly well that there is dynamic codec negotiation in fs from a big thread here on the list. My question is can i use this to overcome transcoding difficulties for ringback. I have g729 licences on the box, but prefer amr for mobile connections. Can i send early media in g729 and then switch to amr for actual call? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130402/c06f1a8c/attachment.html From msc at freeswitch.org Tue Apr 2 03:21:39 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Apr 2013 16:21:39 -0700 Subject: [Freeswitch-users] FusionPBX or other on working FS In-Reply-To: <51595135.1010904@gmail.com> References: <51595135.1010904@gmail.com> Message-ID: On Mon, Apr 1, 2013 at 2:19 AM, Mimiko wrote: > Hello. > > Recently I checked FusionPBX to see what it may help in administer FS. > First I downloaded the web files and pushed to web server. Running for > configuration ended to nothing. It created its tables in DB but the web > page does not work. Then I deployed FusionPBX from precompiled iso. It > seems intuitive for basic setups. > > Now I read that when setting up FusionPBX it will break any > configuration FS had. So I am worried if I will be able to restore the > configs from web GUI of FusionPBX to what it was, as I understood it is > no good to change config files of FS directly when FusionPBX is on them. > Naturally you'll want to back up your entire conf directory to make sure you don't lose anything before you move forward. > > Is it true? Can't I combine manual and via GUI configuring of FS? May be > other GUI will allow me more freedom? > GUIs by definition take away freedom in exchange for convenience. At some point you need to make FusionPBX "aware" of your custom changes. I suspect that it's best to start with a clean install and then use the FusionGUI to add your changes, either through the menus or with the XML editor. It's a pain the first time, but once it's done you'll be GUI-ready. -MC > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/e1b9401e/attachment.html From msc at freeswitch.org Tue Apr 2 03:26:46 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Apr 2013 16:26:46 -0700 Subject: [Freeswitch-users] Jitter Buffer in Freeswitch In-Reply-To: References: Message-ID: This page has good information, plus a link to a ML thread that has a lot more discussion: http://wiki.freeswitch.org/wiki/Jitterbuffer -MC On Mon, Apr 1, 2013 at 1:01 PM, Raghavendra Thodime wrote: > Hi, > I want to know if jitter buffer is enabled by default in the freeswitch. > From the code, it looks like it is not. But I am surprised that I couldn't > see its effect when I start a conference and the quality is very clear. So > wondering if there is any such buffering happens by default. > > Thanks > Raghu > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/9ec99af2/attachment.html From msc at freeswitch.org Tue Apr 2 03:28:43 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Apr 2013 16:28:43 -0700 Subject: [Freeswitch-users] Change order of voicemail message retrival In-Reply-To: <515A03F2.3050506@quentustech.com> References: <5159F9E3.9090204@quentustech.com> <515A03F2.3050506@quentustech.com> Message-ID: Thanks Kristin! that looks like a simple but effective patch. -MC On Mon, Apr 1, 2013 at 3:02 PM, Kristin King wrote: > http://jira.freeswitch.org/browse/FS-5249 > > Kristin King > Quentus Technologies, INC > 1037 NE 65th St, Ste 273 > Seattle, WA 98115 > Main: 877-211-9337 > Office: 206-388-4778 > Fax: 206-462-1861 > Cell: 206-755-7329 > Email: kristin.king at quentustech.com > > > On 04/01/2013 02:19 PM, Kristin King wrote: > > There wasn't an option for this, but I just finished coding one this > > morning. I'm filing the feature request and attaching the patch and > > hopefully it'll be going in shortly. > > > > Kristin King > > Quentus Technologies, INC > > 1037 NE 65th St, Ste 273 > > Seattle, WA 98115 > > Main: 877-211-9337 > > Office: 206-388-4778 > > Fax: 206-462-1861 > > Cell: 206-755-7329 > > Email: kristin.king at quentustech.com > > > > On 04/01/2013 01:43 PM, Muhammad Shahzad wrote: > >> Hi, > >> > >> Is there any variable / parameter to change the order of messages played > >> back while checking voicemail? Currently its FIFO, i am trying to play > >> them in LIFO mode. > >> > >> Thank you. > >> > >> > >> -- > >> Mit freundlichen Gr??en > >> Muhammad Shahzad > >> ----------------------------------- > >> CISCO Rich Media Communication Specialist (CRMCS) > >> CISCO Certified Network Associate (CCNA) > >> Cell: +49 176 99 83 10 85 > >> MSN: shari_786pk at hotmail.com > >> Email: shaheryarkh at googlemail.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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130401/1161ba00/attachment-0001.html From dvl36.ripe.nick at gmail.com Tue Apr 2 03:34:06 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Tue, 2 Apr 2013 02:34:06 +0300 Subject: [Freeswitch-users] Dynamic codec switching In-Reply-To: References: Message-ID: Why not? You should try. During the voice communication, codecs switched well. At least with SIPUA that I tried to use.(not so much,btw) 2013/4/2 Daniel Ivanov > I remember fairly well that there is dynamic codec negotiation in fs from > a big thread here on the list. My question is can i use this to overcome > transcoding difficulties for ringback. I have g729 licences on the box, but > prefer amr for mobile connections. Can i send early media in g729 and then > switch to amr for actual call? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130402/2d51991f/attachment.html From info at shishir.com.np Tue Apr 2 08:54:13 2013 From: info at shishir.com.np (info at shishir.com.np) Date: Mon, 01 Apr 2013 21:54:13 -0700 Subject: [Freeswitch-users] Build Error In-Reply-To: References: Message-ID: I still get the same error after updating too, I even tried on the fresh git checkout, its same issue. Is it only me getting this error ? On 01.04.2013 15:09, DJB International wrote: > Ken, > > I also experienced the built error today, but not sure whether it's > related.? http://jira.freeswitch.org/browse/FS-5248 [27] > > On Mon, Apr 1, 2013 at 2:54 PM, Ken Rice wrote: > >> Update those should be fixed already... >> >> Major updates were rolled in this morning... >> >> On 4/1/13 4:20 PM, "info at shishir.com.np [1]" wrote: >> >> > Hi >> >> I Came across the following error while performing make current >> this >> > >> morning. Please suggest. >> >> quiet_libtool: compile: ?gcc -DHAVE_CONFIG_H -I. >> > -I./src/include >> -I./libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include >> > >> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >> > >> -I/usr/src/freeswitch/libs/libteletone/src >> -I/usr/src/freeswitch/libs/stfu >> > -fPIC -fvisibility=hidden >> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >> > >> -Ilibs/sofia-sip/libsofia-sip-ua/sdp >> -Ilibs/sofia-sip/libsofia-sip-ua/su >> -g >> > -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE >> -D_LARGEFILE64_SOURCE >> > -I/usr/src/freeswitch/libs/apr/include >> > >> -I/usr/src/freeswitch/libs/apr-util/include >> > >> -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib >> > >> -I/usr/src/freeswitch/libs/libtpl-1.5/src >> -I/usr/src/freeswitch/libs/stfu >> > -I/usr/src/freeswitch/libs/sqlite >> -I/usr/src/freeswitch/libs/pcre >> > -I/usr/src/freeswitch/libs/speex/include >> -Ilibs/speex/include >> > -I/usr/src/freeswitch/libs/srtp/include >> > >> -I/usr/src/freeswitch/libs/srtp/crypto/include >> -Ilibs/srtp/crypto/include >> > -I/usr/src/freeswitch/libs/spandsp/src >> > >> -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP >> > >> -I/usr/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT >> > >> -I/usr/src/freeswitch/libs/curl/include >> -I/usr/src/freeswitch/src/include >> > -I/usr/src/freeswitch/src/include >> -I/usr/src/freeswitch/libs/libteletone/src >> > >> -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden >> > >> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb >> -DHAVE_OPENSSL >> -Wall >> > -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT >> > >> libfreeswitch_la-switch_rtp.lo -MD -MP -MF >> > >> .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c ?-fPIC >> -DPIC >> -o >> > .libs/libfreeswitch_la-switch_rtp.o >> cc1: warnings being treated as >> > errors >> src/switch_rtp.c: In function ?handle_ice?: >> src/switch_rtp.c:797: >> > error: format ?%ld? expects type ?long int?, but >> argument 8 has type >> > ?switch_size_t? >> src/switch_rtp.c: In function >> > ?read_rtp_packet?: >> src/switch_rtp.c:3734: error: format ?%ld? expects type >> > ?long int?, but >> argument 9 has type ?switch_size_t? >> src/switch_rtp.c:3906: >> > error: format ?%ld? expects type ?long int?, but >> argument 11 has type >> > ?switch_size_t? >> src/switch_rtp.c: In function >> > ?rtp_common_write?: >> src/switch_rtp.c:5289: error: format ?%ld? expects type >> > ?long int?, but >> argument 9 has type ?switch_size_t? >> src/switch_rtp.c:5293: >> > error: format ?%ld? expects type ?long int?, but >> argument 9 has type >> > ?switch_size_t? >> >> make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 >> make[2]: >> > Leaving directory `/usr/src/freeswitch' >> make[1]: *** [all] Error 2 >> make[1]: >> > Leaving directory `/usr/src/freeswitch' >> make: *** [current] Error >> > 2 >> >> Thanks >> >> __________________________________________________________________ >> > _______ >> Professional FreeSWITCH Consulting >> > Services: >> consulting at freeswitch.org [3] >> http://www.freeswitchsolutions.com [4] >> >> FreeSW >> > ITCH-powered IP PBX: The CudaTel Communication >> > Server >> [5] >> >> Official FreeSWITCH >> > Sites >> http://www.freeswitch.org [6] >> http://wiki.freeswitch.org [7] >> http://www.cluecon [8]. >> > com >> >> FreeSWITCH-users mailing >> > list >> FreeSWITCH-users at lists.freeswitch.org [9] >> http://lists.freeswitch.org/mailman [10] >> > /listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt [11] >> > ions/freeswitch-users >> http://www.freeswitch.org [12] >> >> -- >> Ken >> http://www.FreeSWITCH.org [13] >> http://www.ClueCon.com [14] >> http://www.OSTAG.org [15] >> irc.freenode.net [16] #freeswitch >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [17] >> http://www.freeswitchsolutions.com [18] >> >> >> [19] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [20] >> http://wiki.freeswitch.org [21] >> http://www.cluecon.com [22] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [23] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [24] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [25] >> http://www.freeswitch.org [26] > > > > Links: > ------ > [1] mailto:info at shishir.com.np > [2] mailto:info at shishir.com.np > [3] mailto:consulting at freeswitch.org > [4] http://www.freeswitchsolutions.com > [5] > [6] http://www.freeswitch.org > [7] http://wiki.freeswitch.org > [8] http://www.cluecon > [9] mailto:FreeSWITCH-users at lists.freeswitch.org > [10] http://lists.freeswitch.org/mailman > [11] http://lists.freeswitch.org/mailman/opt > [12] http://www.freeswitch.org > [13] http://www.FreeSWITCH.org > [14] http://www.ClueCon.com > [15] http://www.OSTAG.org > [16] http://irc.freenode.net > [17] mailto:consulting at freeswitch.org > [18] http://www.freeswitchsolutions.com > [19] > [20] http://www.freeswitch.org > [21] http://wiki.freeswitch.org > [22] http://www.cluecon.com > [23] mailto:FreeSWITCH-users at lists.freeswitch.org > [24] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [25] http://lists.freeswitch.org/mailman/options/freeswitch-users > [26] http://www.freeswitch.org > [27] http://jira.freeswitch.org/browse/FS-5248 > [28] mailto:krice at freeswitch.org From info at shishir.com.np Tue Apr 2 10:14:01 2013 From: info at shishir.com.np (info at shishir.com.np) Date: Mon, 01 Apr 2013 23:14:01 -0700 Subject: [Freeswitch-users] Build Error In-Reply-To: References: Message-ID: <548004e6efb1a17d8d97126332f78dd5@shishir.com.np> Thanks guys! Error was resolve with the latest changes on the git. It was resolved after latest push, "git clean -fdx", git pull, with bootstrap and ./configure Thanks again guys! On 01.04.2013 21:54, info at shishir.com.np wrote: > I still get the same error after updating too, I even tried on the > fresh git checkout, its same issue. Is it only me getting this error > ? > > > On 01.04.2013 15:09, DJB International wrote: >> Ken, >> >> I also experienced the built error today, but not sure whether it's >> related.? http://jira.freeswitch.org/browse/FS-5248 [27] >> >> On Mon, Apr 1, 2013 at 2:54 PM, Ken Rice wrote: >> >>> Update those should be fixed already... >>> >>> Major updates were rolled in this morning... >>> >>> On 4/1/13 4:20 PM, "info at shishir.com.np [1]" wrote: >>> >>> > Hi >>> >>> I Came across the following error while performing make current >>> this >>> > >>> morning. Please suggest. >>> >>> quiet_libtool: compile: ?gcc -DHAVE_CONFIG_H -I. >>> > -I./src/include >>> -I./libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include >>> > >>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >>> > >>> -I/usr/src/freeswitch/libs/libteletone/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -fPIC -fvisibility=hidden >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>> > >>> -Ilibs/sofia-sip/libsofia-sip-ua/sdp >>> -Ilibs/sofia-sip/libsofia-sip-ua/su >>> -g >>> > -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE >>> -D_LARGEFILE64_SOURCE >>> > -I/usr/src/freeswitch/libs/apr/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>> > >>> -I/usr/src/freeswitch/libs/libtpl-1.5/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -I/usr/src/freeswitch/libs/sqlite >>> -I/usr/src/freeswitch/libs/pcre >>> > -I/usr/src/freeswitch/libs/speex/include >>> -Ilibs/speex/include >>> > -I/usr/src/freeswitch/libs/srtp/include >>> > >>> -I/usr/src/freeswitch/libs/srtp/crypto/include >>> -Ilibs/srtp/crypto/include >>> > -I/usr/src/freeswitch/libs/spandsp/src >>> > >>> -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP >>> > >>> -I/usr/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT >>> > >>> -I/usr/src/freeswitch/libs/curl/include >>> -I/usr/src/freeswitch/src/include >>> > -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/libs/libteletone/src >>> > >>> -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden >>> > >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb >>> -DHAVE_OPENSSL >>> -Wall >>> > -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT >>> > >>> libfreeswitch_la-switch_rtp.lo -MD -MP -MF >>> > >>> .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c ?-fPIC >>> -DPIC >>> -o >>> > .libs/libfreeswitch_la-switch_rtp.o >>> cc1: warnings being treated as >>> > errors >>> src/switch_rtp.c: In function ?handle_ice?: >>> src/switch_rtp.c:797: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 8 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?read_rtp_packet?: >>> src/switch_rtp.c:3734: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:3906: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 11 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?rtp_common_write?: >>> src/switch_rtp.c:5289: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:5293: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 9 has type >>> > ?switch_size_t? >>> >>> make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 >>> make[2]: >>> > Leaving directory `/usr/src/freeswitch' >>> make[1]: *** [all] Error 2 >>> make[1]: >>> > Leaving directory `/usr/src/freeswitch' >>> make: *** [current] Error >>> > 2 >>> >>> Thanks >>> >>> __________________________________________________________________ >>> > _______ >>> Professional FreeSWITCH Consulting >>> > Services: >>> consulting at freeswitch.org [3] >>> http://www.freeswitchsolutions.com [4] >>> >>> FreeSW >>> > ITCH-powered IP PBX: The CudaTel Communication >>> > Server >>> [5] >>> >>> Official FreeSWITCH >>> > Sites >>> http://www.freeswitch.org [6] >>> http://wiki.freeswitch.org [7] >>> http://www.cluecon [8]. >>> > com >>> >>> FreeSWITCH-users mailing >>> > list >>> FreeSWITCH-users at lists.freeswitch.org [9] >>> http://lists.freeswitch.org/mailman [10] >>> > /listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt [11] >>> > ions/freeswitch-users >>> http://www.freeswitch.org [12] >>> >>> -- >>> Ken >>> http://www.FreeSWITCH.org [13] >>> http://www.ClueCon.com [14] >>> http://www.OSTAG.org [15] >>> irc.freenode.net [16] #freeswitch >>> >>> >> >> >> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org [17] >>> http://www.freeswitchsolutions.com [18] >>> >>> >>> [19] >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org [20] >>> http://wiki.freeswitch.org [21] >>> http://www.cluecon.com [22] >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org [23] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [24] >>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> [25] >>> http://www.freeswitch.org [26] >> >> >> >> Links: >> ------ >> [1] mailto:info at shishir.com.np >> [2] mailto:info at shishir.com.np >> [3] mailto:consulting at freeswitch.org >> [4] http://www.freeswitchsolutions.com >> [5] >> [6] http://www.freeswitch.org >> [7] http://wiki.freeswitch.org >> [8] http://www.cluecon >> [9] mailto:FreeSWITCH-users at lists.freeswitch.org >> [10] http://lists.freeswitch.org/mailman >> [11] http://lists.freeswitch.org/mailman/opt >> [12] http://www.freeswitch.org >> [13] http://www.FreeSWITCH.org >> [14] http://www.ClueCon.com >> [15] http://www.OSTAG.org >> [16] http://irc.freenode.net >> [17] mailto:consulting at freeswitch.org >> [18] http://www.freeswitchsolutions.com >> [19] >> [20] http://www.freeswitch.org >> [21] http://wiki.freeswitch.org >> [22] http://www.cluecon.com >> [23] mailto:FreeSWITCH-users at lists.freeswitch.org >> [24] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [25] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [26] http://www.freeswitch.org >> [27] http://jira.freeswitch.org/browse/FS-5248 >> [28] mailto:krice at 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 jsun at junsun.net Tue Apr 2 10:50:39 2013 From: jsun at junsun.net (Jun Sun) Date: Mon, 01 Apr 2013 23:50:39 -0700 Subject: [Freeswitch-users] bridging two outbound calls In-Reply-To: References: <515635C3.6040509@junsun.net> <5156845C.4070302@junsun.net> <374BC379-8DAA-4CEA-AB0D-2DB313E31209@visionutveckling.se> <51570684.2010203@junsun.net> <5157D80B.6070709@junsun.net> <1FFF97C269757C458224B7C895F35F15257F06@cantor.std.visionutv.se> <5158D289.9090101@junsun.net> <5159DFA6.30007@junsun.net> Message-ID: <515A7FBF.2070905@junsun.net> Mystery solved! I did not open UDP port 16384-32768. I thought I have done that, but apparently took it away (or created a new instance somewhere along the way). Re-opening the ports makes the bridging call happy. Thanks for your help, Steven, Peter. Apparently conferencing outbound calls only generates outbound RTP traffic which can still work even without those ports open. Cheers. Jun On 4/1/2013 2:08 PM, Jun Sun wrote: > Thanks, Steven. I think it's a very reasonable suspect, although it is > puzzling to see conferencing works with two outbound calls. > > I've set freeswitch/ec2 up according to the wiki page, > http://wiki.freeswitch.org/wiki/Amazon_ec2. I will double check. > > Cheers. > > Jun > > On Apr 1, 2013 12:54 PM, "Steven Ayre" > wrote: > > AFAIK on AWS EC2 the server runs behind NAT. My guess is your FS > install isn't configured correctly to be behind NAT, so the server > wasn't able to send media to FS correctly. But it can send correctly > to itself (bypass_media). > > > On 1 April 2013 20:27, Jun Sun > wrote: > > > Hmm, crazy! After enabling "bypass_media" I can start to hear the > conversation from both ends. > > originate > {bypass_media=true}sofia/internal/15107079642 > @216.xxx.xxx.11:5060 > &bridge(sofia/internal/15102991921 > @216.xxx.xxx.11:5060) > > This is even better for my callback purpose. However, I really > thought > without bypassing media would be easier to get work first. Still > feel > strange why it would not work ... > > Cheers. > > Jun > > On 3/31/2013 5:19 PM, Jun Sun wrote: > > > > Thanks for the pastebin suggestion. I have attached pastebin > URL for the > > output from fs_cli console (debug level 7), from initiating > the command > > to both parties picking up the phone. > > > > http://pastebin.com/tZewdvri > > > > I did not touch bypass_media in any way. I assume it should > be off. (And > > this is a fairly standard system, other than that it is > hosted on AWS > > EC2 machine) > > > > Since I"m using the same SIP trunk, even if bypass_media is > set to true, > > it probably should still work. > > > > Thanks in advance. I can get a pcap file if that will be helpful. > > > > Jun > > > > On 3/31/2013 12:18 AM, Peter Olsson wrote: > >> Yes, it's there to separate if the second parameter is for en > >> extension (no '&' prefix) or an application (using '&' > prefix). Read > >> the full description here: > >> http://wiki.freeswitch.org/wiki/Mod_commands#originate. If > you would > >> allow a space in between it would make '&' the second > parameter, and > >> the actual string the third - which is not correct. This is just > >> normal syntax rules for how to parse arguments. > >> > >> About the audio issue - make sure to get debug logs, and a > pcap when > >> making the call, post to pastebin. This is usually a NAT or > codec > >> issue. SInce you said it was working if calling a > conference, another > >> thing to check might be if you're settings bypass_media or not. > >> > >> /Peter > >> ________________________________________ > >> Fr?n: Jun Sun [jsun at junsun.net ] > >> Skickat: den 31 mars 2013 08:30 > >> Till: FreeSWITCH Users Help > >> Cc: Peter Olsson > >> ?mne: Re: [Freeswitch-users] bridging two outbound calls > >> > >> Is there any specific reason why "&" must be immediately > followed by the > >> application name? I found this restriction pretty annoying > (at least for > >> newbies ;0) > >> > >> A simple patch would easily fix this. See one attached. Any > takers? > >> > >> I'm still searching for the solution for no sound after > bridging. Would > >> appreciate any pointers. (Again, this really has to be one > of the > >> simplest cases ... why it has been so hard?) > >> > >> Cheers. > >> > >> Jun > >> > >> On 3/30/2013 8:36 AM, Jun Sun wrote: > >>> > >>> Oh, my god. That is it! After removing the extra space > between "&" and > >>> "bridge", the second number now gets dialed. > >>> > >>> However, I cannot hear each other between these two phones. > I think the > >>> signaling part is working, because hanging up one end will > cause the > >>> other end hung up. However, the media is not flowing through. > >>> > >>> This must be a simple mistake. Any pointers? > >>> > >>> BTW, using conference() app works, i.e., both ends can > connect and talk. > >>> So my system should be in general healthy state. > >>> > >>> Cheers. > >>> > >>> Jun > >>> > >>> On 3/29/2013 11:38 PM, Peter Olsson wrote: > >>>> It looks like you have a space between & and bridge? It > might be my > >>>> email reader though. Anyway, it must be set like this: > &bridge(). > >>>> > >>>> Also, I'm not sure about the tel: stuff, if you can set it > that way, > >>>> especially since there is a whitespace in between as well. > >>>> > >>>> /Peter > >>>> > >>>> 30 mar 2013 kl. 07:27 skrev "Jun Sun" >: > >>>> > >>>>> > >>>>> Yes, I tried. I can reverse the positions of those two > numbers and > >>>>> always the first number gets called and the second number > gets > >>>>> nothing. > >>>>> > >>>>> I start to wonder whether I'm using bridge() application > wrongly. > >>>>> Can it > >>>>> dial out directly to a PSTN number via sofia? > >>>>> > >>>>> I was also fumbling with two orignate commands (followed > by park() > >>>>> application) and uuid_bridge to connect. No failures on > console, > >>>>> but the > >>>>> two lines are not talking. > >>>>> > >>>>> Thanks. > >>>>> > >>>>> Jun > >>>>> > >>>>> On 3/29/2013 6:25 PM, Zoltan Medveczky wrote: > >>>>>> You probably already tried this, but are you able to > place a call to > >>>>>> your party B endpoint at all (i.e. > >>>>>> originate(sofia/internal/18005551212 > >>>>>> >@X.X.X.X:5060)? > >>>>>> > >>>>>> If that's failing, I'd say there's probably something > wrong with your > >>>>>> dial string. Perhaps your SIP UA is not bound to port 5060? > >>>>>> > >>>>>> On Fri, Mar 29, 2013 at 5:45 PM, Jun Sun > > >>>>>> >> wrote: > >>>>>> > >>>>>> I feel really stupid. This has to be one of the > simplest > >>>>>> cases in > >>>>>> freeswitch, but I can't seem to get it work. > >>>>>> > >>>>>> My goal is to originate two outbound calls and > bridge them > >>>>>> together, a > >>>>>> typical callback use case. I like to to do it from > socket > >>>>>> api/fs_cli. > >>>>>> > >>>>>> Here is what I typed in fs_cli: > >>>>>> > >>>>>> originate sofia/internal/15102991912 > >>>>>> >@X.X.X.X:5060 & > >>>>>> bridge(sofia/internal/18005551212 > >>>>>> >@X.X.X.X:5060) > >>>>>> > >>>>>> The first leg is initiated and answered > successfully, but the > >>>>>> second leg > >>>>>> never happens. From the console I don't see any > action done by > >>>>>> FS to do > >>>>>> the bridging part. > >>>>>> > >>>>>> Any idea? Thanks in advance. > >>>>>> > >>>>>> Cheers. > >>>>>> > >>>>>> Jun > >>>>>> > >>>>>> > >>>>>> > _________________________________________________________________________ > >>>>>> > >>>>>> > >>>>>> Professional FreeSWITCH Consulting Services: > >>>>>> consulting at freeswitch.org > > > > >>>>>> http://www.freeswitchsolutions.com > >>>>>> > >>>>>> FreeSWITCH-powered IP PBX: The CudaTel > Communication Server > >>>>>> > >>>>>> > >>>>>> Official FreeSWITCH Sites > >>>>>> http://www.freeswitch.org > >>>>>> http://wiki.freeswitch.org > >>>>>> http://www.cluecon.com > >>>>>> > >>>>>> FreeSWITCH-users mailing list > >>>>>> FreeSWITCH-users at 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 > >>>> > >>> > >>> > >> > >> > >> > >> !DSPAM:5157d35132761492122807! > >> > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Alexander.Haugg at c4b.de Tue Apr 2 11:01:04 2013 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Tue, 2 Apr 2013 07:01:04 +0000 Subject: [Freeswitch-users] Build Error In-Reply-To: <548004e6efb1a17d8d97126332f78dd5@shishir.com.np> References: <548004e6efb1a17d8d97126332f78dd5@shishir.com.np> Message-ID: I have some build errors if i try to build the "Freeswitch.2012.sln". My GIT repository is up-to-date. The first error is: 32>------ Rebuild All started: Project: iksemel, Configuration: Debug Win32 ------ 3> dso_lib.c 32> utility.c 32> stream.c 32>..\..\iksemel\src\stream.c(23): fatal error C1083: Cannot open include file: 'openssl/ssl.h': No such file or directory 32> sha.c 32> sax.c But the file exist: freeswitch\libs\openssl-1.0.1c\include\openssl\ssl.h Thanks -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von info at shishir.com.np Gesendet: Dienstag, 2. April 2013 08:14 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Build Error Thanks guys! Error was resolve with the latest changes on the git. It was resolved after latest push, "git clean -fdx", git pull, with bootstrap and ./configure Thanks again guys! On 01.04.2013 21:54, info at shishir.com.np wrote: > I still get the same error after updating too, I even tried on the > fresh git checkout, its same issue. Is it only me getting this error ? > > > On 01.04.2013 15:09, DJB International wrote: >> Ken, >> >> I also experienced the built error today, but not sure whether it's >> related.? http://jira.freeswitch.org/browse/FS-5248 [27] >> >> On Mon, Apr 1, 2013 at 2:54 PM, Ken Rice wrote: >> >>> Update those should be fixed already... >>> >>> Major updates were rolled in this morning... >>> >>> On 4/1/13 4:20 PM, "info at shishir.com.np [1]" wrote: >>> >>> > Hi >>> >>> I Came across the following error while performing make current >>> this >>> > >>> morning. Please suggest. >>> >>> quiet_libtool: compile: ?gcc -DHAVE_CONFIG_H -I. >>> > -I./src/include >>> -I./libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include >>> > >>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >>> > >>> -I/usr/src/freeswitch/libs/libteletone/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -fPIC -fvisibility=hidden >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>> > >>> -Ilibs/sofia-sip/libsofia-sip-ua/sdp >>> -Ilibs/sofia-sip/libsofia-sip-ua/su >>> -g >>> > -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE >>> -D_LARGEFILE64_SOURCE >>> > -I/usr/src/freeswitch/libs/apr/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>> > >>> -I/usr/src/freeswitch/libs/libtpl-1.5/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -I/usr/src/freeswitch/libs/sqlite >>> -I/usr/src/freeswitch/libs/pcre >>> > -I/usr/src/freeswitch/libs/speex/include >>> -Ilibs/speex/include >>> > -I/usr/src/freeswitch/libs/srtp/include >>> > >>> -I/usr/src/freeswitch/libs/srtp/crypto/include >>> -Ilibs/srtp/crypto/include >>> > -I/usr/src/freeswitch/libs/spandsp/src >>> > >>> -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP >>> > >>> -I/usr/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT >>> > >>> -I/usr/src/freeswitch/libs/curl/include >>> -I/usr/src/freeswitch/src/include >>> > -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/libs/libteletone/src >>> > >>> -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden >>> > >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb >>> -DHAVE_OPENSSL >>> -Wall >>> > -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT >>> > >>> libfreeswitch_la-switch_rtp.lo -MD -MP -MF >>> > >>> .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c ?-fPIC >>> -DPIC >>> -o >>> > .libs/libfreeswitch_la-switch_rtp.o >>> cc1: warnings being treated as >>> > errors >>> src/switch_rtp.c: In function ?handle_ice?: >>> src/switch_rtp.c:797: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 8 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?read_rtp_packet?: >>> src/switch_rtp.c:3734: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:3906: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 11 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?rtp_common_write?: >>> src/switch_rtp.c:5289: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:5293: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 9 has type >>> > ?switch_size_t? >>> >>> make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 >>> make[2]: >>> > Leaving directory `/usr/src/freeswitch' >>> make[1]: *** [all] Error 2 >>> make[1]: >>> > Leaving directory `/usr/src/freeswitch' >>> make: *** [current] Error >>> > 2 >>> >>> Thanks >>> >>> __________________________________________________________________ >>> > _______ >>> Professional FreeSWITCH Consulting >>> > Services: >>> consulting at freeswitch.org [3] >>> http://www.freeswitchsolutions.com [4] >>> >>> FreeSW >>> > ITCH-powered IP PBX: The CudaTel Communication >>> > Server >>> [5] >>> >>> Official FreeSWITCH >>> > Sites >>> http://www.freeswitch.org [6] >>> http://wiki.freeswitch.org [7] >>> http://www.cluecon [8]. >>> > com >>> >>> FreeSWITCH-users mailing >>> > list >>> FreeSWITCH-users at lists.freeswitch.org [9] >>> http://lists.freeswitch.org/mailman [10] >>> > /listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt [11] >>> > ions/freeswitch-users >>> http://www.freeswitch.org [12] >>> >>> -- >>> Ken >>> http://www.FreeSWITCH.org [13] >>> http://www.ClueCon.com [14] >>> http://www.OSTAG.org [15] >>> irc.freenode.net [16] #freeswitch >>> >>> >> >> >> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org [17] >>> http://www.freeswitchsolutions.com [18] >>> >>> >>> [19] >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org [20] >>> http://wiki.freeswitch.org [21] >>> http://www.cluecon.com [22] >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org [23] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [24] >>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> [25] >>> http://www.freeswitch.org [26] >> >> >> >> Links: >> ------ >> [1] mailto:info at shishir.com.np >> [2] mailto:info at shishir.com.np >> [3] mailto:consulting at freeswitch.org >> [4] http://www.freeswitchsolutions.com >> [5] >> [6] http://www.freeswitch.org >> [7] http://wiki.freeswitch.org >> [8] http://www.cluecon >> [9] mailto:FreeSWITCH-users at lists.freeswitch.org >> [10] http://lists.freeswitch.org/mailman >> [11] http://lists.freeswitch.org/mailman/opt >> [12] http://www.freeswitch.org >> [13] http://www.FreeSWITCH.org >> [14] http://www.ClueCon.com >> [15] http://www.OSTAG.org >> [16] http://irc.freenode.net >> [17] mailto:consulting at freeswitch.org >> [18] http://www.freeswitchsolutions.com >> [19] >> [20] http://www.freeswitch.org >> [21] http://wiki.freeswitch.org >> [22] http://www.cluecon.com >> [23] mailto:FreeSWITCH-users at lists.freeswitch.org >> [24] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [25] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [26] http://www.freeswitch.org >> [27] http://jira.freeswitch.org/browse/FS-5248 >> [28] mailto:krice at 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 shaheryarkh at gmail.com Tue Apr 2 11:14:15 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Tue, 2 Apr 2013 08:14:15 +0100 Subject: [Freeswitch-users] Change order of voicemail message retrival In-Reply-To: References: <5159F9E3.9090204@quentustech.com> <515A03F2.3050506@quentustech.com> Message-ID: Great. Thanks guys. When can i expect this patch merged in trunk? Thank you. On Tue, Apr 2, 2013 at 12:28 AM, Michael Collins wrote: > Thanks Kristin! that looks like a simple but effective patch. > -MC > > > On Mon, Apr 1, 2013 at 3:02 PM, Kristin King > wrote: > >> http://jira.freeswitch.org/browse/FS-5249 >> >> Kristin King >> Quentus Technologies, INC >> 1037 NE 65th St, Ste 273 >> Seattle, WA 98115 >> Main: 877-211-9337 >> Office: 206-388-4778 >> Fax: 206-462-1861 >> Cell: 206-755-7329 >> Email: kristin.king at quentustech.com >> >> >> On 04/01/2013 02:19 PM, Kristin King wrote: >> > There wasn't an option for this, but I just finished coding one this >> > morning. I'm filing the feature request and attaching the patch and >> > hopefully it'll be going in shortly. >> > >> > Kristin King >> > Quentus Technologies, INC >> > 1037 NE 65th St, Ste 273 >> > Seattle, WA 98115 >> > Main: 877-211-9337 >> > Office: 206-388-4778 >> > Fax: 206-462-1861 >> > Cell: 206-755-7329 >> > Email: kristin.king at quentustech.com >> > >> > On 04/01/2013 01:43 PM, Muhammad Shahzad wrote: >> >> Hi, >> >> >> >> Is there any variable / parameter to change the order of messages >> played >> >> back while checking voicemail? Currently its FIFO, i am trying to play >> >> them in LIFO mode. >> >> >> >> Thank you. >> >> >> >> >> >> -- >> >> Mit freundlichen Gr??en >> >> Muhammad Shahzad >> >> ----------------------------------- >> >> CISCO Rich Media Communication Specialist (CRMCS) >> >> CISCO Certified Network Associate (CCNA) >> >> Cell: +49 176 99 83 10 85 >> >> MSN: shari_786pk at hotmail.com >> >> Email: shaheryarkh at googlemail.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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.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 > > -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130402/8b0754f4/attachment-0001.html From POlsson at enghouse.com Tue Apr 2 11:22:42 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Tue, 2 Apr 2013 07:22:42 +0000 Subject: [Freeswitch-users] Build Error Message-ID: <1FFF97C269757C458224B7C895F35F1525933A@cantor.std.visionutv.se> In latest git head this solution file has been renamed to: Freeswitch.2008.unsupported.sln So it sounds like you're not on latest. Also, as the name indicates - VS2008 is not actually supported anymore, so I'm not sure if it's even supposed to work. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Alexander Haugg Skickat: den 2 april 2013 09:01 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Build Error I have some build errors if i try to build the "Freeswitch.2012.sln". My GIT repository is up-to-date. The first error is: 32>------ Rebuild All started: Project: iksemel, Configuration: Debug 32>Win32 ------ 3> dso_lib.c 32> utility.c 32> stream.c 32>..\..\iksemel\src\stream.c(23): fatal error C1083: Cannot open 32>include file: 'openssl/ssl.h': No such file or directory 32> sha.c 32> sax.c But the file exist: freeswitch\libs\openssl-1.0.1c\include\openssl\ssl.h Thanks -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von info at shishir.com.np Gesendet: Dienstag, 2. April 2013 08:14 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Build Error Thanks guys! Error was resolve with the latest changes on the git. It was resolved after latest push, "git clean -fdx", git pull, with bootstrap and ./configure Thanks again guys! On 01.04.2013 21:54, info at shishir.com.np wrote: > I still get the same error after updating too, I even tried on the > fresh git checkout, its same issue. Is it only me getting this error ? > > > On 01.04.2013 15:09, DJB International wrote: >> Ken, >> >> I also experienced the built error today, but not sure whether it's >> related.? http://jira.freeswitch.org/browse/FS-5248 [27] >> >> On Mon, Apr 1, 2013 at 2:54 PM, Ken Rice wrote: >> >>> Update those should be fixed already... >>> >>> Major updates were rolled in this morning... >>> >>> On 4/1/13 4:20 PM, "info at shishir.com.np [1]" wrote: >>> >>> > Hi >>> >>> I Came across the following error while performing make current this >>> > >>> morning. Please suggest. >>> >>> quiet_libtool: compile: ?gcc -DHAVE_CONFIG_H -I. >>> > -I./src/include >>> -I./libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include >>> > >>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >>> > >>> -I/usr/src/freeswitch/libs/libteletone/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -fPIC -fvisibility=hidden >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>> > >>> -Ilibs/sofia-sip/libsofia-sip-ua/sdp >>> -Ilibs/sofia-sip/libsofia-sip-ua/su >>> -g >>> > -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE >>> -D_LARGEFILE64_SOURCE >>> > -I/usr/src/freeswitch/libs/apr/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>> > >>> -I/usr/src/freeswitch/libs/libtpl-1.5/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -I/usr/src/freeswitch/libs/sqlite >>> -I/usr/src/freeswitch/libs/pcre >>> > -I/usr/src/freeswitch/libs/speex/include >>> -Ilibs/speex/include >>> > -I/usr/src/freeswitch/libs/srtp/include >>> > >>> -I/usr/src/freeswitch/libs/srtp/crypto/include >>> -Ilibs/srtp/crypto/include >>> > -I/usr/src/freeswitch/libs/spandsp/src >>> > >>> -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP >>> > >>> -I/usr/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT >>> > >>> -I/usr/src/freeswitch/libs/curl/include >>> -I/usr/src/freeswitch/src/include >>> > -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/libs/libteletone/src >>> > >>> -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden >>> > >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb >>> -DHAVE_OPENSSL -Wall >>> > -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT >>> > >>> libfreeswitch_la-switch_rtp.lo -MD -MP -MF >>> > >>> .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c ?-fPIC >>> -DPIC -o >>> > .libs/libfreeswitch_la-switch_rtp.o >>> cc1: warnings being treated as >>> > errors >>> src/switch_rtp.c: In function ?handle_ice?: >>> src/switch_rtp.c:797: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 8 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?read_rtp_packet?: >>> src/switch_rtp.c:3734: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:3906: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 11 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?rtp_common_write?: >>> src/switch_rtp.c:5289: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:5293: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 9 has type >>> > ?switch_size_t? >>> >>> make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 >>> make[2]: >>> > Leaving directory `/usr/src/freeswitch' >>> make[1]: *** [all] Error 2 >>> make[1]: >>> > Leaving directory `/usr/src/freeswitch' >>> make: *** [current] Error >>> > 2 >>> >>> Thanks >>> >>> __________________________________________________________________ >>> > _______ >>> Professional FreeSWITCH Consulting >>> > Services: >>> consulting at freeswitch.org [3] >>> http://www.freeswitchsolutions.com [4] >>> >>> FreeSW >>> > ITCH-powered IP PBX: The CudaTel Communication Server >>> [5] >>> >>> Official FreeSWITCH >>> > Sites >>> http://www.freeswitch.org [6] >>> http://wiki.freeswitch.org [7] >>> http://www.cluecon [8]. >>> > com >>> >>> FreeSWITCH-users mailing >>> > list >>> FreeSWITCH-users at lists.freeswitch.org [9] >>> http://lists.freeswitch.org/mailman [10] >>> > /listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt [11] >>> > ions/freeswitch-users >>> http://www.freeswitch.org [12] >>> >>> -- >>> Ken >>> http://www.FreeSWITCH.org [13] >>> http://www.ClueCon.com [14] >>> http://www.OSTAG.org [15] >>> irc.freenode.net [16] #freeswitch >>> >>> >> >> >> _____________________________________________________________________ >> ____ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org [17] >>> http://www.freeswitchsolutions.com [18] >>> >>> >>> [19] >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org [20] >>> http://wiki.freeswitch.org [21] >>> http://www.cluecon.com [22] >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org [23] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [24] >>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >>> [25] >>> http://www.freeswitch.org [26] >> >> >> >> Links: >> ------ >> [1] mailto:info at shishir.com.np >> [2] mailto:info at shishir.com.np >> [3] mailto:consulting at freeswitch.org >> [4] http://www.freeswitchsolutions.com >> [5] >> [6] http://www.freeswitch.org >> [7] http://wiki.freeswitch.org >> [8] http://www.cluecon >> [9] mailto:FreeSWITCH-users at lists.freeswitch.org >> [10] http://lists.freeswitch.org/mailman >> [11] http://lists.freeswitch.org/mailman/opt >> [12] http://www.freeswitch.org >> [13] http://www.FreeSWITCH.org >> [14] http://www.ClueCon.com >> [15] http://www.OSTAG.org >> [16] http://irc.freenode.net >> [17] mailto:consulting at freeswitch.org [18] >> http://www.freeswitchsolutions.com >> [19] >> [20] http://www.freeswitch.org >> [21] http://wiki.freeswitch.org >> [22] http://www.cluecon.com >> [23] mailto:FreeSWITCH-users at lists.freeswitch.org >> [24] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [25] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [26] http://www.freeswitch.org >> [27] http://jira.freeswitch.org/browse/FS-5248 >> [28] mailto:krice at 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-use > rs > 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 !DSPAM:515a7f6032762128275286! From steveayre at gmail.com Tue Apr 2 11:22:36 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 2 Apr 2013 08:22:36 +0100 Subject: [Freeswitch-users] Dynamic codec switching In-Reply-To: References: Message-ID: Remember AMR only works in passthrough mode. Though that should be fine for a bridge. You're probably looking for this: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_media_reneg On 1 April 2013 23:34, Daniel Ivanov wrote: > I remember fairly well that there is dynamic codec negotiation in fs from > a big thread here on the list. My question is can i use this to overcome > transcoding difficulties for ringback. I have g729 licences on the box, but > prefer amr for mobile connections. Can i send early media in g729 and then > switch to amr for actual call? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130402/517f87ae/attachment.html From chang33.tw at gmail.com Tue Apr 2 11:26:20 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Tue, 02 Apr 2013 15:26:20 +0800 Subject: [Freeswitch-users] choppy audio Message-ID: <515A881C.6050709@gmail.com> Hi, I have two softphone clients commnunicating by G729. The audio quality of direct call between them is perfect. But when talking via FS, I got choppy audio in agent side(client side remains perfect). I had set bypass_media_after_bridge=true in dialplan, not work, and the RTP traffic is still through FS. Any advice? Thanks. Jimmy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130402/c254856c/attachment.html From Alexander.Haugg at c4b.de Tue Apr 2 11:48:11 2013 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Tue, 2 Apr 2013 07:48:11 +0000 Subject: [Freeswitch-users] Build Error In-Reply-To: <1FFF97C269757C458224B7C895F35F1525933A@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1525933A@cantor.std.visionutv.se> Message-ID: No, i'd write "Freeswitch.2012.sln" not 2008 -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Peter Olsson Gesendet: Dienstag, 2. April 2013 09:23 An: 'FreeSWITCH Users Help' Betreff: Re: [Freeswitch-users] Build Error In latest git head this solution file has been renamed to: Freeswitch.2008.unsupported.sln So it sounds like you're not on latest. Also, as the name indicates - VS2008 is not actually supported anymore, so I'm not sure if it's even supposed to work. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Alexander Haugg Skickat: den 2 april 2013 09:01 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Build Error I have some build errors if i try to build the "Freeswitch.2012.sln". My GIT repository is up-to-date. The first error is: 32>------ Rebuild All started: Project: iksemel, Configuration: Debug 32>Win32 ------ 3> dso_lib.c 32> utility.c 32> stream.c 32>..\..\iksemel\src\stream.c(23): fatal error C1083: Cannot open 32>include file: 'openssl/ssl.h': No such file or directory 32> sha.c 32> sax.c But the file exist: freeswitch\libs\openssl-1.0.1c\include\openssl\ssl.h Thanks -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von info at shishir.com.np Gesendet: Dienstag, 2. April 2013 08:14 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Build Error Thanks guys! Error was resolve with the latest changes on the git. It was resolved after latest push, "git clean -fdx", git pull, with bootstrap and ./configure Thanks again guys! On 01.04.2013 21:54, info at shishir.com.np wrote: > I still get the same error after updating too, I even tried on the > fresh git checkout, its same issue. Is it only me getting this error ? > > > On 01.04.2013 15:09, DJB International wrote: >> Ken, >> >> I also experienced the built error today, but not sure whether it's >> related.? http://jira.freeswitch.org/browse/FS-5248 [27] >> >> On Mon, Apr 1, 2013 at 2:54 PM, Ken Rice wrote: >> >>> Update those should be fixed already... >>> >>> Major updates were rolled in this morning... >>> >>> On 4/1/13 4:20 PM, "info at shishir.com.np [1]" wrote: >>> >>> > Hi >>> >>> I Came across the following error while performing make current this >>> > >>> morning. Please suggest. >>> >>> quiet_libtool: compile: ?gcc -DHAVE_CONFIG_H -I. >>> > -I./src/include >>> -I./libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include >>> > >>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >>> > >>> -I/usr/src/freeswitch/libs/libteletone/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -fPIC -fvisibility=hidden >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>> > >>> -Ilibs/sofia-sip/libsofia-sip-ua/sdp >>> -Ilibs/sofia-sip/libsofia-sip-ua/su >>> -g >>> > -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE >>> -D_LARGEFILE64_SOURCE >>> > -I/usr/src/freeswitch/libs/apr/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>> > >>> -I/usr/src/freeswitch/libs/libtpl-1.5/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -I/usr/src/freeswitch/libs/sqlite >>> -I/usr/src/freeswitch/libs/pcre >>> > -I/usr/src/freeswitch/libs/speex/include >>> -Ilibs/speex/include >>> > -I/usr/src/freeswitch/libs/srtp/include >>> > >>> -I/usr/src/freeswitch/libs/srtp/crypto/include >>> -Ilibs/srtp/crypto/include >>> > -I/usr/src/freeswitch/libs/spandsp/src >>> > >>> -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP >>> > >>> -I/usr/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT >>> > >>> -I/usr/src/freeswitch/libs/curl/include >>> -I/usr/src/freeswitch/src/include >>> > -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/libs/libteletone/src >>> > >>> -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden >>> > >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb >>> -DHAVE_OPENSSL -Wall >>> > -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT >>> > >>> libfreeswitch_la-switch_rtp.lo -MD -MP -MF >>> > >>> .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c ?-fPIC >>> -DPIC -o >>> > .libs/libfreeswitch_la-switch_rtp.o >>> cc1: warnings being treated as >>> > errors >>> src/switch_rtp.c: In function ?handle_ice?: >>> src/switch_rtp.c:797: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 8 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?read_rtp_packet?: >>> src/switch_rtp.c:3734: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:3906: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 11 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?rtp_common_write?: >>> src/switch_rtp.c:5289: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:5293: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 9 has type >>> > ?switch_size_t? >>> >>> make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 >>> make[2]: >>> > Leaving directory `/usr/src/freeswitch' >>> make[1]: *** [all] Error 2 >>> make[1]: >>> > Leaving directory `/usr/src/freeswitch' >>> make: *** [current] Error >>> > 2 >>> >>> Thanks >>> >>> __________________________________________________________________ >>> > _______ >>> Professional FreeSWITCH Consulting >>> > Services: >>> consulting at freeswitch.org [3] >>> http://www.freeswitchsolutions.com [4] >>> >>> FreeSW >>> > ITCH-powered IP PBX: The CudaTel Communication Server >>> [5] >>> >>> Official FreeSWITCH >>> > Sites >>> http://www.freeswitch.org [6] >>> http://wiki.freeswitch.org [7] >>> http://www.cluecon [8]. >>> > com >>> >>> FreeSWITCH-users mailing >>> > list >>> FreeSWITCH-users at lists.freeswitch.org [9] >>> http://lists.freeswitch.org/mailman [10] >>> > /listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt [11] >>> > ions/freeswitch-users >>> http://www.freeswitch.org [12] >>> >>> -- >>> Ken >>> http://www.FreeSWITCH.org [13] >>> http://www.ClueCon.com [14] >>> http://www.OSTAG.org [15] >>> irc.freenode.net [16] #freeswitch >>> >>> >> >> >> _____________________________________________________________________ >> ____ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org [17] >>> http://www.freeswitchsolutions.com [18] >>> >>> >>> [19] >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org [20] >>> http://wiki.freeswitch.org [21] >>> http://www.cluecon.com [22] >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org [23] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [24] >>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >>> [25] >>> http://www.freeswitch.org [26] >> >> >> >> Links: >> ------ >> [1] mailto:info at shishir.com.np >> [2] mailto:info at shishir.com.np >> [3] mailto:consulting at freeswitch.org >> [4] http://www.freeswitchsolutions.com >> [5] >> [6] http://www.freeswitch.org >> [7] http://wiki.freeswitch.org >> [8] http://www.cluecon >> [9] mailto:FreeSWITCH-users at lists.freeswitch.org >> [10] http://lists.freeswitch.org/mailman >> [11] http://lists.freeswitch.org/mailman/opt >> [12] http://www.freeswitch.org >> [13] http://www.FreeSWITCH.org >> [14] http://www.ClueCon.com >> [15] http://www.OSTAG.org >> [16] http://irc.freenode.net >> [17] mailto:consulting at freeswitch.org [18] >> http://www.freeswitchsolutions.com >> [19] >> [20] http://www.freeswitch.org >> [21] http://wiki.freeswitch.org >> [22] http://www.cluecon.com >> [23] mailto:FreeSWITCH-users at lists.freeswitch.org >> [24] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [25] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [26] http://www.freeswitch.org >> [27] http://jira.freeswitch.org/browse/FS-5248 >> [28] mailto:krice at 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-use > rs > 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 !DSPAM:515a7f6032762128275286! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From POlsson at enghouse.com Tue Apr 2 11:55:29 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Tue, 2 Apr 2013 07:55:29 +0000 Subject: [Freeswitch-users] Build Error Message-ID: <1FFF97C269757C458224B7C895F35F152593C2@cantor.std.visionutv.se> Ahh - sorry, I read to quickly :) Have you tried doing a "git clean -fdx" before building - it might help. If that doesn't help, please report to Jira. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Alexander Haugg Skickat: den 2 april 2013 09:48 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Build Error No, i'd write "Freeswitch.2012.sln" not 2008 -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Peter Olsson Gesendet: Dienstag, 2. April 2013 09:23 An: 'FreeSWITCH Users Help' Betreff: Re: [Freeswitch-users] Build Error In latest git head this solution file has been renamed to: Freeswitch.2008.unsupported.sln So it sounds like you're not on latest. Also, as the name indicates - VS2008 is not actually supported anymore, so I'm not sure if it's even supposed to work. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Alexander Haugg Skickat: den 2 april 2013 09:01 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Build Error I have some build errors if i try to build the "Freeswitch.2012.sln". My GIT repository is up-to-date. The first error is: 32>------ Rebuild All started: Project: iksemel, Configuration: Debug 32>Win32 ------ 3> dso_lib.c 32> utility.c 32> stream.c 32>..\..\iksemel\src\stream.c(23): fatal error C1083: Cannot open 32>include file: 'openssl/ssl.h': No such file or directory 32> sha.c 32> sax.c But the file exist: freeswitch\libs\openssl-1.0.1c\include\openssl\ssl.h Thanks -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von info at shishir.com.np Gesendet: Dienstag, 2. April 2013 08:14 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Build Error Thanks guys! Error was resolve with the latest changes on the git. It was resolved after latest push, "git clean -fdx", git pull, with bootstrap and ./configure Thanks again guys! On 01.04.2013 21:54, info at shishir.com.np wrote: > I still get the same error after updating too, I even tried on the > fresh git checkout, its same issue. Is it only me getting this error ? > > > On 01.04.2013 15:09, DJB International wrote: >> Ken, >> >> I also experienced the built error today, but not sure whether it's >> related.? http://jira.freeswitch.org/browse/FS-5248 [27] >> >> On Mon, Apr 1, 2013 at 2:54 PM, Ken Rice wrote: >> >>> Update those should be fixed already... >>> >>> Major updates were rolled in this morning... >>> >>> On 4/1/13 4:20 PM, "info at shishir.com.np [1]" wrote: >>> >>> > Hi >>> >>> I Came across the following error while performing make current this >>> > >>> morning. Please suggest. >>> >>> quiet_libtool: compile: ?gcc -DHAVE_CONFIG_H -I. >>> > -I./src/include >>> -I./libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include >>> > >>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >>> > >>> -I/usr/src/freeswitch/libs/libteletone/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -fPIC -fvisibility=hidden >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>> > >>> -Ilibs/sofia-sip/libsofia-sip-ua/sdp >>> -Ilibs/sofia-sip/libsofia-sip-ua/su >>> -g >>> > -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE >>> -D_LARGEFILE64_SOURCE >>> > -I/usr/src/freeswitch/libs/apr/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>> > >>> -I/usr/src/freeswitch/libs/libtpl-1.5/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -I/usr/src/freeswitch/libs/sqlite >>> -I/usr/src/freeswitch/libs/pcre >>> > -I/usr/src/freeswitch/libs/speex/include >>> -Ilibs/speex/include >>> > -I/usr/src/freeswitch/libs/srtp/include >>> > >>> -I/usr/src/freeswitch/libs/srtp/crypto/include >>> -Ilibs/srtp/crypto/include >>> > -I/usr/src/freeswitch/libs/spandsp/src >>> > >>> -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP >>> > >>> -I/usr/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT >>> > >>> -I/usr/src/freeswitch/libs/curl/include >>> -I/usr/src/freeswitch/src/include >>> > -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/libs/libteletone/src >>> > >>> -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden >>> > >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb >>> -DHAVE_OPENSSL -Wall >>> > -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT >>> > >>> libfreeswitch_la-switch_rtp.lo -MD -MP -MF >>> > >>> .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c ?-fPIC >>> -DPIC -o >>> > .libs/libfreeswitch_la-switch_rtp.o >>> cc1: warnings being treated as >>> > errors >>> src/switch_rtp.c: In function ?handle_ice?: >>> src/switch_rtp.c:797: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 8 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?read_rtp_packet?: >>> src/switch_rtp.c:3734: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:3906: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 11 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?rtp_common_write?: >>> src/switch_rtp.c:5289: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:5293: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 9 has type >>> > ?switch_size_t? >>> >>> make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 >>> make[2]: >>> > Leaving directory `/usr/src/freeswitch' >>> make[1]: *** [all] Error 2 >>> make[1]: >>> > Leaving directory `/usr/src/freeswitch' >>> make: *** [current] Error >>> > 2 >>> >>> Thanks >>> >>> __________________________________________________________________ >>> > _______ >>> Professional FreeSWITCH Consulting >>> > Services: >>> consulting at freeswitch.org [3] >>> http://www.freeswitchsolutions.com [4] >>> >>> FreeSW >>> > ITCH-powered IP PBX: The CudaTel Communication Server >>> [5] >>> >>> Official FreeSWITCH >>> > Sites >>> http://www.freeswitch.org [6] >>> http://wiki.freeswitch.org [7] >>> http://www.cluecon [8]. >>> > com >>> >>> FreeSWITCH-users mailing >>> > list >>> FreeSWITCH-users at lists.freeswitch.org [9] >>> http://lists.freeswitch.org/mailman [10] >>> > /listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt [11] >>> > ions/freeswitch-users >>> http://www.freeswitch.org [12] >>> >>> -- >>> Ken >>> http://www.FreeSWITCH.org [13] >>> http://www.ClueCon.com [14] >>> http://www.OSTAG.org [15] >>> irc.freenode.net [16] #freeswitch >>> >>> >> >> >> _____________________________________________________________________ >> ____ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org [17] >>> http://www.freeswitchsolutions.com [18] >>> >>> >>> [19] >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org [20] >>> http://wiki.freeswitch.org [21] >>> http://www.cluecon.com [22] >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org [23] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [24] >>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >>> [25] >>> http://www.freeswitch.org [26] >> >> >> >> Links: >> ------ >> [1] mailto:info at shishir.com.np >> [2] mailto:info at shishir.com.np >> [3] mailto:consulting at freeswitch.org >> [4] http://www.freeswitchsolutions.com >> [5] >> [6] http://www.freeswitch.org >> [7] http://wiki.freeswitch.org >> [8] http://www.cluecon >> [9] mailto:FreeSWITCH-users at lists.freeswitch.org >> [10] http://lists.freeswitch.org/mailman >> [11] http://lists.freeswitch.org/mailman/opt >> [12] http://www.freeswitch.org >> [13] http://www.FreeSWITCH.org >> [14] http://www.ClueCon.com >> [15] http://www.OSTAG.org >> [16] http://irc.freenode.net >> [17] mailto:consulting at freeswitch.org [18] >> http://www.freeswitchsolutions.com >> [19] >> [20] http://www.freeswitch.org >> [21] http://wiki.freeswitch.org >> [22] http://www.cluecon.com >> [23] mailto:FreeSWITCH-users at lists.freeswitch.org >> [24] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [25] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [26] http://www.freeswitch.org >> [27] http://jira.freeswitch.org/browse/FS-5248 >> [28] mailto:krice at 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-use > rs > 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 !DSPAM:515a899132765256711661! From avi at avimarcus.net Tue Apr 2 12:38:46 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 2 Apr 2013 11:38:46 +0300 Subject: [Freeswitch-users] Change order of voicemail message retrival In-Reply-To: References: <5159F9E3.9090204@quentustech.com> <515A03F2.3050506@quentustech.com> Message-ID: If you want to speed things up, apply it manually, test it, and post your results to Jira. -Avi On Tue, Apr 2, 2013 at 10:14 AM, Muhammad Shahzad wrote: > Great. Thanks guys. When can i expect this patch merged in trunk? > > Thank you. > > > On Tue, Apr 2, 2013 at 12:28 AM, Michael Collins wrote: > >> Thanks Kristin! that looks like a simple but effective patch. >> -MC >> >> >> On Mon, Apr 1, 2013 at 3:02 PM, Kristin King < >> kristin.king at quentustech.com> wrote: >> >>> http://jira.freeswitch.org/browse/FS-5249 >>> >>> Kristin King >>> Quentus Technologies, INC >>> 1037 NE 65th St, Ste 273 >>> Seattle, WA 98115 >>> Main: 877-211-9337 >>> Office: 206-388-4778 >>> Fax: 206-462-1861 >>> Cell: 206-755-7329 >>> Email: kristin.king at quentustech.com >>> >>> >>> On 04/01/2013 02:19 PM, Kristin King wrote: >>> > There wasn't an option for this, but I just finished coding one this >>> > morning. I'm filing the feature request and attaching the patch and >>> > hopefully it'll be going in shortly. >>> > >>> > Kristin King >>> > Quentus Technologies, INC >>> > 1037 NE 65th St, Ste 273 >>> > Seattle, WA 98115 >>> > Main: 877-211-9337 >>> > Office: 206-388-4778 >>> > Fax: 206-462-1861 >>> > Cell: 206-755-7329 >>> > Email: kristin.king at quentustech.com >>> > >>> > On 04/01/2013 01:43 PM, Muhammad Shahzad wrote: >>> >> Hi, >>> >> >>> >> Is there any variable / parameter to change the order of messages >>> played >>> >> back while checking voicemail? Currently its FIFO, i am trying to play >>> >> them in LIFO mode. >>> >> >>> >> Thank you. >>> >> >>> >> >>> >> -- >>> >> Mit freundlichen Gr??en >>> >> Muhammad Shahzad >>> >> ----------------------------------- >>> >> CISCO Rich Media Communication Specialist (CRMCS) >>> >> CISCO Certified Network Associate (CCNA) >>> >> Cell: +49 176 99 83 10 85 >>> >> MSN: shari_786pk at hotmail.com >>> >> Email: shaheryarkh at googlemail.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 >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.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 >> >> > > > -- > Mit freundlichen Gr??en > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.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/20130402/c63ef6a4/attachment-0001.html From danb.lists at gmail.com Tue Apr 2 12:43:15 2013 From: danb.lists at gmail.com (DanB) Date: Tue, 02 Apr 2013 10:43:15 +0200 Subject: [Freeswitch-users] [ANN] fsock.go - FreeSWITCH socket client for Go masters Message-ID: <515A9A23.8010209@gmail.com> Hey Guys, We are very excited to announce that fsock.go is now available to be used with MIT license. What is fsock.go: * FreeSWITCH socket client library written exclusively in Go programming language (http://http://golang.org) and freely supported by ITsysCOM. * Designed to be simple: the whole library should never exceed one file. * Unit tests included. * Asynchronous processing (each command runs into it's own goroutine). * Automatic reconnects (given as parameter on instantiation). * Syslogs for transparency. Please feel free to experiment and fork us on github: https://github.com/cgrates/fsock For any additional questions you can use CGRateS mailing list on Google Groups (https://groups.google.com/forum/#!forum/cgrates). DanB From Alexander.Haugg at c4b.de Tue Apr 2 12:54:54 2013 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Tue, 2 Apr 2013 08:54:54 +0000 Subject: [Freeswitch-users] Build Error In-Reply-To: <1FFF97C269757C458224B7C895F35F152593C2@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F152593C2@cantor.std.visionutv.se> Message-ID: No problem ;-). Yes i have, i was doing the same what "info at shishir.com.np" wrote. I had create a jira ticket "FS-5251" /Alex -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Peter Olsson Gesendet: Dienstag, 2. April 2013 09:55 An: 'FreeSWITCH Users Help' Betreff: Re: [Freeswitch-users] Build Error Ahh - sorry, I read to quickly :) Have you tried doing a "git clean -fdx" before building - it might help. If that doesn't help, please report to Jira. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Alexander Haugg Skickat: den 2 april 2013 09:48 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Build Error No, i'd write "Freeswitch.2012.sln" not 2008 -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Peter Olsson Gesendet: Dienstag, 2. April 2013 09:23 An: 'FreeSWITCH Users Help' Betreff: Re: [Freeswitch-users] Build Error In latest git head this solution file has been renamed to: Freeswitch.2008.unsupported.sln So it sounds like you're not on latest. Also, as the name indicates - VS2008 is not actually supported anymore, so I'm not sure if it's even supposed to work. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Alexander Haugg Skickat: den 2 april 2013 09:01 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Build Error I have some build errors if i try to build the "Freeswitch.2012.sln". My GIT repository is up-to-date. The first error is: 32>------ Rebuild All started: Project: iksemel, Configuration: Debug 32>Win32 ------ 3> dso_lib.c 32> utility.c 32> stream.c 32>..\..\iksemel\src\stream.c(23): fatal error C1083: Cannot open 32>include file: 'openssl/ssl.h': No such file or directory 32> sha.c 32> sax.c But the file exist: freeswitch\libs\openssl-1.0.1c\include\openssl\ssl.h Thanks -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von info at shishir.com.np Gesendet: Dienstag, 2. April 2013 08:14 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Build Error Thanks guys! Error was resolve with the latest changes on the git. It was resolved after latest push, "git clean -fdx", git pull, with bootstrap and ./configure Thanks again guys! On 01.04.2013 21:54, info at shishir.com.np wrote: > I still get the same error after updating too, I even tried on the > fresh git checkout, its same issue. Is it only me getting this error ? > > > On 01.04.2013 15:09, DJB International wrote: >> Ken, >> >> I also experienced the built error today, but not sure whether it's >> related.? http://jira.freeswitch.org/browse/FS-5248 [27] >> >> On Mon, Apr 1, 2013 at 2:54 PM, Ken Rice wrote: >> >>> Update those should be fixed already... >>> >>> Major updates were rolled in this morning... >>> >>> On 4/1/13 4:20 PM, "info at shishir.com.np [1]" wrote: >>> >>> > Hi >>> >>> I Came across the following error while performing make current this >>> > >>> morning. Please suggest. >>> >>> quiet_libtool: compile: ?gcc -DHAVE_CONFIG_H -I. >>> > -I./src/include >>> -I./libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include >>> > >>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >>> > >>> -I/usr/src/freeswitch/libs/libteletone/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -fPIC -fvisibility=hidden >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>> > >>> -Ilibs/sofia-sip/libsofia-sip-ua/sdp >>> -Ilibs/sofia-sip/libsofia-sip-ua/su >>> -g >>> > -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE >>> -D_LARGEFILE64_SOURCE >>> > -I/usr/src/freeswitch/libs/apr/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>> > >>> -I/usr/src/freeswitch/libs/libtpl-1.5/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -I/usr/src/freeswitch/libs/sqlite >>> -I/usr/src/freeswitch/libs/pcre >>> > -I/usr/src/freeswitch/libs/speex/include >>> -Ilibs/speex/include >>> > -I/usr/src/freeswitch/libs/srtp/include >>> > >>> -I/usr/src/freeswitch/libs/srtp/crypto/include >>> -Ilibs/srtp/crypto/include >>> > -I/usr/src/freeswitch/libs/spandsp/src >>> > >>> -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP >>> > >>> -I/usr/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT >>> > >>> -I/usr/src/freeswitch/libs/curl/include >>> -I/usr/src/freeswitch/src/include >>> > -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/libs/libteletone/src >>> > >>> -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden >>> > >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb >>> -DHAVE_OPENSSL -Wall >>> > -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT >>> > >>> libfreeswitch_la-switch_rtp.lo -MD -MP -MF >>> > >>> .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c ?-fPIC >>> -DPIC -o >>> > .libs/libfreeswitch_la-switch_rtp.o >>> cc1: warnings being treated as >>> > errors >>> src/switch_rtp.c: In function ?handle_ice?: >>> src/switch_rtp.c:797: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 8 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?read_rtp_packet?: >>> src/switch_rtp.c:3734: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:3906: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 11 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?rtp_common_write?: >>> src/switch_rtp.c:5289: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:5293: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 9 has type >>> > ?switch_size_t? >>> >>> make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 >>> make[2]: >>> > Leaving directory `/usr/src/freeswitch' >>> make[1]: *** [all] Error 2 >>> make[1]: >>> > Leaving directory `/usr/src/freeswitch' >>> make: *** [current] Error >>> > 2 >>> >>> Thanks >>> >>> __________________________________________________________________ >>> > _______ >>> Professional FreeSWITCH Consulting >>> > Services: >>> consulting at freeswitch.org [3] >>> http://www.freeswitchsolutions.com [4] >>> >>> FreeSW >>> > ITCH-powered IP PBX: The CudaTel Communication Server >>> [5] >>> >>> Official FreeSWITCH >>> > Sites >>> http://www.freeswitch.org [6] >>> http://wiki.freeswitch.org [7] >>> http://www.cluecon [8]. >>> > com >>> >>> FreeSWITCH-users mailing >>> > list >>> FreeSWITCH-users at lists.freeswitch.org [9] >>> http://lists.freeswitch.org/mailman [10] >>> > /listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt [11] >>> > ions/freeswitch-users >>> http://www.freeswitch.org [12] >>> >>> -- >>> Ken >>> http://www.FreeSWITCH.org [13] >>> http://www.ClueCon.com [14] >>> http://www.OSTAG.org [15] >>> irc.freenode.net [16] #freeswitch >>> >>> >> >> >> _____________________________________________________________________ >> ____ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org [17] >>> http://www.freeswitchsolutions.com [18] >>> >>> >>> [19] >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org [20] >>> http://wiki.freeswitch.org [21] >>> http://www.cluecon.com [22] >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org [23] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [24] >>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >>> [25] >>> http://www.freeswitch.org [26] >> >> >> >> Links: >> ------ >> [1] mailto:info at shishir.com.np >> [2] mailto:info at shishir.com.np >> [3] mailto:consulting at freeswitch.org >> [4] http://www.freeswitchsolutions.com >> [5] >> [6] http://www.freeswitch.org >> [7] http://wiki.freeswitch.org >> [8] http://www.cluecon >> [9] mailto:FreeSWITCH-users at lists.freeswitch.org >> [10] http://lists.freeswitch.org/mailman >> [11] http://lists.freeswitch.org/mailman/opt >> [12] http://www.freeswitch.org >> [13] http://www.FreeSWITCH.org >> [14] http://www.ClueCon.com >> [15] http://www.OSTAG.org >> [16] http://irc.freenode.net >> [17] mailto:consulting at freeswitch.org [18] >> http://www.freeswitchsolutions.com >> [19] >> [20] http://www.freeswitch.org >> [21] http://wiki.freeswitch.org >> [22] http://www.cluecon.com >> [23] mailto:FreeSWITCH-users at lists.freeswitch.org >> [24] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [25] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [26] http://www.freeswitch.org >> [27] http://jira.freeswitch.org/browse/FS-5248 >> [28] mailto:krice at 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-use > rs > 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 !DSPAM:515a899132765256711661! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From bpriddy at bryantschools.org Tue Apr 2 17:21:21 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Tue, 2 Apr 2013 08:21:21 -0500 Subject: [Freeswitch-users] Weirdness Message-ID: Why would everyone of my extensions be able to call a certain extension number such as (3003) but when a particular user calls that number FS tells that user (3006) that 3003 is busy? 2013-04-02 08:18:56.104087 [DEBUG] sofia.c:5603 Channel sofia/internal/ sip:3003 at 10.25.190.8:1024 entering state [calling][0] 2013-04-02 08:18:56.254094 [DEBUG] sofia.c:5603 Channel sofia/internal/ sip:3003 at 10.25.190.8:1024 entering state [terminated][486] 2013-04-02 08:18:56.254094 [DEBUG] switch_channel.c:2994 (sofia/internal/ sip:3003 at 10.25.190.8:1024) Callstate Change RINGING -> HANGUP 2013-04-02 08:18:56.254094 [NOTICE] sofia.c:6387 Hangup sofia/internal/ sip:3003 at 10.25.190.8:1024 [CS_CONSUME_MEDIA] [USER_BUSY] 2013-04-02 08:18:56.254094 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:3003 at 10.25.190.8:1024) Running State Change CS_HANGUP 2013-04-02 08:18:56.254094 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/sip:3003 at 10.25.190.8:1024 [KILL] 2013-04-02 08:18:56.254094 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:3003 at 10.25.190.8:1024 [BREAK] 2013-04-02 08:18:56.254094 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:3003 at 10.25.190.8:1024) State HANGUP 2013-04-02 08:18:56.254094 [DEBUG] mod_sofia.c:503 Channel sofia/internal/ sip:3003 at 10.25.190.8:1024 hanging up, cause: USER_BUSY 2013-04-02 08:18:56.274104 [DEBUG] switch_ivr_originate.c:3531 Originate Resulted in Error Cause: 17 [USER_BUSY] 2013-04-02 08:18:56.274104 [NOTICE] switch_ivr_originate.c:2608 Cannot create outgoing channel of type [user] cause: [USER_BUSY] 2013-04-02 08:18:56.274104 [DEBUG] switch_ivr_originate.c:3531 Originate Resulted in Error Cause: 17 [USER_BUSY] 2013-04-02 08:18:56.274104 [INFO] mod_dptools.c:3052 Originate Failed. Cause: USER_BUSY EXECUTE sofia/internal/3006 at pbx.bryantschools.org answer() -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130402/5182e738/attachment-0001.html From shaheryarkh at gmail.com Tue Apr 2 17:35:56 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Tue, 2 Apr 2013 14:35:56 +0100 Subject: [Freeswitch-users] Change order of voicemail message retrival In-Reply-To: References: <5159F9E3.9090204@quentustech.com> <515A03F2.3050506@quentustech.com> Message-ID: ok, i will do it tonight and update jira. Thanks again for your prompt help. Thank you. On Tue, Apr 2, 2013 at 9:38 AM, Avi Marcus wrote: > If you want to speed things up, apply it manually, test it, and post your > results to Jira. > > -Avi > > > On Tue, Apr 2, 2013 at 10:14 AM, Muhammad Shahzad wrote: > >> Great. Thanks guys. When can i expect this patch merged in trunk? >> >> Thank you. >> >> >> On Tue, Apr 2, 2013 at 12:28 AM, Michael Collins wrote: >> >>> Thanks Kristin! that looks like a simple but effective patch. >>> -MC >>> >>> >>> On Mon, Apr 1, 2013 at 3:02 PM, Kristin King < >>> kristin.king at quentustech.com> wrote: >>> >>>> http://jira.freeswitch.org/browse/FS-5249 >>>> >>>> Kristin King >>>> Quentus Technologies, INC >>>> 1037 NE 65th St, Ste 273 >>>> Seattle, WA 98115 >>>> Main: 877-211-9337 >>>> Office: 206-388-4778 >>>> Fax: 206-462-1861 >>>> Cell: 206-755-7329 >>>> Email: kristin.king at quentustech.com >>>> >>>> >>>> On 04/01/2013 02:19 PM, Kristin King wrote: >>>> > There wasn't an option for this, but I just finished coding one this >>>> > morning. I'm filing the feature request and attaching the patch and >>>> > hopefully it'll be going in shortly. >>>> > >>>> > Kristin King >>>> > Quentus Technologies, INC >>>> > 1037 NE 65th St, Ste 273 >>>> > Seattle, WA 98115 >>>> > Main: 877-211-9337 >>>> > Office: 206-388-4778 >>>> > Fax: 206-462-1861 >>>> > Cell: 206-755-7329 >>>> > Email: kristin.king at quentustech.com >>>> > >>>> > On 04/01/2013 01:43 PM, Muhammad Shahzad wrote: >>>> >> Hi, >>>> >> >>>> >> Is there any variable / parameter to change the order of messages >>>> played >>>> >> back while checking voicemail? Currently its FIFO, i am trying to >>>> play >>>> >> them in LIFO mode. >>>> >> >>>> >> Thank you. >>>> >> >>>> >> >>>> >> -- >>>> >> Mit freundlichen Gr??en >>>> >> Muhammad Shahzad >>>> >> ----------------------------------- >>>> >> CISCO Rich Media Communication Specialist (CRMCS) >>>> >> CISCO Certified Network Associate (CCNA) >>>> >> Cell: +49 176 99 83 10 85 >>>> >> MSN: shari_786pk at hotmail.com >>>> >> Email: shaheryarkh at googlemail.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 >>>> >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.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 >>> >>> >> >> >> -- >> Mit freundlichen Gr??en >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +49 176 99 83 10 85 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.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 > > -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130402/c3359879/attachment.html From trever.adams at gmail.com Tue Apr 2 18:47:49 2013 From: trever.adams at gmail.com (Trever L. Adams) Date: Tue, 02 Apr 2013 08:47:49 -0600 Subject: [Freeswitch-users] Transfer leg B of call when hung up by application (such as rxfax) Message-ID: <515AEF95.9090700@gmail.com> Hello everyone, I am still working on getting my parents to a place where they can ditch their fax machine. I have rxfax working in an autodetect way. It works great. The problem I am having, is I need to announce to them that they can hang up once rxfax answers. I am using a python script to call rxfax via session.execute. I do have some other things I am working on, so it would be great to see how to do it with python or lua and with dialplan. execute_on_answer or exec_after_bridge_* seem like they would, but I am betting they work on the a leg of the call, not the b leg. intercept_un*_* may also be interesting, but again, it looks like a leg. Well, as I write this, I am reading about intercept. I can grab the bleg, but not sure how to trigger this and to send this to an extension to play the audio. Any help would be greatly appreciated. Trever -- "The world is full of people who have never, since childhood, met an open doorway with an open mind." -- E.B. White -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130402/fe2daec4/attachment-0001.bin From henry.houfeng at gmail.com Tue Apr 2 18:06:36 2013 From: henry.houfeng at gmail.com (henry.houfeng) Date: Tue, 2 Apr 2013 07:06:36 -0700 (PDT) Subject: [Freeswitch-users] start_tone_detect, but no event is fired Message-ID: <1364911596726-7589332.post@n2.nabble.com> Hi, I want to detect call progress tone with start_tone_detect, but I can't get the event fired. My dialplan is: The call senario is: Caller A is a user registered on local FS. Callee B is a user on remote gateway. A-------INVITE------------>B A<------100 Trying---------B A<-----183 with SDP-------B A<------RTP of Ring Tone--B The early media of ring tone or busy tone is transferred from B to A after 183. But I can't see the DETECTED_TONE is fired. I have set loglevel to 3 in spandsp.conf.xml: I can only find some Tone segment at a frequencey less than 10: freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:12.618263 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = 4, f2 = 5, duration = 48 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:12.738255 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = -1, f2 = -1, duration = 128 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:12.858249 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = 4, f2 = 5, duration = 112 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:12.958243 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = -1, f2 = -1, duration = 112 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:13.018238 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = 4, f2 = 5, duration = 48 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:13.058236 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = -1, f2 = -1, duration = 48 Is my dialplan incorrect? I will very appreciate your suggestion. Thanks! Regards, Henry -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/start-tone-detect-but-no-event-is-fired-tp7589332.html Sent from the freeswitch-users mailing list archive at Nabble.com. From marketing at cluecon.com Tue Apr 2 18:44:55 2013 From: marketing at cluecon.com (Michael Collins) Date: Tue, 2 Apr 2013 07:44:55 -0700 Subject: [Freeswitch-users] ClueCon 2013 - Call For Speakers Message-ID: [image: https://mail-attachment.googleusercontent.com/attachment/u/0/?ui=2&ik=43fec29535&view=att&th=13cf45a70d222cc4&attid=0.1&disp=inline&realattid=f_hddkcyjh0&safe=1&zw&saduie=AG9B_P-p4MyvJI_ZqTegUZpanklK&sadet=1361819341430&sads=tdlIrK0Sx3z615AVwvfWzTRj7Bw] ClueCon - the open source IP communications conference by developers, fordevelopers - would like to announce that we are having an open call for speaking proposals for this year's event. If you have an idea fora technical presentation for ClueCon 2013 then we would like to hear about it. What makes a great ClueCon presentation? The tech savvy crowd that attends ClueCon *loves *technical presentations. In general, the more technical the presentation, the better. If you are thinking about a presentation then consider these points: - ClueCon talks are 30 minutes in length, including Q&A time with the audience - ClueCon has a special focus on open source VoIP and telephony projects like FreeSWITCH, Asterisk, OpenSIPS, and Kamailio - Attendees enjoy hearing about projects built with open source tools, especially those in a production environment - Highly technical discussions that show the nuts and bolts are especially well-liked - The audience appreciates seeing and participating in live demonstrations - We are especially interested in WebRTC-related talks and demonstrations Please send your proposals to marketing at cluecon.com. Be sure to include the following items: - Working title - Brief description of the talk (abstract) - Name of the presenter Don't delay! There are a limited number of openings. We will contact you as soon as your talk has been approved and will inform you of the scheduled time. ClueCon 2013 Registration Information ClueCon 2013 registration is now open!. Visit the registration page for details. Be sure to book your room at the Hyatt Chicago Magnificent Mileand qualify for the $300 discount. As always, feel free to call us at 877.742.CLUE (877.742.2583) if you have any questions about ClueCon 2013. Also, keep in mind that the FreeSWITCH community has a conference calleach Wednesday at 1PM Eastern time. This is a great opportunity to talk about open source telephony and get to know a number folks who will be at ClueCon 2013. Stay tuned for more news about ClueCon speakers, sponsors, and related events! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130402/0a5faa63/attachment.html From fdelawarde at wirelessmundi.com Tue Apr 2 19:40:33 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?=) Date: Tue, 02 Apr 2013 17:40:33 +0200 Subject: [Freeswitch-users] B64 audio codec? Message-ID: <1364917233.10160.90.camel@luna.madrid.commsmundi.com> Hi all, Anyone has some info about the recently commited mod_b64 codec? The source refers to it as "The B64 ultra-low delay audio codec", and links to a webpage www.b64-codec.org which does not exist! Anyway, it sounds interesting and doesn't seem to be an april fools joke! Thanks, Fran?ois. From steveu at coppice.org Tue Apr 2 20:11:54 2013 From: steveu at coppice.org (Steve Underwood) Date: Wed, 03 Apr 2013 00:11:54 +0800 Subject: [Freeswitch-users] B64 audio codec? In-Reply-To: <1364917233.10160.90.camel@luna.madrid.commsmundi.com> References: <1364917233.10160.90.camel@luna.madrid.commsmundi.com> Message-ID: <515B034A.3010204@coppice.org> On 04/02/2013 11:40 PM, Fran?ois wrote: > Hi all, > > Anyone has some info about the recently commited mod_b64 codec? > > The source refers to it as "The B64 ultra-low delay audio codec", and > links to a webpage www.b64-codec.org which does not exist! > > Anyway, it sounds interesting and doesn't seem to be an april fools > joke! > When did base 64 stop being the basis of April 1st jokes? Steve From jleung at v10networks.ca Tue Apr 2 20:56:44 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Tue, 2 Apr 2013 09:56:44 -0700 Subject: [Freeswitch-users] B64 audio codec? In-Reply-To: <1364917233.10160.90.camel@luna.madrid.commsmundi.com> References: <1364917233.10160.90.camel@luna.madrid.commsmundi.com> Message-ID: <003801ce2fc3$0f3a99f0$2dafcdd0$@v10networks.ca> Personally I still have a hunch that mod_b64 is a subtle April Fool's Day joke. Even though it's a joke, it's still a very functional codec. ;) > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Fran?ois > Sent: Tuesday, April 2, 2013 8:41 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] B64 audio codec? > > Hi all, > > Anyone has some info about the recently commited mod_b64 codec? > > The source refers to it as "The B64 ultra-low delay audio codec", and links to a > webpage www.b64-codec.org which does not exist! > > Anyway, it sounds interesting and doesn't seem to be an april fools joke! > > Thanks, > Fran?ois. > > > __________________________________________________________ > _______________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveayre at gmail.com Tue Apr 2 21:12:49 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 2 Apr 2013 18:12:49 +0100 Subject: [Freeswitch-users] RFC4904 Trunkgroup Identity In-Reply-To: References: Message-ID: Try checking for the required information in the info app output or XML CDR. If it's present in there, then you should be able to examine it from the dialplan and act upon it. -Steve On 1 April 2013 22:49, Michael Collins wrote: > I don't believe we support this as such. However, I've never seen a > situation where FS can't route a call based on available criteria. Are > these trunkgroup values in specific headers? If so it's possible to extract > that information and use it for routing in the dialplan. > > If you have some call examples that you can put on pastebin.freeswitch.orgthat would be helpful. > > Thanks, > MC > > On Sun, Mar 31, 2013 at 7:17 PM, Kevin Kennedy wrote: > >> I am new to freeswitch as far as testing it, but have been on the >> user-list for a long time. I have searched through my archive of emails as >> well as searched on google for any answer that I can find on this. I am >> looking for the configuration options for RFC4904, trunk-group identity. >> This is where you can set Freeswitch up to send unscreened calls through >> your trunkgroup using the tgrp and trunk-context that is in the >> registration for every outbound call. Any help would be 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.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/20130402/526d960a/attachment-0001.html From steveayre at gmail.com Tue Apr 2 22:03:12 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 2 Apr 2013 19:03:12 +0100 Subject: [Freeswitch-users] Weirdness In-Reply-To: References: Message-ID: Unless you're using Limit the phone itself is probably sending the SIP response 486 User Busy On 2 April 2013 14:21, Blake Priddy wrote: > Why would everyone of my extensions be able to call a certain extension > number such as (3003) but when a particular user calls that number FS tells > that user (3006) that 3003 is busy? > > 2013-04-02 08:18:56.104087 [DEBUG] sofia.c:5603 Channel sofia/internal/ > sip:3003 at 10.25.190.8:1024 entering state [calling][0] > 2013-04-02 08:18:56.254094 [DEBUG] sofia.c:5603 Channel sofia/internal/ > sip:3003 at 10.25.190.8:1024 entering state [terminated][486] > 2013-04-02 08:18:56.254094 [DEBUG] switch_channel.c:2994 (sofia/internal/ > sip:3003 at 10.25.190.8:1024) Callstate Change RINGING -> HANGUP > 2013-04-02 08:18:56.254094 [NOTICE] sofia.c:6387 Hangup sofia/internal/ > sip:3003 at 10.25.190.8:1024 [CS_CONSUME_MEDIA] [USER_BUSY] > 2013-04-02 08:18:56.254094 [DEBUG] switch_core_state_machine.c:415 > (sofia/internal/sip:3003 at 10.25.190.8:1024) Running State Change CS_HANGUP > 2013-04-02 08:18:56.254094 [DEBUG] switch_channel.c:3017 Send signal > sofia/internal/sip:3003 at 10.25.190.8:1024 [KILL] > 2013-04-02 08:18:56.254094 [DEBUG] switch_core_session.c:1283 Send signal > sofia/internal/sip:3003 at 10.25.190.8:1024 [BREAK] > 2013-04-02 08:18:56.254094 [DEBUG] switch_core_state_machine.c:667 > (sofia/internal/sip:3003 at 10.25.190.8:1024) State HANGUP > 2013-04-02 08:18:56.254094 [DEBUG] mod_sofia.c:503 Channel sofia/internal/ > sip:3003 at 10.25.190.8:1024 hanging up, cause: USER_BUSY > 2013-04-02 08:18:56.274104 [DEBUG] switch_ivr_originate.c:3531 Originate > Resulted in Error Cause: 17 [USER_BUSY] > 2013-04-02 08:18:56.274104 [NOTICE] switch_ivr_originate.c:2608 Cannot > create outgoing channel of type [user] cause: [USER_BUSY] > 2013-04-02 08:18:56.274104 [DEBUG] switch_ivr_originate.c:3531 Originate > Resulted in Error Cause: 17 [USER_BUSY] > 2013-04-02 08:18:56.274104 [INFO] mod_dptools.c:3052 Originate Failed. > Cause: USER_BUSY > EXECUTE sofia/internal/3006 at pbx.bryantschools.org answer() > > > -- > > *Blakelund Priddy* > Network & Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130402/48473415/attachment.html From william.king at quentustech.com Tue Apr 2 22:11:03 2013 From: william.king at quentustech.com (William King) Date: Tue, 02 Apr 2013 11:11:03 -0700 Subject: [Freeswitch-users] Weirdness In-Reply-To: References: Message-ID: <515B1F37.6060702@quentustech.com> Blake, If you turn on a sip trace with "sofia profile internal siptrace on" you should be able to see why FS is responding to the caller with USER_BUSY. Usually I see this because someone accidentally hit DoNotDisturb, or they have a misconfiguration in the call forwarding feature on their phone. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 04/02/2013 06:21 AM, Blake Priddy wrote: > Why would everyone of my extensions be able to call a certain extension > number such as (3003) but when a particular user calls that number FS > tells that user (3006) that 3003 is busy? > > 2013-04-02 08:18:56.104087 [DEBUG] sofia.c:5603 Channel > sofia/internal/sip:3003 at 10.25.190.8:1024 > entering state [calling][0] > 2013-04-02 08:18:56.254094 [DEBUG] sofia.c:5603 Channel > sofia/internal/sip:3003 at 10.25.190.8:1024 > entering state [terminated][486] > 2013-04-02 08:18:56.254094 [DEBUG] switch_channel.c:2994 > (sofia/internal/sip:3003 at 10.25.190.8:1024 > ) Callstate Change RINGING -> HANGUP > 2013-04-02 08:18:56.254094 [NOTICE] sofia.c:6387 Hangup > sofia/internal/sip:3003 at 10.25.190.8:1024 > [CS_CONSUME_MEDIA] [USER_BUSY] > 2013-04-02 08:18:56.254094 [DEBUG] switch_core_state_machine.c:415 > (sofia/internal/sip:3003 at 10.25.190.8:1024 > ) Running State Change CS_HANGUP > 2013-04-02 08:18:56.254094 [DEBUG] switch_channel.c:3017 Send signal > sofia/internal/sip:3003 at 10.25.190.8:1024 > [KILL] > 2013-04-02 08:18:56.254094 [DEBUG] switch_core_session.c:1283 Send > signal sofia/internal/sip:3003 at 10.25.190.8:1024 > [BREAK] > 2013-04-02 08:18:56.254094 [DEBUG] switch_core_state_machine.c:667 > (sofia/internal/sip:3003 at 10.25.190.8:1024 > ) State HANGUP > 2013-04-02 08:18:56.254094 [DEBUG] mod_sofia.c:503 Channel > sofia/internal/sip:3003 at 10.25.190.8:1024 > hanging up, cause: USER_BUSY > 2013-04-02 08:18:56.274104 [DEBUG] switch_ivr_originate.c:3531 Originate > Resulted in Error Cause: 17 [USER_BUSY] > 2013-04-02 08:18:56.274104 [NOTICE] switch_ivr_originate.c:2608 Cannot > create outgoing channel of type [user] cause: [USER_BUSY] > 2013-04-02 08:18:56.274104 [DEBUG] switch_ivr_originate.c:3531 Originate > Resulted in Error Cause: 17 [USER_BUSY] > 2013-04-02 08:18:56.274104 [INFO] mod_dptools.c:3052 Originate Failed. > Cause: USER_BUSY > EXECUTE sofia/internal/3006 at pbx.bryantschools.org > answer() > > > -- > > *Blakelund Priddy* > Network & Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bpriddy at bryantschools.org Tue Apr 2 22:21:17 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Tue, 2 Apr 2013 13:21:17 -0500 Subject: [Freeswitch-users] Weirdness In-Reply-To: <515B1F37.6060702@quentustech.com> References: <515B1F37.6060702@quentustech.com> Message-ID: Thank you all! On Tue, Apr 2, 2013 at 1:11 PM, William King wrote: > Blake, > > If you turn on a sip trace with "sofia profile internal siptrace on" you > should be able to see why FS is responding to the caller with USER_BUSY. > Usually I see this because someone accidentally hit DoNotDisturb, or > they have a misconfiguration in the call forwarding feature on their phone. > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 04/02/2013 06:21 AM, Blake Priddy wrote: > > Why would everyone of my extensions be able to call a certain extension > > number such as (3003) but when a particular user calls that number FS > > tells that user (3006) that 3003 is busy? > > > > 2013-04-02 08:18:56.104087 [DEBUG] sofia.c:5603 Channel > > sofia/internal/sip:3003 at 10.25.190.8:1024 > > entering state [calling][0] > > 2013-04-02 08:18:56.254094 [DEBUG] sofia.c:5603 Channel > > sofia/internal/sip:3003 at 10.25.190.8:1024 > > entering state [terminated][486] > > 2013-04-02 08:18:56.254094 [DEBUG] switch_channel.c:2994 > > (sofia/internal/sip:3003 at 10.25.190.8:1024 > > ) Callstate Change RINGING -> HANGUP > > 2013-04-02 08:18:56.254094 [NOTICE] sofia.c:6387 Hangup > > sofia/internal/sip:3003 at 10.25.190.8:1024 > > [CS_CONSUME_MEDIA] [USER_BUSY] > > 2013-04-02 08:18:56.254094 [DEBUG] switch_core_state_machine.c:415 > > (sofia/internal/sip:3003 at 10.25.190.8:1024 > > ) Running State Change CS_HANGUP > > 2013-04-02 08:18:56.254094 [DEBUG] switch_channel.c:3017 Send signal > > sofia/internal/sip:3003 at 10.25.190.8:1024 > > [KILL] > > 2013-04-02 08:18:56.254094 [DEBUG] switch_core_session.c:1283 Send > > signal sofia/internal/sip:3003 at 10.25.190.8:1024 > > [BREAK] > > 2013-04-02 08:18:56.254094 [DEBUG] switch_core_state_machine.c:667 > > (sofia/internal/sip:3003 at 10.25.190.8:1024 > > ) State HANGUP > > 2013-04-02 08:18:56.254094 [DEBUG] mod_sofia.c:503 Channel > > sofia/internal/sip:3003 at 10.25.190.8:1024 > > hanging up, cause: USER_BUSY > > 2013-04-02 08:18:56.274104 [DEBUG] switch_ivr_originate.c:3531 Originate > > Resulted in Error Cause: 17 [USER_BUSY] > > 2013-04-02 08:18:56.274104 [NOTICE] switch_ivr_originate.c:2608 Cannot > > create outgoing channel of type [user] cause: [USER_BUSY] > > 2013-04-02 08:18:56.274104 [DEBUG] switch_ivr_originate.c:3531 Originate > > Resulted in Error Cause: 17 [USER_BUSY] > > 2013-04-02 08:18:56.274104 [INFO] mod_dptools.c:3052 Originate Failed. > > Cause: USER_BUSY > > EXECUTE sofia/internal/3006 at pbx.bryantschools.org > > answer() > > > > > > -- > > > > *Blakelund Priddy* > > Network & Systems Engineer > > Bryant Public School District > > Bryant, Arkansas 72022 > > http://www.bryantschools.org > > p 501-653-5038 > > f 501-847-5656 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130402/67b7362a/attachment-0001.html From mike at jerris.com Tue Apr 2 22:23:06 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 2 Apr 2013 12:23:06 -0600 Subject: [Freeswitch-users] B64 audio codec? In-Reply-To: <003801ce2fc3$0f3a99f0$2dafcdd0$@v10networks.ca> References: <1364917233.10160.90.camel@luna.madrid.commsmundi.com> <003801ce2fc3$0f3a99f0$2dafcdd0$@v10networks.ca> Message-ID: <7847442903959552389@unknownmsgid> Good jokes are functional. On Apr 2, 2013, at 12:00 PM, Jeff Leung wrote: > Personally I still have a hunch that mod_b64 is a subtle April Fool's Day joke. Even though it's a joke, it's still a very functional codec. ;) > >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- >> users-bounces at lists.freeswitch.org] On Behalf Of Fran?ois >> Sent: Tuesday, April 2, 2013 8:41 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] B64 audio codec? >> >> Hi all, >> >> Anyone has some info about the recently commited mod_b64 codec? >> >> The source refers to it as "The B64 ultra-low delay audio codec", and links to a >> webpage www.b64-codec.org which does not exist! >> >> Anyway, it sounds interesting and doesn't seem to be an april fools joke! >> >> Thanks, >> Fran?ois. >> >> >> __________________________________________________________ >> _______________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 drk at drkngs.net Tue Apr 2 22:33:48 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Tue, 02 Apr 2013 11:33:48 -0700 Subject: [Freeswitch-users] Build Error In-Reply-To: Message-ID: <20130402183348.0aedd30c@mail.tritonwest.net> It looks like it's gonna be some amount of work to get it to build on windows again... --Dave _____ From: Alexander Haugg [mailto:Alexander.Haugg at c4b.de] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 02 Apr 2013 00:01:04 -0700 Subject: Re: [Freeswitch-users] Build Error I have some build errors if i try to build the "Freeswitch.2012.sln". My GIT repository is up-to-date. The first error is: 32>------ Rebuild All started: Project: iksemel, Configuration: Debug Win32 ------ 3> dso_lib.c 32> utility.c 32> stream.c 32>..\..\iksemel\src\stream.c(23): fatal error C1083: Cannot open include file: 'openssl/ssl.h': No such file or directory 32> sha.c 32> sax.c But the file exist: freeswitch\libs\openssl-1.0.1c\include\openssl\ssl.h Thanks -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von info at shishir.com.np Gesendet: Dienstag, 2. April 2013 08:14 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Build Error Thanks guys! Error was resolve with the latest changes on the git. It was resolved after latest push, "git clean -fdx", git pull, with bootstrap and ./configure Thanks again guys! On 01.04.2013 21:54, info at shishir.com.np wrote: > I still get the same error after updating too, I even tried on the > fresh git checkout, its same issue. Is it only me getting this error ? > > > On 01.04.2013 15:09, DJB International wrote: >> Ken, >> >> I also experienced the built error today, but not sure whether it's >> related. http://jira.freeswitch.org/browse/FS-5248 [27] >> >> On Mon, Apr 1, 2013 at 2:54 PM, Ken Rice wrote: >> >>> Update those should be fixed already... >>> >>> Major updates were rolled in this morning... >>> >>> On 4/1/13 4:20 PM, "info at shishir.com.np [1]" wrote: >>> >>> > Hi >>> >>> I Came across the following error while performing make current >>> this >>> > >>> morning. Please suggest. >>> >>> quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. >>> > -I./src/include >>> -I./libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include >>> > >>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >>> > >>> -I/usr/src/freeswitch/libs/libteletone/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -fPIC -fvisibility=hidden >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>> > >>> -Ilibs/sofia-sip/libsofia-sip-ua/sdp >>> -Ilibs/sofia-sip/libsofia-sip-ua/su >>> -g >>> > -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE >>> -D_LARGEFILE64_SOURCE >>> > -I/usr/src/freeswitch/libs/apr/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>> > >>> -I/usr/src/freeswitch/libs/libtpl-1.5/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -I/usr/src/freeswitch/libs/sqlite >>> -I/usr/src/freeswitch/libs/pcre >>> > -I/usr/src/freeswitch/libs/speex/include >>> -Ilibs/speex/include >>> > -I/usr/src/freeswitch/libs/srtp/include >>> > >>> -I/usr/src/freeswitch/libs/srtp/crypto/include >>> -Ilibs/srtp/crypto/include >>> > -I/usr/src/freeswitch/libs/spandsp/src >>> > >>> -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP >>> > >>> -I/usr/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT >>> > >>> -I/usr/src/freeswitch/libs/curl/include >>> -I/usr/src/freeswitch/src/include >>> > -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/libs/libteletone/src >>> > >>> -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden >>> > >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb >>> -DHAVE_OPENSSL >>> -Wall >>> > -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT >>> > >>> libfreeswitch_la-switch_rtp.lo -MD -MP -MF >>> > >>> .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c -fPIC >>> -DPIC >>> -o >>> > .libs/libfreeswitch_la-switch_rtp.o >>> cc1: warnings being treated as >>> > errors >>> src/switch_rtp.c: In function ?handle_ice?: >>> src/switch_rtp.c:797: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 8 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?read_rtp_packet?: >>> src/switch_rtp.c:3734: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:3906: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 11 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?rtp_common_write?: >>> src/switch_rtp.c:5289: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:5293: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 9 has type >>> > ?switch_size_t? >>> >>> make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 >>> make[2]: >>> > Leaving directory `/usr/src/freeswitch' >>> make[1]: *** [all] Error 2 >>> make[1]: >>> > Leaving directory `/usr/src/freeswitch' >>> make: *** [current] Error >>> > 2 >>> >>> Thanks >>> >>> __________________________________________________________________ >>> > _______ >>> Professional FreeSWITCH Consulting >>> > Services: >>> consulting at freeswitch.org [3] >>> http://www.freeswitchsolutions.com [4] >>> >>> FreeSW >>> > ITCH-powered IP PBX: The CudaTel Communication >>> > Server >>> [5] >>> >>> Official FreeSWITCH >>> > Sites >>> http://www.freeswitch.org [6] >>> http://wiki.freeswitch.org [7] >>> http://www.cluecon [8]. >>> > com >>> >>> FreeSWITCH-users mailing >>> > list >>> FreeSWITCH-users at lists.freeswitch.org [9] >>> http://lists.freeswitch.org/mailman [10] >>> > /listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt [11] >>> > ions/freeswitch-users >>> http://www.freeswitch.org [12] >>> >>> -- >>> Ken >>> http://www.FreeSWITCH.org [13] >>> http://www.ClueCon.com [14] >>> http://www.OSTAG.org [15] >>> irc.freenode.net [16] #freeswitch >>> >>> >> >> >> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org [17] >>> http://www.freeswitchsolutions.com [18] >>> >>> >>> [19] >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org [20] >>> http://wiki.freeswitch.org [21] >>> http://www.cluecon.com [22] >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org [23] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [24] >>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> [25] >>> http://www.freeswitch.org [26] >> >> >> >> Links: >> ------ >> [1] mailto:info at shishir.com.np >> [2] mailto:info at shishir.com.np >> [3] mailto:consulting at freeswitch.org >> [4] http://www.freeswitchsolutions.com >> [5] >> [6] http://www.freeswitch.org >> [7] http://wiki.freeswitch.org >> [8] http://www.cluecon >> [9] mailto:FreeSWITCH-users at lists.freeswitch.org >> [10] http://lists.freeswitch.org/mailman >> [11] http://lists.freeswitch.org/mailman/opt >> [12] http://www.freeswitch.org >> [13] http://www.FreeSWITCH.org >> [14] http://www.ClueCon.com >> [15] http://www.OSTAG.org >> [16] http://irc.freenode.net >> [17] mailto:consulting at freeswitch.org >> [18] http://www.freeswitchsolutions.com >> [19] >> [20] http://www.freeswitch.org >> [21] http://wiki.freeswitch.org >> [22] http://www.cluecon.com >> [23] mailto:FreeSWITCH-users at lists.freeswitch.org >> [24] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [25] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [26] http://www.freeswitch.org >> [27] http://jira.freeswitch.org/browse/FS-5248 >> [28] mailto:krice at 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/20130402/bf231f5f/attachment-0001.html From scott at infobunker.com Tue Apr 2 21:06:29 2013 From: scott at infobunker.com (Scott Ocken) Date: Tue, 02 Apr 2013 12:06:29 -0500 Subject: [Freeswitch-users] Yum RPM Repo Metadat Corrupt Message-ID: <20130402120629.48067gkrrp18ri0w@mail.infobunker.com> It looks like the Yum RPM Repo for Free Switch is corrupt. I am getting: http://files.freeswitch.org/yum/6/x86_64/repodata/filelists.xml.gz: [Errno -1] Metadata file does not match checksum Trying other mirror. http://files.freeswitch.org/yum/6/x86_64/repodata/primary.xml.gz: [Errno 14] Downloaded more than max size for http://files.freeswitch.org/yum/6/x86_64/repodata/primary.xml.gz: 101360 > 91756 Trying other mirror. Error: failure: repodata/primary.xml.gz from freeswitch: [Errno 256] No more mirrors to try. Thanks Scott -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-keys Size: 1324 bytes Desc: PGP Public Key Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130402/9cf07952/attachment.bin From fs at tcowan.net Tue Apr 2 21:28:19 2013 From: fs at tcowan.net (fs at tcowan.net) Date: Tue, 02 Apr 2013 13:28:19 -0400 Subject: [Freeswitch-users] =?utf-8?q?record=5Fsession_wav_files_are_corru?= =?utf-8?q?pted?= Message-ID: After making a recording with record_session the wav files are corrupted and wont open. Audacity will say the wav is corrupt and itunes and windows media player wont play them at all. If I try to import the audio as raw format in audacity you can hear some high pitched talking that plays back very quickly. I have no clue how to diagnose what is wrong or where to even begin diagnosing why the recording isnt working right. Is there anything that I could of missed with the setup that would cause it not to record properly? My outbound route looks like this: From schoch+freeswitch.org at xwin32.com Wed Apr 3 01:46:17 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 2 Apr 2013 14:46:17 -0700 Subject: [Freeswitch-users] record_session wav files are corrupted In-Reply-To: References: Message-ID: On Tue, Apr 2, 2013 at 10:28 AM, wrote: > If I try to import the > audio as raw format in audacity you can hear some high pitched talking > that plays back very quickly. > When you import as raw data, try changing the sample rate to 8000. The reason Audacity can't open the file is because the header, which specifies the encoding, channels, sample rate, etc. is damaged or missing. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130402/cd78555f/attachment.html From clive at lansink.co.nz Wed Apr 3 03:12:07 2013 From: clive at lansink.co.nz (Clive Lansink) Date: Wed, 3 Apr 2013 12:12:07 +1300 Subject: [Freeswitch-users] Recording incoming calls Message-ID: <20130402231339.CEF21DA0F9@jlo.kiwilink.co.nz> An embedded and charset-unspecified text was scrubbed... Name: not available Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/b17be5b7/attachment.pl From shaharhd at gmail.com Wed Apr 3 03:48:14 2013 From: shaharhd at gmail.com (Shahar) Date: Tue, 2 Apr 2013 19:48:14 -0400 Subject: [Freeswitch-users] Spandsp modems as non-root? In-Reply-To: <50703D76.7010301@integrafin.co.uk> References: <50703D76.7010301@integrafin.co.uk> Message-ID: Alex, I've added another section exactly on this subject. you might want to have a look at http://wiki.freeswitch.org/wiki/HylaFax#Modem_devices_permissions_issue On Sat, Oct 6, 2012 at 10:17 AM, Alex Crow wrote: > Hi, > > Has anyone figured out how to get the spandsp emulated modems running > when Freeswitch is running as a non-root user yet? > > I don't even get the /dev/pts/pts* devices added either on debian or > Ubuntu. > > Thanks > > Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130402/7aa67cc8/attachment.html From itsusama at gmail.com Wed Apr 3 07:14:14 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Wed, 3 Apr 2013 08:14:14 +0500 Subject: [Freeswitch-users] Mod Managed Conf Call Message-ID: <02d201ce3019$55af71c0$010e5540$@gmail.com> Hey, Any updates on when the conference call is going to be scheduled for? The thread kinda died. Regards. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of freeswitch-users-request at lists.freeswitch.org Sent: Tuesday, April 02, 2013 11:34 PM To: freeswitch-users at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 82, Issue 19 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." From krice at freeswitch.org Wed Apr 3 07:25:25 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 2 Apr 2013 22:25:25 -0500 Subject: [Freeswitch-users] Mod Managed Conf Call In-Reply-To: <02d201ce3019$55af71c0$010e5540$@gmail.com> References: <02d201ce3019$55af71c0$010e5540$@gmail.com> Message-ID: it will happen soon during the regular weekly conference call Ken Sent from my iPad On Apr 2, 2013, at 22:14, "Usama Zaidi" wrote: > Hey, > > Any updates on when the conference call is going to be scheduled for? The > thread kinda died. > > Regards. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > freeswitch-users-request at lists.freeswitch.org > Sent: Tuesday, April 02, 2013 11:34 PM > To: freeswitch-users at lists.freeswitch.org > Subject: FreeSWITCH-users Digest, Vol 82, Issue 19 > > 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..." > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kennedy4260 at gmail.com Wed Apr 3 08:58:43 2013 From: kennedy4260 at gmail.com (Kevin Kennedy) Date: Tue, 2 Apr 2013 21:58:43 -0700 Subject: [Freeswitch-users] RFC4904 Trunkgroup Identity In-Reply-To: References: Message-ID: Michael, Thanks for the reply. The tgrp and trunk-context would be parameters in the contact header. It would look something like this. Contact: ... On Apr 2, 2013 10:17 AM, "Steven Ayre" wrote: > Try checking for the required information in the info app output or XML > CDR. > > > > If it's present in there, then you should be able to examine it from the > dialplan and act upon it. > > -Steve > > > On 1 April 2013 22:49, Michael Collins wrote: > >> I don't believe we support this as such. However, I've never seen a >> situation where FS can't route a call based on available criteria. Are >> these trunkgroup values in specific headers? If so it's possible to extract >> that information and use it for routing in the dialplan. >> >> If you have some call examples that you can put on >> pastebin.freeswitch.org that would be helpful. >> >> Thanks, >> MC >> >> On Sun, Mar 31, 2013 at 7:17 PM, Kevin Kennedy wrote: >> >>> I am new to freeswitch as far as testing it, but have been on the >>> user-list for a long time. I have searched through my archive of emails as >>> well as searched on google for any answer that I can find on this. I am >>> looking for the configuration options for RFC4904, trunk-group identity. >>> This is where you can set Freeswitch up to send unscreened calls through >>> your trunkgroup using the tgrp and trunk-context that is in the >>> registration for every outbound call. Any help would be 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.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/20130402/7be23092/attachment.html From avi at avimarcus.net Wed Apr 3 14:00:52 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 3 Apr 2013 13:00:52 +0300 Subject: [Freeswitch-users] RFC4904 Trunkgroup Identity In-Reply-To: References: Message-ID: Great. Then run the info application, see the variable it's stored in, and you can create a condition to regex it and you use that result for routing. -Avi Marcus BestFone On Wed, Apr 3, 2013 at 7:58 AM, Kevin Kennedy wrote: > Michael, > Thanks for the reply. The tgrp and trunk-context would be parameters in > the contact header. It would look something like this. > > Contact: trunk-context=example.com at gw1.example.com;user=phone> > > ... > On Apr 2, 2013 10:17 AM, "Steven Ayre" wrote: > >> Try checking for the required information in the info app output or XML >> CDR. >> >> >> >> If it's present in there, then you should be able to examine it from the >> dialplan and act upon it. >> >> -Steve >> >> >> On 1 April 2013 22:49, Michael Collins wrote: >> >>> I don't believe we support this as such. However, I've never seen a >>> situation where FS can't route a call based on available criteria. Are >>> these trunkgroup values in specific headers? If so it's possible to extract >>> that information and use it for routing in the dialplan. >>> >>> If you have some call examples that you can put on >>> pastebin.freeswitch.org that would be helpful. >>> >>> Thanks, >>> MC >>> >>> On Sun, Mar 31, 2013 at 7:17 PM, Kevin Kennedy wrote: >>> >>>> I am new to freeswitch as far as testing it, but have been on the >>>> user-list for a long time. I have searched through my archive of emails as >>>> well as searched on google for any answer that I can find on this. I am >>>> looking for the configuration options for RFC4904, trunk-group identity. >>>> This is where you can set Freeswitch up to send unscreened calls through >>>> your trunkgroup using the tgrp and trunk-context that is in the >>>> registration for every outbound call. Any help would be 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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.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/20130403/dccd5d0f/attachment-0001.html From b2m at a-cti.com Wed Apr 3 14:06:12 2013 From: b2m at a-cti.com (Bala Murugan Mahendran) Date: Wed, 3 Apr 2013 15:36:12 +0530 Subject: [Freeswitch-users] Outbound blocking code Message-ID: Looking to block few countries (or) need to allow few countries to make outbound call, is there a wiki examples for this? Thanks, Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/260618af/attachment.html From sertys at gmail.com Wed Apr 3 14:21:34 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Wed, 3 Apr 2013 12:21:34 +0200 Subject: [Freeswitch-users] Dynamic codec switching In-Reply-To: References: Message-ID: Exactly the reneg. I revisited the thread and will try it as soon as i have time. Is there a call quality metric i can use to check periodically on socket api and change codec accordingly? On Apr 2, 2013 9:30 AM, "Steven Ayre" wrote: > Remember AMR only works in passthrough mode. Though that should be fine > for a bridge. > > You're probably looking for this: > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_media_reneg > > > On 1 April 2013 23:34, Daniel Ivanov wrote: > >> I remember fairly well that there is dynamic codec negotiation in fs from >> a big thread here on the list. My question is can i use this to overcome >> transcoding difficulties for ringback. I have g729 licences on the box, but >> prefer amr for mobile connections. Can i send early media in g729 and then >> switch to amr for actual call? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20130403/236e101c/attachment.html From ashish at nms.co.in Wed Apr 3 11:38:15 2013 From: ashish at nms.co.in (Ashish gautam) Date: Wed, 3 Apr 2013 13:08:15 +0530 Subject: [Freeswitch-users] Getting FreeTDM originate status Message-ID: Hi, I have connected a PRI line to FreeSWITCH working with mod_freetdm enabled. Calls are going properly. Now, what I want is to get the originate status of each call i.e. whether the call was connected, failed, user busy or whatever the status is. How can I get this status from FreeSWTICH or FreeTDM? I am generating outgoing calls through event socket which receives originate request from a Perl script running on the same server. Please help and throw some light on this. -- Regards, Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/64a22212/attachment.html From thomas.lee at octon.net Wed Apr 3 14:50:54 2013 From: thomas.lee at octon.net (Thomas Lee) Date: Wed, 3 Apr 2013 03:50:54 -0700 (PDT) Subject: [Freeswitch-users] Can we add parameters to Request URI in SIP INVITE Message? FreeSWITCH how to get those parameters? Message-ID: <1364986254768-7589355.post@n2.nabble.com> Can we add parameters to Request URI in SIP INVITE Message? FreeSWITCH how to get those parameters? Thanks Regards, Thomas Lee -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-we-add-parameters-to-Request-URI-in-SIP-INVITE-Message-FreeSWITCH-how-to-get-those-parameters-tp7589355.html Sent from the freeswitch-users mailing list archive at Nabble.com. From thomas.lee at octon.net Wed Apr 3 15:42:14 2013 From: thomas.lee at octon.net (Thomas Lee) Date: Wed, 3 Apr 2013 04:42:14 -0700 (PDT) Subject: [Freeswitch-users] FreeSWITCH how to configure for Multi-Tenant? Message-ID: <1364989334549-7589356.post@n2.nabble.com> FreeSWITCH how to configure for Multi-Tenant? Thanks Regards, Thomas Lee -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-how-to-configure-for-Multi-Tenant-tp7589356.html Sent from the freeswitch-users mailing list archive at Nabble.com. From vipkilla at gmail.com Wed Apr 3 16:41:12 2013 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 3 Apr 2013 08:41:12 -0400 Subject: [Freeswitch-users] FreeSWITCH how to configure for Multi-Tenant? In-Reply-To: <1364989334549-7589356.post@n2.nabble.com> References: <1364989334549-7589356.post@n2.nabble.com> Message-ID: http://wiki.freeswitch.org/wiki/Multi-tenant On Wed, Apr 3, 2013 at 7:42 AM, Thomas Lee wrote: > FreeSWITCH how to configure for Multi-Tenant? > > Thanks > > Regards, > Thomas Lee > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-how-to-configure-for-Multi-Tenant-tp7589356.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/20130403/705907ee/attachment.html From acrow at integrafin.co.uk Wed Apr 3 17:38:55 2013 From: acrow at integrafin.co.uk (Alex Crow) Date: Wed, 03 Apr 2013 14:38:55 +0100 Subject: [Freeswitch-users] Spandsp modems as non-root? In-Reply-To: References: <50703D76.7010301@integrafin.co.uk> Message-ID: <515C30EF.8020301@integrafin.co.uk> Thanks Shahar (and Nestor who authored the patch)! Alex ----Original Message---- *Subject:* Re: [Freeswitch-users] Spandsp modems as non-root? *From:* Shahar *To:* FreeSWITCH Users Help *CC:* *Date:* Tue, 2 Apr 2013 19:48:14 -0400 > Alex, > > I've added another section exactly on this subject. > you might want to have a look at > http://wiki.freeswitch.org/wiki/HylaFax#Modem_devices_permissions_issue > > > > > On Sat, Oct 6, 2012 at 10:17 AM, Alex Crow > wrote: > > Hi, > > Has anyone figured out how to get the spandsp emulated modems running > when Freeswitch is running as a non-root user yet? > > I don't even get the /dev/pts/pts* devices added either on debian > or Ubuntu. > > Thanks > > Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 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 29 Clement's Lane, London EC4N 7AE 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/20130403/e748f603/attachment-0001.html From victor.chukalovskiy at gmail.com Wed Apr 3 17:42:29 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Wed, 03 Apr 2013 09:42:29 -0400 Subject: [Freeswitch-users] Can we add parameters to Request URI in SIP INVITE Message? FreeSWITCH how to get those parameters? In-Reply-To: <1364986254768-7589355.post@n2.nabble.com> References: <1364986254768-7589355.post@n2.nabble.com> Message-ID: <515C31C5.4020004@gmail.com> Yes. Examples (do this before bridge): Will attach "user=phone" SIP URI parameter in the INVITE request line Will attach "npdi=yes" TEL URI parameter in the INVITE request line -Victor On 04/03/2013 06:50 AM, Thomas Lee wrote: > Can we add parameters to Request URI in SIP INVITE Message? > FreeSWITCH how to get those parameters? > > Thanks > > Regards, > Thomas Lee > > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-we-add-parameters-to-Request-URI-in-SIP-INVITE-Message-FreeSWITCH-how-to-get-those-parameters-tp7589355.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 andrew at cassidywebservices.co.uk Wed Apr 3 17:50:11 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 3 Apr 2013 14:50:11 +0100 Subject: [Freeswitch-users] Spandsp modems as non-root? In-Reply-To: <515C30EF.8020301@integrafin.co.uk> References: <50703D76.7010301@integrafin.co.uk> <515C30EF.8020301@integrafin.co.uk> Message-ID: (Vaguely) related... I have a potential customer that still uses dial up banking. Which bank still lets them do this I have no idea. Could that be done in this way or in G.711 over SIP? On 3 April 2013 14:38, Alex Crow wrote: > Thanks Shahar (and Nestor who authored the patch)! > > Alex > > > ----Original Message---- > *Subject:* Re: [Freeswitch-users] Spandsp modems as non-root? > *From:* Shahar > *To:* FreeSWITCH Users Help > *CC:* > *Date:* Tue, 2 Apr 2013 19:48:14 -0400 > > Alex, > > I've added another section exactly on this subject. > you might want to have a look at > http://wiki.freeswitch.org/wiki/HylaFax#Modem_devices_permissions_issue > > > > > On Sat, Oct 6, 2012 at 10:17 AM, Alex Crow wrote: > >> Hi, >> >> Has anyone figured out how to get the spandsp emulated modems running >> when Freeswitch is running as a non-root user yet? >> >> I don't even get the /dev/pts/pts* devices added either on debian or >> Ubuntu. >> >> Thanks >> >> Alex >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > _________________________________________________________________________ > 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 > > > > > -- > 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 > 29 Clement's Lane, London EC4N 7AE > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/77789c66/attachment.html From fdelawarde at wirelessmundi.com Wed Apr 3 18:16:26 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?=) Date: Wed, 03 Apr 2013 16:16:26 +0200 Subject: [Freeswitch-users] B64 audio codec? In-Reply-To: <7847442903959552389@unknownmsgid> References: <1364917233.10160.90.camel@luna.madrid.commsmundi.com> <003801ce2fc3$0f3a99f0$2dafcdd0$@v10networks.ca> <7847442903959552389@unknownmsgid> Message-ID: <1364998586.10160.194.camel@luna.madrid.commsmundi.com> Hehe, wasn't quite sure as this was commited the 29th.. Still, I'll give it a try, base64 audio sounds like lots of fun! On Tue, 2013-04-02 at 12:23 -0600, Michael Jerris wrote: > Good jokes are functional. > > On Apr 2, 2013, at 12:00 PM, Jeff Leung wrote: > > > Personally I still have a hunch that mod_b64 is a subtle April Fool's Day joke. Even though it's a joke, it's still a very functional codec. ;) > > > >> -----Original Message----- > >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > >> users-bounces at lists.freeswitch.org] On Behalf Of Fran?ois > >> Sent: Tuesday, April 2, 2013 8:41 AM > >> To: freeswitch-users at lists.freeswitch.org > >> Subject: [Freeswitch-users] B64 audio codec? > >> > >> Hi all, > >> > >> Anyone has some info about the recently commited mod_b64 codec? > >> > >> The source refers to it as "The B64 ultra-low delay audio codec", and links to a > >> webpage www.b64-codec.org which does not exist! > >> > >> Anyway, it sounds interesting and doesn't seem to be an april fools joke! > >> > >> Thanks, > >> Fran?ois. > >> > >> > >> __________________________________________________________ > >> _______________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 thomas.lee at octon.net Wed Apr 3 18:04:05 2013 From: thomas.lee at octon.net (Thomas Lee) Date: Wed, 3 Apr 2013 07:04:05 -0700 (PDT) Subject: [Freeswitch-users] Can we add parameters to Request URI in SIP INVITE Message? FreeSWITCH how to get those parameters? In-Reply-To: <515C31C5.4020004@gmail.com> References: <1364986254768-7589355.post@n2.nabble.com> <515C31C5.4020004@gmail.com> Message-ID: <1364997845455-7589362.post@n2.nabble.com> Hi Victor, I will try it out. Thank you for your help. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-we-add-parameters-to-Request-URI-in-SIP-INVITE-Message-FreeSWITCH-how-to-get-those-parameters-tp7589355p7589362.html Sent from the freeswitch-users mailing list archive at Nabble.com. From thomas.lee at octon.net Wed Apr 3 18:05:34 2013 From: thomas.lee at octon.net (Thomas Lee) Date: Wed, 3 Apr 2013 07:05:34 -0700 (PDT) Subject: [Freeswitch-users] FreeSWITCH how to configure for Multi-Tenant? In-Reply-To: References: <1364989334549-7589356.post@n2.nabble.com> Message-ID: <1364997934407-7589363.post@n2.nabble.com> Hi Vik, Thank you for your help. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-how-to-configure-for-Multi-Tenant-tp7589356p7589363.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Wed Apr 3 20:35:18 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 3 Apr 2013 09:35:18 -0700 Subject: [Freeswitch-users] choppy audio In-Reply-To: <515A881C.6050709@gmail.com> References: <515A881C.6050709@gmail.com> Message-ID: Post the debug log for each call to pastebin.freeswitch.org. Let's all take a look at that first and see if there are any clues as to what is happening. -MC On Tue, Apr 2, 2013 at 12:26 AM, Jimmy Chang wrote: > Hi, > > I have two softphone clients commnunicating by G729. > The audio quality of direct call between them is perfect. > But when talking via FS, I got choppy audio in agent side(client side > remains perfect). > I had set bypass_media_after_bridge=true in dialplan, not work, and the > RTP traffic is still through FS. > > Any advice? > > Thanks. > Jimmy > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/e54ec391/attachment-0001.html From sdevoy at bizfocused.com Wed Apr 3 20:37:21 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 3 Apr 2013 12:37:21 -0400 Subject: [Freeswitch-users] One of my extensions Caller ID says Anonymous! Message-ID: <0c0801ce3089$839fe4f0$8adfaed0$@bizfocused.com> Hi, I have the current head (minus a week or so) I have many extensions/users defined almost identically in the directory: (redacted with ***'s) For almost everyone, when dialing a number, the log says: 2013-04-03 *** [INFO] mod_dialplan_xml.c:557 Processing 220 <220>->141454*** in context from-internal-***** For user 210, I get: 2013-04-03 *** [INFO] mod_dialplan_xml.c:557 Processing Anonymous <210>->141454*** in context from-internal-****** The phones are cisco 504Gs. Any idea where that is coming from our how I can fix it? Thanks Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/d54391fa/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/d54391fa/attachment-0001.gif From anthony.minessale at gmail.com Wed Apr 3 20:44:02 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 3 Apr 2013 11:44:02 -0500 Subject: [Freeswitch-users] One of my extensions Caller ID says Anonymous! In-Reply-To: <0c0801ce3089$839fe4f0$8adfaed0$@bizfocused.com> References: <0c0801ce3089$839fe4f0$8adfaed0$@bizfocused.com> Message-ID: Look at the sip trace and see if that is what they are putting in the caller id field. I don't believe we set that value anywhere in the code. You can override it with If you are provisioning or registering the phones you can possible set the name field in the config. On Wed, Apr 3, 2013 at 11:37 AM, Sean Devoy wrote: > Hi,**** > > ** ** > > I have the current head (minus a week or so)**** > > ** ** > > I have many extensions/users defined almost identically in the directory:* > *** > > (redacted with ***?s)**** > > **** > > **** > > **** > > *** > * > > **** > > **** > > *** > * > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > For almost everyone, when dialing a number, the log says:**** > > 2013-04-03 *** [INFO] mod_dialplan_xml.c:557 Processing 220 <220>->141454*** > in context from-internal-********* > > ** ** > > For user 210, I get:**** > > 2013-04-03 *** [INFO] mod_dialplan_xml.c:557 Processing *Anonymous* <210>->141454*** > in context from-internal-********** > > ** ** > > The phones are cisco 504Gs.**** > > ** ** > > Any idea where that is coming from our how I can fix it?**** > > ** ** > > Thanks**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130403/c90d1438/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/c90d1438/attachment-0001.gif From msc at freeswitch.org Wed Apr 3 20:47:11 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 3 Apr 2013 09:47:11 -0700 Subject: [Freeswitch-users] Getting FreeTDM originate status In-Reply-To: References: Message-ID: If you know the uuid of the call you can get any piece of information with uuid_getvar. You can also use uuid_dump to get the complete picture. I believe you can check several vars for information. I'd look at things like endpoint_disposition and DIALSTATUS vars specifically, although I recommend uuid_dump on an existing channel in various scenarios so that you can get a clear picture of what's happening. Of course, if the originate failed then you'll get that as the result of the originate command instead of a "+OK" -MC On Wed, Apr 3, 2013 at 12:38 AM, Ashish gautam wrote: > Hi, > > I have connected a PRI line to FreeSWITCH working with mod_freetdm > enabled. Calls are going properly. Now, what I want is to get the originate > status of each call i.e. whether the call was connected, failed, user busy > or whatever the status is. How can I get this status from FreeSWTICH or > FreeTDM? I am generating outgoing calls through event socket which receives > originate request from a Perl script running on the same server. > > Please help and throw some light on this. > > -- > Regards, > Ashish > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/9483c727/attachment.html From sdevoy at bizfocused.com Wed Apr 3 20:51:40 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 3 Apr 2013 12:51:40 -0400 Subject: [Freeswitch-users] FreeSWITCH how to configure for Multi-Tenant? In-Reply-To: <1364989334549-7589356.post@n2.nabble.com> References: <1364989334549-7589356.post@n2.nabble.com> Message-ID: <0c4401ce308b$83cb4fd0$8b61ef70$@bizfocused.com> Thomas, I have used what used to be multi-tenant in the wiki and has now been moved to Multiple Companies. It works great for Multiple "Groups of Users" whether you call them tenants or Companies and the wiki is more complete in its explanation. Be sure to look over: http://wiki.freeswitch.org/wiki/Multiple_Companies Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Thomas Lee Sent: Wednesday, April 03, 2013 7:42 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] FreeSWITCH how to configure for Multi-Tenant? FreeSWITCH how to configure for Multi-Tenant? Thanks Regards, Thomas Lee -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-how-to-configure-fo r-Multi-Tenant-tp7589356.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 msc at freeswitch.org Wed Apr 3 21:10:52 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 3 Apr 2013 10:10:52 -0700 Subject: [Freeswitch-users] Come join the conference call! Message-ID: We have several items to discuss, so please join us! http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_03 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/36b225c1/attachment.html From mytemike72 at gmail.com Wed Apr 3 21:15:33 2013 From: mytemike72 at gmail.com (Michael Lutz) Date: Wed, 3 Apr 2013 19:15:33 +0200 Subject: [Freeswitch-users] Regeneration of DTMF Message-ID: Hi, I have a problem, which I am trying to resolve, but can not exactly figure out where it is going wrong. I have an inbound call, this call comes in via SIP and uses inband dtmf, at the begining of the dialplan I enable dtmf detection using spandsp_start_dtmf. this works fine, and my Lua recognizes digits correctly. The tricky part is that I bridge this call in Lua using an api call "originate", this call is forwarded to the same switch, and is picked up by another Lua script. This script, is waiting for a custom event, to end the lua, and is bridged with a 3rd call. so the 1st and 3rd call can hear each other. This 3rd call is initiated asynchronosly by an esl server. (this al works fine, and is not 'the issue'..) The problem is the receiving end (3rd leg) is receiving the DTMF pressed by the 1st leg twice. When I eavesdrop the 2nd leg, i only hear the dtmf once, when i eavesdrop the 3rd leg, I can hear the dtmf twice. So it is somewhere generated along the way. I have tryed several different settings, using stop_dtmf_generate on different legs, but can not seem to diable this regeneration of this extra dtmf. Any help would be appreciated as this is really causing issues on my side, ps, I know this '3rd leg' principle might look a bit weird, but cannot be avoided, ps2. When my inbound call comes in using rfc2833, everything works perfectly. Best regards, Michael Lutz. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/8830fbdb/attachment.html From avi at avimarcus.net Wed Apr 3 21:32:25 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 3 Apr 2013 20:32:25 +0300 Subject: [Freeswitch-users] Regeneration of DTMF In-Reply-To: References: Message-ID: I can probably explain the issue to you, but I don't really know how to fix it: 1) Leg A comes in with inband. 2) Your leg B does start_dtmf and detects the inband dtmf. 3) You bridge to leg C which negotiates rfc2833. It gets the rfc2833 events from leg B. But! start_dtmf can't remove the dtmf from the leg A. So the leg A inband dtmf is ALSO being passed along. This however is only a problem if leg C has start_dtmf too. The default dialplan only triggers start_dtmf if there is no rfc2833 negotiated. But don't count on remote parties to do the same.... -Avi Marcus BestFone On Wed, Apr 3, 2013 at 8:15 PM, Michael Lutz wrote: > Hi, > > I have a problem, which I am trying to resolve, but can not exactly figure > out where it is going wrong. > > I have an inbound call, this call comes in via SIP and uses inband dtmf, > at the begining of the dialplan I enable dtmf detection using > spandsp_start_dtmf. this works fine, and my Lua recognizes digits correctly. > The tricky part is that I bridge this call in Lua using an api call > "originate", this call is forwarded to the same switch, and is picked up by > another Lua script. > This script, is waiting for a custom event, to end the lua, and is bridged > with a 3rd call. so the 1st and 3rd call can hear each other. This 3rd call > is initiated asynchronosly by an esl server. (this al works fine, and is > not 'the issue'..) > > The problem is the receiving end (3rd leg) is receiving the DTMF pressed > by the 1st leg twice. When I eavesdrop the 2nd leg, i only hear the dtmf > once, when i eavesdrop the 3rd leg, I can hear the dtmf twice. So it is > somewhere generated along the way. > > I have tryed several different settings, using stop_dtmf_generate on > different legs, but can not seem to diable this regeneration of this extra > dtmf. > > Any help would be appreciated as this is really causing issues on my side, > > ps, I know this '3rd leg' principle might look a bit weird, but cannot be > avoided, > ps2. When my inbound call comes in using rfc2833, everything works > perfectly. > > > Best regards, > Michael Lutz. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/06c11b6f/attachment.html From drk at drkngs.net Wed Apr 3 21:50:31 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Wed, 03 Apr 2013 10:50:31 -0700 Subject: [Freeswitch-users] Mod Managed Conf Call In-Reply-To: Message-ID: <20130403175031.7b0dab31@mail.tritonwest.net> Ken this was going to be out of band. Sorry for the delay. My hand had a dispute with a knife, and it's out of comission for about a week, so It's going to have to wait till next week, when I'm back to tying. Can everyoine that want's to be included, e-mail me directly drk at drkngs.net and we can schedulel a time. I see the doctor on Friday, so I should be able to type normal after that appointment. --Dave _____ From: Ken Rice [mailto:krice at freeswitch.org] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Cc: [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 02 Apr 2013 20:25:25 -0700 Subject: Re: [Freeswitch-users] Mod Managed Conf Call it will happen soon during the regular weekly conference call Ken Sent from my iPad On Apr 2, 2013, at 22:14, "Usama Zaidi" wrote: > Hey, > > Any updates on when the conference call is going to be scheduled for? The > thread kinda died. > > Regards. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > freeswitch-users-request at lists.freeswitch.org > Sent: Tuesday, April 02, 2013 11:34 PM > To: freeswitch-users at lists.freeswitch.org > Subject: FreeSWITCH-users Digest, Vol 82, Issue 19 > > 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..." > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20130403/851bb495/attachment-0001.html From krice at freeswitch.org Wed Apr 3 23:10:13 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 03 Apr 2013 14:10:13 -0500 Subject: [Freeswitch-users] Mod Managed Conf Call In-Reply-To: <20130403175031.7b0dab31@mail.tritonwest.net> Message-ID: Its going to be inband... Talk to Collins On 4/3/13 12:50 PM, "Dave R. Kompel" wrote: > Ken this was going to be out of band. Sorry for the delay. My hand had a > dispute with a knife, and it's out of comission for about a week, so It's > going to have to wait till next week, when I'm back to tying. > > Can everyoine that want's to be included, e-mail me directly drk at drkngs.net > and we can schedulel a time. I see the doctor on Friday, so I should be able > to type normal after that appointment. > > --Dave > >> >> From: Ken Rice [mailto:krice at freeswitch.org] >> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] >> Cc: >> [mailto:freeswitch-users at lists.freeswitch.org] >> Sent: Tue, 02 Apr 2013 20:25:25 -0700 >> Subject: Re: [Freeswitch-users] Mod Managed Conf Call >> >> it will happen soon during the regular weekly conference call >> >> Ken >> Sent from my iPad >> >> On Apr 2, 2013, at 22:14, "Usama Zaidi" wrote: >> >>> > Hey, >>> > >>> > Any updates on when the conference call is going to be scheduled for? The >>> > thread kinda died. >>> > >>> > Regards. >>> > >>> > -----Original Message----- >>> > From: freeswitch-users-bounces at lists.freeswitch.org >>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >>> > freeswitch-users-request at lists.freeswitch.org >>> > Sent: Tuesday, April 02, 2013 11:34 PM >>> > To: freeswitch-users at lists.freeswitch.org >>> > Subject: FreeSWITCH-users Digest, Vol 82, Issue 19 >>> > >>> > 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..." >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/01f37561/attachment.html From shaharhd at gmail.com Wed Apr 3 23:19:44 2013 From: shaharhd at gmail.com (Shahar) Date: Wed, 3 Apr 2013 15:19:44 -0400 Subject: [Freeswitch-users] Spandsp modems as non-root? In-Reply-To: References: <50703D76.7010301@integrafin.co.uk> <515C30EF.8020301@integrafin.co.uk> Message-ID: you're asking about using the spandsp softmodem for implementing dialup PPP connections? On Wed, Apr 3, 2013 at 9:50 AM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > (Vaguely) related... I have a potential customer that still uses dial up > banking. Which bank still lets them do this I have no idea. Could that be > done in this way or in G.711 over SIP? > > > On 3 April 2013 14:38, Alex Crow wrote: > >> Thanks Shahar (and Nestor who authored the patch)! >> >> Alex >> >> >> ----Original Message---- >> *Subject:* Re: [Freeswitch-users] Spandsp modems as non-root? >> *From:* Shahar >> *To:* FreeSWITCH Users Help >> *CC:* >> *Date:* Tue, 2 Apr 2013 19:48:14 -0400 >> >> Alex, >> >> I've added another section exactly on this subject. >> you might want to have a look at >> http://wiki.freeswitch.org/wiki/HylaFax#Modem_devices_permissions_issue >> >> >> >> >> On Sat, Oct 6, 2012 at 10:17 AM, Alex Crow wrote: >> >>> Hi, >>> >>> Has anyone figured out how to get the spandsp emulated modems running >>> when Freeswitch is running as a non-root user yet? >>> >>> I don't even get the /dev/pts/pts* devices added either on debian or >>> Ubuntu. >>> >>> Thanks >>> >>> Alex >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 has been scanned for viruses and >> dangerous content by *MailScanner* , and >> is >> believed to be clean. >> >> _________________________________________________________________________ >> 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 >> >> >> >> >> -- >> 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 >> 29 Clement's Lane, London EC4N 7AE >> 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 >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/3285e539/attachment-0001.html From shaharhd at gmail.com Wed Apr 3 23:20:11 2013 From: shaharhd at gmail.com (Shahar) Date: Wed, 3 Apr 2013 15:20:11 -0400 Subject: [Freeswitch-users] Spandsp modems as non-root? In-Reply-To: References: <50703D76.7010301@integrafin.co.uk> <515C30EF.8020301@integrafin.co.uk> Message-ID: Alex - we're you able to utilize the /dev/FS devices with HylaFax? On Wed, Apr 3, 2013 at 3:19 PM, Shahar wrote: > you're asking about using the spandsp softmodem for implementing dialup > PPP connections? > > > On Wed, Apr 3, 2013 at 9:50 AM, Andrew Cassidy < > andrew at cassidywebservices.co.uk> wrote: > >> (Vaguely) related... I have a potential customer that still uses dial up >> banking. Which bank still lets them do this I have no idea. Could that be >> done in this way or in G.711 over SIP? >> >> >> On 3 April 2013 14:38, Alex Crow wrote: >> >>> Thanks Shahar (and Nestor who authored the patch)! >>> >>> Alex >>> >>> >>> ----Original Message---- >>> *Subject:* Re: [Freeswitch-users] Spandsp modems as non-root? >>> *From:* Shahar >>> *To:* FreeSWITCH Users Help >>> *CC:* >>> *Date:* Tue, 2 Apr 2013 19:48:14 -0400 >>> >>> Alex, >>> >>> I've added another section exactly on this subject. >>> you might want to have a look at >>> http://wiki.freeswitch.org/wiki/HylaFax#Modem_devices_permissions_issue >>> >>> >>> >>> >>> On Sat, Oct 6, 2012 at 10:17 AM, Alex Crow wrote: >>> >>>> Hi, >>>> >>>> Has anyone figured out how to get the spandsp emulated modems running >>>> when Freeswitch is running as a non-root user yet? >>>> >>>> I don't even get the /dev/pts/pts* devices added either on debian or >>>> Ubuntu. >>>> >>>> Thanks >>>> >>>> Alex >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 has been scanned for viruses and >>> dangerous content by *MailScanner* , and >>> is >>> believed to be clean. >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >>> >>> >>> -- >>> 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 >>> 29 Clement's Lane, London EC4N 7AE >>> 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 >>> >>> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 *F >> *03300 100 961 >> *E *andrew at cassidywebservices.co.uk >> *W *www.cassidywebservices.co.uk >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/73008cb1/attachment.html From andretodd at verizon.net Wed Apr 3 23:21:19 2013 From: andretodd at verizon.net (Andre) Date: Wed, 03 Apr 2013 15:21:19 -0400 Subject: [Freeswitch-users] Mod Managed Conf Call In-Reply-To: References: <20130403175031.7b0dab31@mail.tritonwest.net> Message-ID: <02b501ce30a0$6c8020c0$45806240$@verizon.net> When it's decided can you send the information needed to join for new users like me? Thanks Andre From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Wednesday, April 3, 2013 3:10 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Mod Managed Conf Call Its going to be inband... Talk to Collins On 4/3/13 12:50 PM, "Dave R. Kompel" wrote: Ken this was going to be out of band. Sorry for the delay. My hand had a dispute with a knife, and it's out of comission for about a week, so It's going to have to wait till next week, when I'm back to tying. Can everyoine that want's to be included, e-mail me directly drk at drkngs.net and we can schedulel a time. I see the doctor on Friday, so I should be able to type normal after that appointment. --Dave _____ From: Ken Rice [mailto:krice at freeswitch.org] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Cc: [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 02 Apr 2013 20:25:25 -0700 Subject: Re: [Freeswitch-users] Mod Managed Conf Call it will happen soon during the regular weekly conference call Ken Sent from my iPad On Apr 2, 2013, at 22:14, "Usama Zaidi" wrote: > Hey, > > Any updates on when the conference call is going to be scheduled for? The > thread kinda died. > > Regards. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > freeswitch-users-request at lists.freeswitch.org > Sent: Tuesday, April 02, 2013 11:34 PM > To: freeswitch-users at lists.freeswitch.org > Subject: FreeSWITCH-users Digest, Vol 82, Issue 19 > > 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..." > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/17273678/attachment-0001.html From kennedy4260 at gmail.com Wed Apr 3 22:41:08 2013 From: kennedy4260 at gmail.com (Kevin Kennedy) Date: Wed, 3 Apr 2013 11:41:08 -0700 Subject: [Freeswitch-users] RFC4904 Trunkgroup Identity In-Reply-To: References: Message-ID: Can this information be added into the registration as well? On Apr 3, 2013 3:03 AM, "Avi Marcus" wrote: > Great. Then run the info application, see the variable it's stored in, and > you can create a condition to regex it and you use that result for routing. > > -Avi Marcus > BestFone > > On Wed, Apr 3, 2013 at 7:58 AM, Kevin Kennedy wrote: > >> Michael, >> Thanks for the reply. The tgrp and trunk-context would be parameters in >> the contact header. It would look something like this. >> >> Contact: > trunk-context=example.com at gw1.example.com;user=phone> >> >> ... >> On Apr 2, 2013 10:17 AM, "Steven Ayre" wrote: >> >>> Try checking for the required information in the info app output or XML >>> CDR. >>> >>> >>> >>> If it's present in there, then you should be able to examine it from the >>> dialplan and act upon it. >>> >>> -Steve >>> >>> >>> On 1 April 2013 22:49, Michael Collins wrote: >>> >>>> I don't believe we support this as such. However, I've never seen a >>>> situation where FS can't route a call based on available criteria. Are >>>> these trunkgroup values in specific headers? If so it's possible to extract >>>> that information and use it for routing in the dialplan. >>>> >>>> If you have some call examples that you can put on >>>> pastebin.freeswitch.org that would be helpful. >>>> >>>> Thanks, >>>> MC >>>> >>>> On Sun, Mar 31, 2013 at 7:17 PM, Kevin Kennedy wrote: >>>> >>>>> I am new to freeswitch as far as testing it, but have been on the >>>>> user-list for a long time. I have searched through my archive of emails as >>>>> well as searched on google for any answer that I can find on this. I am >>>>> looking for the configuration options for RFC4904, trunk-group identity. >>>>> This is where you can set Freeswitch up to send unscreened calls through >>>>> your trunkgroup using the tgrp and trunk-context that is in the >>>>> registration for every outbound call. Any help would be 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.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/20130403/d18f4161/attachment.html From msc at freeswitch.org Wed Apr 3 23:48:53 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 3 Apr 2013 12:48:53 -0700 Subject: [Freeswitch-users] Outbound blocking code In-Reply-To: References: Message-ID: You could start with this: http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_16:_Block_certain_codes -MC On Wed, Apr 3, 2013 at 3:06 AM, Bala Murugan Mahendran wrote: > Looking to block few countries (or) need to allow few countries to make > outbound call, is there a wiki examples for this? > > Thanks, > Bala > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/3692aa8b/attachment.html From sdevoy at bizfocused.com Thu Apr 4 00:36:29 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 3 Apr 2013 16:36:29 -0400 Subject: [Freeswitch-users] One of my extensions Caller ID says Anonymous! In-Reply-To: References: <0c0801ce3089$839fe4f0$8adfaed0$@bizfocused.com> Message-ID: <0e6f01ce30aa$ebb5b210$c3211630$@bizfocused.com> Thanks A.M. I have to figure out where that is stored in the 504G's config. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, April 03, 2013 12:44 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One of my extensions Caller ID says Anonymous! Look at the sip trace and see if that is what they are putting in the caller id field. I don't believe we set that value anywhere in the code. You can override it with If you are provisioning or registering the phones you can possible set the name field in the config. On Wed, Apr 3, 2013 at 11:37 AM, Sean Devoy wrote: Hi, I have the current head (minus a week or so) I have many extensions/users defined almost identically in the directory: (redacted with ***'s) For almost everyone, when dialing a number, the log says: 2013-04-03 *** [INFO] mod_dialplan_xml.c:557 Processing 220 <220>->141454*** in context from-internal-***** For user 210, I get: 2013-04-03 *** [INFO] mod_dialplan_xml.c:557 Processing Anonymous <210>->141454*** in context from-internal-****** The phones are cisco 504Gs. Any idea where that is coming from our how I can fix it? Thanks Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130403/29d4d80c/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/29d4d80c/attachment-0001.gif From vipkilla at gmail.com Thu Apr 4 00:53:58 2013 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 3 Apr 2013 16:53:58 -0400 Subject: [Freeswitch-users] One of my extensions Caller ID says Anonymous! In-Reply-To: <0e6f01ce30aa$ebb5b210$c3211630$@bizfocused.com> References: <0c0801ce3089$839fe4f0$8adfaed0$@bizfocused.com> <0e6f01ce30aa$ebb5b210$c3211630$@bizfocused.com> Message-ID: I'm having the same problem w/ the Cisco What did you do to fix it? On Wed, Apr 3, 2013 at 4:36 PM, Sean Devoy wrote: > Thanks A.M.**** > > ** ** > > I have to figure out where that is stored in the 504G?s config.**** > > ** ** > > Sean**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Wednesday, April 03, 2013 12:44 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] One of my extensions Caller ID says > Anonymous!**** > > ** ** > > Look at the sip trace and see if that is what they are putting in the > caller id field.**** > > I don't believe we set that value anywhere in the code. You can override > it with**** > > **** > > ** ** > > If you are provisioning or registering the phones you can possible set the > name field in the config.**** > > ** ** > > ** ** > > On Wed, Apr 3, 2013 at 11:37 AM, Sean Devoy wrote: > **** > > **** > > Hi,**** > > **** > > I have the current head (minus a week or so)**** > > **** > > I have many extensions/users defined almost identically in the directory:* > *** > > (redacted with ***?s)**** > > **** > > **** > > **** > > *** > * > > **** > > **** > > *** > * > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > For almost everyone, when dialing a number, the log says:**** > > 2013-04-03 *** [INFO] mod_dialplan_xml.c:557 Processing 220 <220>->141454*** > in context from-internal-********* > > **** > > For user 210, I get:**** > > 2013-04-03 *** [INFO] mod_dialplan_xml.c:557 Processing *Anonymous* <210>->141454*** > in context from-internal-********** > > **** > > The phones are cisco 504Gs.**** > > **** > > Any idea where that is coming from our how I can fix it?**** > > **** > > Thanks**** > > Sean**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at 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/20130403/1d831480/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/1d831480/attachment.gif From cmason at frontiernetworks.ca Thu Apr 4 03:09:27 2013 From: cmason at frontiernetworks.ca (Colin Mason) Date: Wed, 3 Apr 2013 19:09:27 -0400 Subject: [Freeswitch-users] CRIT switch_ivr_bridge in log file Message-ID: <0D1C698866F66045A6201FD0F59CAC900147722B8F@EX.frontier.local> Hello, I'm using FreeSWITCH version: FreeSWITCH Version 1.2.8+git~20130402T040229Z~b72d2c32d7 (git b72d2c3 2013-04-02 04:02:29Z) An inbound call is sent to my box and after the far end (caller) ACKs my 200 OK, FreeSWITCH prints this to the log and then sends off a SIP INFO packet to the callee with caller id information. In the log CALLERNAME and CALLERNUM are replaced by the actual caller details: 2013-04-03 19:01:30.899243 [CRIT] switch_ivr_bridge.c:133 SEND sofia/carrier/XXXXXXXXXX [CALLERNAME][CALLERNUM] 2013-04-03 19:01:30.899243 [CRIT] switch_ivr_bridge.c:133 SEND sofia/internet/XXXXXXXXXX at X.X.X.X:5060 [CALLERNAME][CALLERNUM] It's a critical error and it is showing up for every inbound call. Everything appears to be operating normally. Is this error just printing information to the log or should I be concerned about it? Thanks, Colin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/69a67c81/attachment-0001.html From drk at drkngs.net Thu Apr 4 03:23:34 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Wed, 03 Apr 2013 16:23:34 -0700 Subject: [Freeswitch-users] Mod Managed Conf Call In-Reply-To: Message-ID: <20130403232334.a4d0f38d@mail.tritonwest.net> I can do a shorter one "inband" but thas was going to be out of band (not on a meeting) that would be more intense, including having ppl follow along and build full application themself. --Dave _____ From: Ken Rice [mailto:krice at freeswitch.org] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Wed, 03 Apr 2013 12:10:13 -0700 Subject: Re: [Freeswitch-users] Mod Managed Conf Call Its going to be inband... Talk to Collins On 4/3/13 12:50 PM, "Dave R. Kompel" wrote: Ken this was going to be out of band. Sorry for the delay. My hand had a dispute with a knife, and it's out of comission for about a week, so It's going to have to wait till next week, when I'm back to tying. Can everyoine that want's to be included, e-mail me directly drk at drkngs.net and we can schedulel a time. I see the doctor on Friday, so I should be able to type normal after that appointment. --Dave _____ From: Ken Rice [mailto:krice at freeswitch.org] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Cc: [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 02 Apr 2013 20:25:25 -0700 Subject: Re: [Freeswitch-users] Mod Managed Conf Call it will happen soon during the regular weekly conference call Ken Sent from my iPad On Apr 2, 2013, at 22:14, "Usama Zaidi" wrote: > Hey, > > Any updates on when the conference call is going to be scheduled for? The > thread kinda died. > > Regards. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > freeswitch-users-request at lists.freeswitch.org > Sent: Tuesday, April 02, 2013 11:34 PM > To: freeswitch-users at lists.freeswitch.org > Subject: FreeSWITCH-users Digest, Vol 82, Issue 19 > > 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..." > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/d2e071ee/attachment.html From anthony.minessale at gmail.com Thu Apr 4 03:43:33 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 3 Apr 2013 18:43:33 -0500 Subject: [Freeswitch-users] CRIT switch_ivr_bridge in log file In-Reply-To: <0D1C698866F66045A6201FD0F59CAC900147722B8F@EX.frontier.local> References: <0D1C698866F66045A6201FD0F59CAC900147722B8F@EX.frontier.local> Message-ID: That was an extra debug line, its been removed in latest code in the repo. Its harmless to you and it will be gone when you update. On Wed, Apr 3, 2013 at 6:09 PM, Colin Mason wrote: > Hello,**** > > ** ** > > I?m using FreeSWITCH version:**** > > FreeSWITCH Version 1.2.8+git~20130402T040229Z~b72d2c32d7 (git b72d2c3 > 2013-04-02 04:02:29Z)**** > > ** ** > > An inbound call is sent to my box and after the far end (caller) ACKs my > 200 OK, FreeSWITCH prints this to the log and then sends off a SIP INFO > packet to the callee with caller id information. In the log CALLERNAME and > CALLERNUM are replaced by the actual caller details:**** > > ** ** > > ** ** > > 2013-04-03 19:01:30.899243 [CRIT] switch_ivr_bridge.c:133 SEND > sofia/carrier/XXXXXXXXXX [CALLERNAME][CALLERNUM]**** > > 2013-04-03 19:01:30.899243 [CRIT] switch_ivr_bridge.c:133 SEND > sofia/internet/XXXXXXXXXX at X.X.X.X:5060 [CALLERNAME][CALLERNUM]**** > > ** ** > > ** ** > > It?s a critical error and it is showing up for every inbound call. > Everything appears to be operating normally. Is this error just printing > information to the log or should I be concerned about it?**** > > ** ** > > Thanks,**** > > Colin**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130403/4d52a4c1/attachment.html From adrian.fuentes at hovanetworks.com Thu Apr 4 04:18:00 2013 From: adrian.fuentes at hovanetworks.com (Adrian Fuentes) Date: Wed, 3 Apr 2013 18:18:00 -0600 Subject: [Freeswitch-users] Can be modified switch_m_sdp? Message-ID: <0D53731D-535B-4AD6-9F87-D88690D0A18C@hovanetworks.com> Hi all Is it possible to replace the sdp c = IN IP4 172.25.1.9 through 172.25.1.9 c = IN IP4? variable_switch_m_sdp: [v=0 o=- 13649160524104272291 13649160524104272291 IN IP4 10.5.86.99 s=it-ssw c=IN IP4 172.25.1.9 t=0 0 m=audio 6150 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - ] Regards Adrian Fuentes From jnankin at gmail.com Thu Apr 4 04:19:04 2013 From: jnankin at gmail.com (Josh Nankin) Date: Wed, 3 Apr 2013 19:19:04 -0500 Subject: [Freeswitch-users] Is disable-rtp-auto-adjust broken? Message-ID: I keep seeing lines like the following in my logs: 2013-04-03 19:14:09.252741 [INFO] switch_rtp.c:3684 Auto Changing port from xx.xx.xx.xx:25682 to xx.xx.xx.xx:25622 I have tried using the disable-rtp-auto-adjust option in my sip profile, as well as specifying it in my dialplan as well using the respective variable. Nothing seems to stop this. I'm trying to disable auto changing ports because when receiving a fax Freeswitch claims to be transmitting a CED message in the logs, but this message is not showing up in packet capture flows when viewing calls in Wireshark. My thoughts are currently that this is because freeswitch is not sending packets to the port requested by my SIP provider, and instead sending it to some unexpected port. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/9f5e2be1/attachment-0001.html From krice at freeswitch.org Thu Apr 4 05:07:29 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 3 Apr 2013 20:07:29 -0500 Subject: [Freeswitch-users] Can be modified switch_m_sdp? In-Reply-To: <0D53731D-535B-4AD6-9F87-D88690D0A18C@hovanetworks.com> References: <0D53731D-535B-4AD6-9F87-D88690D0A18C@hovanetworks.com> Message-ID: what are you trying to do? you need to be specific... you can modify the sdp possibly but the end goal tells us how to advise this Ken Sent from my iPad On Apr 3, 2013, at 19:18, Adrian Fuentes wrote: > Hi all > > Is it possible to replace the sdp c = IN IP4 172.25.1.9 through 172.25.1.9 c = IN IP4? > > variable_switch_m_sdp: [v=0 > o=- 13649160524104272291 13649160524104272291 IN IP4 10.5.86.99 > s=it-ssw > c=IN IP4 172.25.1.9 > t=0 0 > m=audio 6150 RTP/AVP 8 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=silenceSupp:off - - - - > ] > > Regards > Adrian Fuentes > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From thomas.lee at octon.net Thu Apr 4 06:54:58 2013 From: thomas.lee at octon.net (Thomas Lee) Date: Wed, 3 Apr 2013 19:54:58 -0700 (PDT) Subject: [Freeswitch-users] FreeSWITCH how to configure for Multi-Tenant? In-Reply-To: <0c4401ce308b$83cb4fd0$8b61ef70$@bizfocused.com> References: <1364989334549-7589356.post@n2.nabble.com> <0c4401ce308b$83cb4fd0$8b61ef70$@bizfocused.com> Message-ID: <1365044098024-7589388.post@n2.nabble.com> Hi Sean, Thank you for your information. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-how-to-configure-for-Multi-Tenant-tp7589356p7589388.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jnankin at gmail.com Thu Apr 4 06:58:19 2013 From: jnankin at gmail.com (Josh Nankin) Date: Wed, 3 Apr 2013 21:58:19 -0500 Subject: [Freeswitch-users] Is disable-rtp-auto-adjust broken? In-Reply-To: References: Message-ID: For some reason the dialplan variable wasn't working for me, and I put the parameter in the portion of my sip profile, not the settings area. Putting this in the settings section of my external sip profile worked. On Wed, Apr 3, 2013 at 7:19 PM, Josh Nankin wrote: > I keep seeing lines like the following in my logs: > > 2013-04-03 19:14:09.252741 [INFO] switch_rtp.c:3684 Auto Changing port > from xx.xx.xx.xx:25682 to xx.xx.xx.xx:25622 > > I have tried using the disable-rtp-auto-adjust option in my sip profile, > as well as specifying it in my dialplan as well using the respective > variable. Nothing seems to stop this. > > I'm trying to disable auto changing ports because when receiving a fax > Freeswitch claims to be transmitting a CED message in the logs, but this > message is not showing up in packet capture flows when viewing calls in > Wireshark. My thoughts are currently that this is because freeswitch is > not sending packets to the port requested by my SIP provider, and instead > sending it to some unexpected port. > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/c017f3d2/attachment.html From anthony.minessale at gmail.com Thu Apr 4 07:19:03 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 3 Apr 2013 22:19:03 -0500 Subject: [Freeswitch-users] Is disable-rtp-auto-adjust broken? In-Reply-To: References: Message-ID: report a jira please http://jira.freeswitch.org On Wed, Apr 3, 2013 at 9:58 PM, Josh Nankin wrote: > For some reason the dialplan variable wasn't working for me, and I put the > parameter in the portion of my sip profile, not the settings > area. Putting this in the settings section of my external sip profile > worked. > > > On Wed, Apr 3, 2013 at 7:19 PM, Josh Nankin wrote: > >> I keep seeing lines like the following in my logs: >> >> 2013-04-03 19:14:09.252741 [INFO] switch_rtp.c:3684 Auto Changing port >> from xx.xx.xx.xx:25682 to xx.xx.xx.xx:25622 >> >> I have tried using the disable-rtp-auto-adjust option in my sip profile, >> as well as specifying it in my dialplan as well using the respective >> variable. Nothing seems to stop this. >> >> I'm trying to disable auto changing ports because when receiving a fax >> Freeswitch claims to be transmitting a CED message in the logs, but this >> message is not showing up in packet capture flows when viewing calls in >> Wireshark. My thoughts are currently that this is because freeswitch is >> not sending packets to the port requested by my SIP provider, and instead >> sending it to some unexpected port. >> >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130403/3bf24bb2/attachment.html From info at shishir.com.np Thu Apr 4 07:21:25 2013 From: info at shishir.com.np (info at shishir.com.np) Date: Wed, 03 Apr 2013 20:21:25 -0700 Subject: [Freeswitch-users] =?utf-8?q?FreeSWITCH_how_to_configure_for_Mult?= =?utf-8?q?i-Tenant=3F?= In-Reply-To: <1365044098024-7589388.post@n2.nabble.com> References: <1364989334549-7589356.post@n2.nabble.com> <0c4401ce308b$83cb4fd0$8b61ef70$@bizfocused.com> <1365044098024-7589388.post@n2.nabble.com> Message-ID: <3cfd5cb3530036d52adee60f6ef7bcb9@shishir.com.np> Here is an another example for multi tenant configuration http://wiki.freeswitch.org/wiki/Multi-tenant Regards, On 03.04.2013 19:54, Thomas Lee wrote: > Hi Sean, > > Thank you for your information. > > > > > -- > View this message in context: > > http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-how-to-configure-for-Multi-Tenant-tp7589356p7589388.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 thomas.lee at octon.net Thu Apr 4 07:34:17 2013 From: thomas.lee at octon.net (Thomas Lee) Date: Wed, 3 Apr 2013 20:34:17 -0700 (PDT) Subject: [Freeswitch-users] When doing re-INVITE procedure.... Message-ID: <1365046457506-7589392.post@n2.nabble.com> Hi, It's SIP re-INVITE problem. Is it necessary for adding "a=sendrecv" on SDP field, when we're doing re-INVITE procedure? Thanks Regards, Thomas Lee -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/When-doing-re-INVITE-procedure-tp7589392.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Thu Apr 4 12:02:27 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 4 Apr 2013 09:02:27 +0100 Subject: [Freeswitch-users] When doing re-INVITE procedure.... In-Reply-To: <1365046457506-7589392.post@n2.nabble.com> References: <1365046457506-7589392.post@n2.nabble.com> Message-ID: It indicates you plan to have 2-way audio, so yes. Omitting it entirely should default to sendrecv. Can you enlighten us on what the actual problem you're having is? -Steve On 4 April 2013 04:34, Thomas Lee wrote: > Hi, > > It's SIP re-INVITE problem. > > Is it necessary for adding "a=sendrecv" on SDP field, when we're doing > re-INVITE procedure? > > Thanks > > Regards, > Thomas Lee > > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/When-doing-re-INVITE-procedure-tp7589392.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/20130404/7af6f832/attachment-0001.html From Alexander.Haugg at c4b.de Thu Apr 4 12:17:13 2013 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Thu, 4 Apr 2013 08:17:13 +0000 Subject: [Freeswitch-users] Build Error In-Reply-To: <20130402183348.0aedd30c@mail.tritonwest.net> References: <20130402183348.0aedd30c@mail.tritonwest.net> Message-ID: It is fixed, thanks. After ?git pull? and ?git clean ?fdx? tested with win32 Debug/Release and x64 Debug/Release Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Dave R. Kompel Gesendet: Dienstag, 2. April 2013 20:34 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Build Error It looks like it's gonna be some amount of work to get it to build on windows again... --Dave ________________________________ From: Alexander Haugg [mailto:Alexander.Haugg at c4b.de] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 02 Apr 2013 00:01:04 -0700 Subject: Re: [Freeswitch-users] Build Error I have some build errors if i try to build the "Freeswitch.2012.sln". My GIT repository is up-to-date. The first error is: 32>------ Rebuild All started: Project: iksemel, Configuration: Debug Win32 ------ 3> dso_lib.c 32> utility.c 32> stream.c 32>..\..\iksemel\src\stream.c(23): fatal error C1083: Cannot open include file: 'openssl/ssl.h': No such file or directory 32> sha.c 32> sax.c But the file exist: freeswitch\libs\openssl-1.0.1c\include\openssl\ssl.h Thanks -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von info at shishir.com.np Gesendet: Dienstag, 2. April 2013 08:14 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Build Error Thanks guys! Error was resolve with the latest changes on the git. It was resolved after latest push, "git clean -fdx", git pull, with bootstrap and ./configure Thanks again guys! On 01.04.2013 21:54, info at shishir.com.np wrote: > I still get the same error after updating too, I even tried on the > fresh git checkout, its same issue. Is it only me getting this error ? > > > On 01.04.2013 15:09, DJB International wrote: >> Ken, >> >> I also experienced the built error today, but not sure whether it's >> related. http://jira.freeswitch.org/browse/FS-5248 [27] >> >> On Mon, Apr 1, 2013 at 2:54 PM, Ken Rice wrote: >> >>> Update those should be fixed already... >>> >>> Major updates were rolled in this morning... >>> >>> On 4/1/13 4:20 PM, "info at shishir.com.np [1]" wrote: >>> >>> > Hi >>> >>> I Came across the following error while performing make current >>> this >>> > >>> morning. Please suggest. >>> >>> quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. >>> > -I./src/include >>> -I./libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include >>> > >>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >>> > >>> -I/usr/src/freeswitch/libs/libteletone/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -fPIC -fvisibility=hidden >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>> > >>> -Ilibs/sofia-sip/libsofia-sip-ua/sdp >>> -Ilibs/sofia-sip/libsofia-sip-ua/su >>> -g >>> > -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE >>> -D_LARGEFILE64_SOURCE >>> > -I/usr/src/freeswitch/libs/apr/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/include >>> > >>> -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>> > >>> -I/usr/src/freeswitch/libs/libtpl-1.5/src >>> -I/usr/src/freeswitch/libs/stfu >>> > -I/usr/src/freeswitch/libs/sqlite >>> -I/usr/src/freeswitch/libs/pcre >>> > -I/usr/src/freeswitch/libs/speex/include >>> -Ilibs/speex/include >>> > -I/usr/src/freeswitch/libs/srtp/include >>> > >>> -I/usr/src/freeswitch/libs/srtp/crypto/include >>> -Ilibs/srtp/crypto/include >>> > -I/usr/src/freeswitch/libs/spandsp/src >>> > >>> -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP >>> > >>> -I/usr/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT >>> > >>> -I/usr/src/freeswitch/libs/curl/include >>> -I/usr/src/freeswitch/src/include >>> > -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/libs/libteletone/src >>> > >>> -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden >>> > >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb >>> -DHAVE_OPENSSL >>> -Wall >>> > -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT >>> > >>> libfreeswitch_la-switch_rtp.lo -MD -MP -MF >>> > >>> .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c -fPIC >>> -DPIC >>> -o >>> > .libs/libfreeswitch_la-switch_rtp.o >>> cc1: warnings being treated as >>> > errors >>> src/switch_rtp.c: In function ?handle_ice?: >>> src/switch_rtp.c:797: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 8 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?read_rtp_packet?: >>> src/switch_rtp.c:3734: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:3906: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 11 has type >>> > ?switch_size_t? >>> src/switch_rtp.c: In function >>> > ?rtp_common_write?: >>> src/switch_rtp.c:5289: error: format ?%ld? expects type >>> > ?long int?, but >>> argument 9 has type ?switch_size_t? >>> src/switch_rtp.c:5293: >>> > error: format ?%ld? expects type ?long int?, but >>> argument 9 has type >>> > ?switch_size_t? >>> >>> make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 >>> make[2]: >>> > Leaving directory `/usr/src/freeswitch' >>> make[1]: *** [all] Error 2 >>> make[1]: >>> > Leaving directory `/usr/src/freeswitch' >>> make: *** [current] Error >>> > 2 >>> >>> Thanks >>> >>> __________________________________________________________________ >>> > _______ >>> Professional FreeSWITCH Consulting >>> > Services: >>> consulting at freeswitch.org [3] >>> http://www.freeswitchsolutions.com [4] >>> >>> FreeSW >>> > ITCH-powered IP PBX: The CudaTel Communication >>> > Server >>> [5] >>> >>> Official FreeSWITCH >>> > Sites >>> http://www.freeswitch.org [6] >>> http://wiki.freeswitch.org [7] >>> http://www.cluecon [8]. >>> > com >>> >>> FreeSWITCH-users mailing >>> > list >>> FreeSWITCH-users at lists.freeswitch.org [9] >>> http://lists.freeswitch.org/mailman [10] >>> > /listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt [11] >>> > ions/freeswitch-users >>> http://www.freeswitch.org [12] >>> >>> -- >>> Ken >>> http://www.FreeSWITCH.org [13] >>> http://www.ClueCon.com [14] >>> http://www.OSTAG.org [15] >>> irc.freenode.net [16] #freeswitch >>> >>> >> >> >> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org [17] >>> http://www.freeswitchsolutions.com [18] >>> >>> >>> [19] >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org [20] >>> http://wiki.freeswitch.org [21] >>> http://www.cluecon.com [22] >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org [23] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [24] >>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> [25] >>> http://www.freeswitch.org [26] >> >> >> >> Links: >> ------ >> [1] mailto:info at shishir.com.np >> [2] mailto:info at shishir.com.np >> [3] mailto:consulting at freeswitch.org >> [4] http://www.freeswitchsolutions.com >> [5] >> [6] http://www.freeswitch.org >> [7] http://wiki.freeswitch.org >> [8] http://www.cluecon >> [9] mailto:FreeSWITCH-users at lists.freeswitch.org >> [10] http://lists.freeswitch.org/mailman >> [11] http://lists.freeswitch.org/mailman/opt >> [12] http://www.freeswitch.org >> [13] http://www.FreeSWITCH.org >> [14] http://www.ClueCon.com >> [15] http://www.OSTAG.org >> [16] http://irc.freenode.net >> [17] mailto:consulting at freeswitch.org >> [18] http://www.freeswitchsolutions.com >> [19] >> [20] http://www.freeswitch.org >> [21] http://wiki.freeswitch.org >> [22] http://www.cluecon.com >> [23] mailto:FreeSWITCH-users at lists.freeswitch.org >> [24] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [25] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [26] http://www.freeswitch.org >> [27] http://jira.freeswitch.org/browse/FS-5248 >> [28] mailto:krice at 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/20130404/6fc17d4f/attachment-0001.html From mytemike72 at gmail.com Thu Apr 4 13:05:53 2013 From: mytemike72 at gmail.com (Michael Lutz) Date: Thu, 4 Apr 2013 11:05:53 +0200 Subject: [Freeswitch-users] Regeneration of DTMF In-Reply-To: References: Message-ID: Hi Avi, Thanks for you response and explanation, the thing is my leg C does't have start_dtmf, I even tried forcing stop_dtmf, and stop_dtmf_generate on leg B and C, but still this happens., It looks like whats happening: Leg A, has inband DTMF and is using start_dtmf. Leg B is bridged with (from) Leg-A, and is not using start_dtmf, when I eavesdrop the leg-B I can only hear dtmf's once.. (so far so good) Leg C dials out to external number, and is not using start_dtmf (unless somewhere 'under water') , and is bridged with leg-B using uuid_bridge. When I eavesdrop leg-C, I can hear dtmf's twice. So somewhere in Leg-B or C based on receiving the generated rfc2833, FS is generating these dtmf's as 'audio'? I would expect 'stop-dtmf_generate' would stop that? (though this is not documented well) So what I need, is either way to stop generating (converting) the inband dtmf to rfc2833, or to stop converting rfc2833 back to inband in leg B ... (I guess?) Good we both know what the issue is, now find someone that can help fix it ... ;-) Regards, Mike. 2013/4/3 Avi Marcus > I can probably explain the issue to you, but I don't really know how to > fix it: > > 1) Leg A comes in with inband. > 2) Your leg B does start_dtmf and detects the inband dtmf. > 3) You bridge to leg C which negotiates rfc2833. It gets the rfc2833 > events from leg B. > > But! start_dtmf can't remove the dtmf from the leg A. So the leg A inband > dtmf is ALSO being passed along. This however is only a problem if leg C > has start_dtmf too. The default dialplan only triggers start_dtmf if there > is no rfc2833 negotiated. But don't count on remote parties to do the > same.... > > > > -Avi Marcus > BestFone > > > On Wed, Apr 3, 2013 at 8:15 PM, Michael Lutz wrote: > >> Hi, >> >> I have a problem, which I am trying to resolve, but can not exactly >> figure out where it is going wrong. >> >> I have an inbound call, this call comes in via SIP and uses inband dtmf, >> at the begining of the dialplan I enable dtmf detection using >> spandsp_start_dtmf. this works fine, and my Lua recognizes digits correctly. >> The tricky part is that I bridge this call in Lua using an api call >> "originate", this call is forwarded to the same switch, and is picked up by >> another Lua script. >> This script, is waiting for a custom event, to end the lua, and is >> bridged with a 3rd call. so the 1st and 3rd call can hear each other. This >> 3rd call is initiated asynchronosly by an esl server. (this al works fine, >> and is not 'the issue'..) >> >> The problem is the receiving end (3rd leg) is receiving the DTMF pressed >> by the 1st leg twice. When I eavesdrop the 2nd leg, i only hear the dtmf >> once, when i eavesdrop the 3rd leg, I can hear the dtmf twice. So it is >> somewhere generated along the way. >> >> I have tryed several different settings, using stop_dtmf_generate on >> different legs, but can not seem to diable this regeneration of this extra >> dtmf. >> >> Any help would be appreciated as this is really causing issues on my >> side, >> >> ps, I know this '3rd leg' principle might look a bit weird, but cannot be >> avoided, >> ps2. When my inbound call comes in using rfc2833, everything works >> perfectly. >> >> >> Best regards, >> Michael Lutz. >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20130404/eba9e1c7/attachment.html From thomas.lee at octon.net Thu Apr 4 13:44:52 2013 From: thomas.lee at octon.net (Thomas Lee) Date: Thu, 4 Apr 2013 02:44:52 -0700 (PDT) Subject: [Freeswitch-users] When doing re-INVITE procedure.... In-Reply-To: References: <1365046457506-7589392.post@n2.nabble.com> Message-ID: <1365068692124-7589397.post@n2.nabble.com> Using some Media Server to make sip call for testing long call duration. Enable Session Timer on Media Server.... UAS <-----------> UAS We found that some Media Server was responding "488 Not acceptable here". Please see the below. (re-INVITE's SDP) v=0 o=hiQ9200 4568120130303170933 1595408506 IN IP4 172.27.14.21 s=Phone Call via hiQ9200 SIPCA c=IN IP4 172.28.66.34 t=0 0 m=audio 35240 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=gpmd:8 vbd=yes a=silenceSupp:off - - - - a=ptime:20 Thanks Regards, Thomas Lee -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/When-doing-re-INVITE-procedure-tp7589392p7589397.html Sent from the freeswitch-users mailing list archive at Nabble.com. From avi at avimarcus.net Thu Apr 4 13:48:03 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 4 Apr 2013 12:48:03 +0300 Subject: [Freeswitch-users] Regeneration of DTMF In-Reply-To: References: Message-ID: Can you get an fs_cli log and/or pcap of the call? Should help understand / see what's going on... -Avi On Thu, Apr 4, 2013 at 12:05 PM, Michael Lutz wrote: > Hi Avi, > > Thanks for you response and explanation, the thing is my leg C does't have > start_dtmf, I even tried forcing stop_dtmf, and stop_dtmf_generate on leg B > and C, but still this happens., > > It looks like whats happening: > Leg A, has inband DTMF and is using start_dtmf. Leg B is bridged with > (from) Leg-A, and is not using start_dtmf, when I eavesdrop the leg-B I can > only hear dtmf's once.. (so far so good) > Leg C dials out to external number, and is not using start_dtmf (unless > somewhere 'under water') , and is bridged with leg-B using > uuid_bridge. When I eavesdrop leg-C, I can hear dtmf's twice. > > So somewhere in Leg-B or C based on receiving the generated rfc2833, FS is > generating these dtmf's as 'audio'? I would expect 'stop-dtmf_generate' > would stop that? (though this is not documented well) > > So what I need, is either way to stop generating (converting) the inband > dtmf to rfc2833, or to stop converting rfc2833 back to inband in leg B ... > (I guess?) > > Good we both know what the issue is, now find someone that can help fix it > ... ;-) > > Regards, > Mike. > > 2013/4/3 Avi Marcus > >> I can probably explain the issue to you, but I don't really know how to >> fix it: >> >> 1) Leg A comes in with inband. >> 2) Your leg B does start_dtmf and detects the inband dtmf. >> 3) You bridge to leg C which negotiates rfc2833. It gets the rfc2833 >> events from leg B. >> >> But! start_dtmf can't remove the dtmf from the leg A. So the leg A inband >> dtmf is ALSO being passed along. This however is only a problem if leg C >> has start_dtmf too. The default dialplan only triggers start_dtmf if there >> is no rfc2833 negotiated. But don't count on remote parties to do the >> same.... >> >> >> >> -Avi Marcus >> BestFone >> >> >> On Wed, Apr 3, 2013 at 8:15 PM, Michael Lutz wrote: >> >>> Hi, >>> >>> I have a problem, which I am trying to resolve, but can not exactly >>> figure out where it is going wrong. >>> >>> I have an inbound call, this call comes in via SIP and uses inband dtmf, >>> at the begining of the dialplan I enable dtmf detection using >>> spandsp_start_dtmf. this works fine, and my Lua recognizes digits correctly. >>> The tricky part is that I bridge this call in Lua using an api call >>> "originate", this call is forwarded to the same switch, and is picked up by >>> another Lua script. >>> This script, is waiting for a custom event, to end the lua, and is >>> bridged with a 3rd call. so the 1st and 3rd call can hear each other. This >>> 3rd call is initiated asynchronosly by an esl server. (this al works fine, >>> and is not 'the issue'..) >>> >>> The problem is the receiving end (3rd leg) is receiving the DTMF pressed >>> by the 1st leg twice. When I eavesdrop the 2nd leg, i only hear the dtmf >>> once, when i eavesdrop the 3rd leg, I can hear the dtmf twice. So it is >>> somewhere generated along the way. >>> >>> I have tryed several different settings, using stop_dtmf_generate on >>> different legs, but can not seem to diable this regeneration of this extra >>> dtmf. >>> >>> Any help would be appreciated as this is really causing issues on my >>> side, >>> >>> ps, I know this '3rd leg' principle might look a bit weird, but cannot >>> be avoided, >>> ps2. When my inbound call comes in using rfc2833, everything works >>> perfectly. >>> >>> >>> Best regards, >>> Michael Lutz. >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20130404/7812b64d/attachment-0001.html From mytemike72 at gmail.com Thu Apr 4 16:31:12 2013 From: mytemike72 at gmail.com (Michael Lutz) Date: Thu, 4 Apr 2013 14:31:12 +0200 Subject: [Freeswitch-users] Regeneration of DTMF In-Reply-To: References: Message-ID: Hi Avi, I could get a pcap of leg-a, and a p-cap of leg-c, but the leg-b is an 'intercal' call to the same switch, by doing an originate to external/phonenumber at localswitchIP They don't appear on the wirehark logs as i suspect it be routed internally as it is recognized as 'self' ... Unless you have other ways of capturing those.. The problem is, I have 6 production switches, all handling traffic, and these are the switches this specific (using inband dtmf) inbound provider routes te traffic to. My test switch, is not able to receive calls from this provider. I have no other provider that can route traffic with just inband dtmf to my test switch.... Mike. 2013/4/4 Avi Marcus > Can you get an fs_cli log and/or pcap of the call? Should help understand > / see what's going on... > -Avi > > > On Thu, Apr 4, 2013 at 12:05 PM, Michael Lutz wrote: > >> Hi Avi, >> >> Thanks for you response and explanation, the thing is my leg C does't >> have start_dtmf, I even tried forcing stop_dtmf, and stop_dtmf_generate on >> leg B and C, but still this happens., >> >> It looks like whats happening: >> Leg A, has inband DTMF and is using start_dtmf. Leg B is bridged with >> (from) Leg-A, and is not using start_dtmf, when I eavesdrop the leg-B I can >> only hear dtmf's once.. (so far so good) >> Leg C dials out to external number, and is not using start_dtmf (unless >> somewhere 'under water') , and is bridged with leg-B using >> uuid_bridge. When I eavesdrop leg-C, I can hear dtmf's twice. >> >> So somewhere in Leg-B or C based on receiving the generated rfc2833, FS >> is generating these dtmf's as 'audio'? I would expect 'stop-dtmf_generate' >> would stop that? (though this is not documented well) >> >> So what I need, is either way to stop generating (converting) the inband >> dtmf to rfc2833, or to stop converting rfc2833 back to inband in leg B ... >> (I guess?) >> >> Good we both know what the issue is, now find someone that can help fix >> it ... ;-) >> >> Regards, >> Mike. >> >> 2013/4/3 Avi Marcus >> >>> I can probably explain the issue to you, but I don't really know how to >>> fix it: >>> >>> 1) Leg A comes in with inband. >>> 2) Your leg B does start_dtmf and detects the inband dtmf. >>> 3) You bridge to leg C which negotiates rfc2833. It gets the rfc2833 >>> events from leg B. >>> >>> But! start_dtmf can't remove the dtmf from the leg A. So the leg A >>> inband dtmf is ALSO being passed along. This however is only a problem if >>> leg C has start_dtmf too. The default dialplan only triggers start_dtmf if >>> there is no rfc2833 negotiated. But don't count on remote parties to do the >>> same.... >>> >>> >>> >>> -Avi Marcus >>> BestFone >>> >>> >>> On Wed, Apr 3, 2013 at 8:15 PM, Michael Lutz wrote: >>> >>>> Hi, >>>> >>>> I have a problem, which I am trying to resolve, but can not exactly >>>> figure out where it is going wrong. >>>> >>>> I have an inbound call, this call comes in via SIP and uses inband >>>> dtmf, at the begining of the dialplan I enable dtmf detection using >>>> spandsp_start_dtmf. this works fine, and my Lua recognizes digits correctly. >>>> The tricky part is that I bridge this call in Lua using an api call >>>> "originate", this call is forwarded to the same switch, and is picked up by >>>> another Lua script. >>>> This script, is waiting for a custom event, to end the lua, and is >>>> bridged with a 3rd call. so the 1st and 3rd call can hear each other. This >>>> 3rd call is initiated asynchronosly by an esl server. (this al works fine, >>>> and is not 'the issue'..) >>>> >>>> The problem is the receiving end (3rd leg) is receiving the DTMF >>>> pressed by the 1st leg twice. When I eavesdrop the 2nd leg, i only hear the >>>> dtmf once, when i eavesdrop the 3rd leg, I can hear the dtmf twice. So it >>>> is somewhere generated along the way. >>>> >>>> I have tryed several different settings, using stop_dtmf_generate on >>>> different legs, but can not seem to diable this regeneration of this extra >>>> dtmf. >>>> >>>> Any help would be appreciated as this is really causing issues on my >>>> side, >>>> >>>> ps, I know this '3rd leg' principle might look a bit weird, but cannot >>>> be avoided, >>>> ps2. When my inbound call comes in using rfc2833, everything works >>>> perfectly. >>>> >>>> >>>> Best regards, >>>> Michael Lutz. >>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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/20130404/1b0aa801/attachment.html From vitaliy.davudov at vts24.ru Thu Apr 4 16:45:53 2013 From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=) Date: Thu, 04 Apr 2013 16:45:53 +0400 Subject: [Freeswitch-users] make current error Message-ID: <515D7601.8030708@vts24.ru> Hi, list! I've installed FreeSWITCH Version 1.2.3+git~20120920T220849Z~f718a5e8e6 (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z) Now, I try to update to the latest tree. While running "make current" in FS source directory, there is an error: ... make[1]: Leaving directory `/usr/src/freeswitch' make all make[1]: Entering directory `/usr/src/freeswitch' CDPATH="${ZSH_VERSION+.}:" && cd . && /bin/sh /usr/src/freeswitch/build/config/missing --run aclocal-1.11 cd . && /bin/sh /usr/src/freeswitch/build/config/missing --run automake-1.11 --foreign CDPATH="${ZSH_VERSION+.}:" && cd . && /bin/sh /usr/src/freeswitch/build/config/missing --run autoconf configure.in:141: error: possibly undefined macro: AC_PROG_LIBTOOL If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. make[1]: *** [configure] Error 1 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 In addition, already installed: # rpm -qa | grep autoconf autoconf-2.62-12.3 # rpm -qa | grep libtool libtool-1.5.22-7.el5_4 What might be the reasons of this error? May be new versions of libtool/autoconf is required? -- Best regards, Vitaly. From steveayre at gmail.com Thu Apr 4 16:56:27 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 4 Apr 2013 13:56:27 +0100 Subject: [Freeswitch-users] make current error In-Reply-To: <515D7601.8030708@vts24.ru> References: <515D7601.8030708@vts24.ru> Message-ID: Try checking out a fresh copy with 'git clone' On 4 April 2013 13:45, ??????? ??????? wrote: > Hi, list! > > I've installed FreeSWITCH Version 1.2.3+git~20120920T220849Z~f718a5e8e6 > (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z) > > Now, I try to update to the latest tree. While running "make current" in > FS source directory, there is an error: > > ... > make[1]: Leaving directory `/usr/src/freeswitch' > make all > make[1]: Entering directory `/usr/src/freeswitch' > CDPATH="${ZSH_VERSION+.}:" && cd . && /bin/sh > /usr/src/freeswitch/build/config/missing --run aclocal-1.11 > cd . && /bin/sh /usr/src/freeswitch/build/config/missing --run > automake-1.11 --foreign > CDPATH="${ZSH_VERSION+.}:" && cd . && /bin/sh > /usr/src/freeswitch/build/config/missing --run autoconf > configure.in:141: error: possibly undefined macro: AC_PROG_LIBTOOL > If this token and others are legitimate, please use > m4_pattern_allow. > See the Autoconf documentation. > make[1]: *** [configure] Error 1 > make[1]: Leaving directory `/usr/src/freeswitch' > make: *** [current] Error 2 > > > In addition, already installed: > > # rpm -qa | grep autoconf > autoconf-2.62-12.3 > # rpm -qa | grep libtool > libtool-1.5.22-7.el5_4 > > What might be the reasons of this error? May be new versions of > libtool/autoconf is required? > > -- > Best regards, > Vitaly. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/d0f13720/attachment-0001.html From shaharhd at gmail.com Thu Apr 4 16:57:12 2013 From: shaharhd at gmail.com (Shahar) Date: Thu, 4 Apr 2013 08:57:12 -0400 Subject: [Freeswitch-users] Spandsp modems as non-root? In-Reply-To: References: <50703D76.7010301@integrafin.co.uk> <515C30EF.8020301@integrafin.co.uk> Message-ID: Andrew, you got me wondering on this ... :) I guest your idea was to hook up the mgetty to the /dev/pts/pts* or the /dev/FS/FS* devices, and then getting *pppd *to listen on those. The only concern I might on this is that the G711 might not get you the fastest dialup speeds (56K) but from what I read online it might support 9.6K ~ 33K (depends on the G711 sample rate your sip provider uses). but in any case, it's worth a try! :) On Wed, Apr 3, 2013 at 3:20 PM, Shahar wrote: > Alex - we're you able to utilize the /dev/FS devices with HylaFax? > > > On Wed, Apr 3, 2013 at 3:19 PM, Shahar wrote: > >> you're asking about using the spandsp softmodem for implementing dialup >> PPP connections? >> >> >> On Wed, Apr 3, 2013 at 9:50 AM, Andrew Cassidy < >> andrew at cassidywebservices.co.uk> wrote: >> >>> (Vaguely) related... I have a potential customer that still uses dial up >>> banking. Which bank still lets them do this I have no idea. Could that be >>> done in this way or in G.711 over SIP? >>> >>> >>> On 3 April 2013 14:38, Alex Crow wrote: >>> >>>> Thanks Shahar (and Nestor who authored the patch)! >>>> >>>> Alex >>>> >>>> >>>> ----Original Message---- >>>> *Subject:* Re: [Freeswitch-users] Spandsp modems as non-root? >>>> *From:* Shahar >>>> *To:* FreeSWITCH Users Help >>>> *CC:* >>>> *Date:* Tue, 2 Apr 2013 19:48:14 -0400 >>>> >>>> Alex, >>>> >>>> I've added another section exactly on this subject. >>>> you might want to have a look at >>>> http://wiki.freeswitch.org/wiki/HylaFax#Modem_devices_permissions_issue >>>> >>>> >>>> >>>> >>>> On Sat, Oct 6, 2012 at 10:17 AM, Alex Crow wrote: >>>> >>>>> Hi, >>>>> >>>>> Has anyone figured out how to get the spandsp emulated modems running >>>>> when Freeswitch is running as a non-root user yet? >>>>> >>>>> I don't even get the /dev/pts/pts* devices added either on debian or >>>>> Ubuntu. >>>>> >>>>> Thanks >>>>> >>>>> Alex >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 has been scanned for viruses and >>>> dangerous content by *MailScanner* , and >>>> is >>>> believed to be clean. >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>>> >>>> >>>> >>>> -- >>>> 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 >>>> 29 Clement's Lane, London EC4N 7AE >>>> 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 >>>> >>>> >>> >>> >>> -- >>> *Andrew Cassidy BSc (Hons) MBCS SSCA* >>> Managing Director >>> >>> >>> *T *03300 100 960 *F >>> *03300 100 961 >>> *E *andrew at cassidywebservices.co.uk >>> *W *www.cassidywebservices.co.uk >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/e4cbeff7/attachment.html From andrew at cassidywebservices.co.uk Thu Apr 4 17:04:34 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 4 Apr 2013 14:04:34 +0100 Subject: [Freeswitch-users] Spandsp modems as non-root? In-Reply-To: References: <50703D76.7010301@integrafin.co.uk> <515C30EF.8020301@integrafin.co.uk> Message-ID: Yeah that's I'm thinking of. It's a little crazy but might give it a try :) On 4 April 2013 13:57, Shahar wrote: > Andrew, > you got me wondering on this ... :) > > I guest your idea was to hook up the mgetty to the /dev/pts/pts* or the > /dev/FS/FS* devices, and then getting *pppd *to listen on those. > > The only concern I might on this is that the G711 might not get you the > fastest dialup speeds (56K) but from what I read online it might support > 9.6K ~ 33K (depends on the G711 sample rate your sip provider uses). > > but in any case, it's worth a try! :) > > > On Wed, Apr 3, 2013 at 3:20 PM, Shahar wrote: > >> Alex - we're you able to utilize the /dev/FS devices with HylaFax? >> >> >> On Wed, Apr 3, 2013 at 3:19 PM, Shahar wrote: >> >>> you're asking about using the spandsp softmodem for implementing dialup >>> PPP connections? >>> >>> >>> On Wed, Apr 3, 2013 at 9:50 AM, Andrew Cassidy < >>> andrew at cassidywebservices.co.uk> wrote: >>> >>>> (Vaguely) related... I have a potential customer that still uses dial >>>> up banking. Which bank still lets them do this I have no idea. Could that >>>> be done in this way or in G.711 over SIP? >>>> >>>> >>>> On 3 April 2013 14:38, Alex Crow wrote: >>>> >>>>> Thanks Shahar (and Nestor who authored the patch)! >>>>> >>>>> Alex >>>>> >>>>> >>>>> ----Original Message---- >>>>> *Subject:* Re: [Freeswitch-users] Spandsp modems as non-root? >>>>> *From:* Shahar >>>>> *To:* FreeSWITCH Users Help >>>>> *CC:* >>>>> *Date:* Tue, 2 Apr 2013 19:48:14 -0400 >>>>> >>>>> Alex, >>>>> >>>>> I've added another section exactly on this subject. >>>>> you might want to have a look at >>>>> http://wiki.freeswitch.org/wiki/HylaFax#Modem_devices_permissions_issue >>>>> >>>>> >>>>> >>>>> >>>>> On Sat, Oct 6, 2012 at 10:17 AM, Alex Crow wrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> Has anyone figured out how to get the spandsp emulated modems running >>>>>> when Freeswitch is running as a non-root user yet? >>>>>> >>>>>> I don't even get the /dev/pts/pts* devices added either on debian or >>>>>> Ubuntu. >>>>>> >>>>>> Thanks >>>>>> >>>>>> Alex >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 has been scanned for viruses and >>>>> dangerous content by *MailScanner* , >>>>> and is >>>>> believed to be clean. >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> 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 >>>>> 29 Clement's Lane, London EC4N 7AE >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> *Andrew Cassidy BSc (Hons) MBCS SSCA* >>>> Managing Director >>>> >>>> >>>> *T *03300 100 960 *F >>>> *03300 100 961 >>>> *E *andrew at cassidywebservices.co.uk >>>> *W *www.cassidywebservices.co.uk >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/28f948d8/attachment-0001.html From tnsampaio at bsd.com.br Thu Apr 4 17:17:37 2013 From: tnsampaio at bsd.com.br (Tiago Sampaio) Date: Thu, 04 Apr 2013 10:17:37 -0300 Subject: [Freeswitch-users] mod_callcenter + eavesdrop Message-ID: <515D7D71.8020304@bsd.com.br> Hi ALL.. Im currently using eavesdroping on outgoing calls using: and is working fine! Now I want to hear calls from my agents, but i dun know how to insert in spymap what agent get the call from queue, any tips? im using options from callcenter.conf: Another question is: How to put in cdr lost calls, example: queue support have agent1 and agent2. Incomming call, first ring to agent1 for 10sec and then ring and answer agent2. I wanna log the event of not answer from agent1 Im thinking about write an application to listen esl an caputure the events about callcenter and then log to cdr, am i write? Hugs! From steveu at coppice.org Thu Apr 4 18:42:56 2013 From: steveu at coppice.org (Steve Underwood) Date: Thu, 04 Apr 2013 22:42:56 +0800 Subject: [Freeswitch-users] Spandsp modems as non-root? In-Reply-To: References: <50703D76.7010301@integrafin.co.uk> <515C30EF.8020301@integrafin.co.uk> Message-ID: <515D9170.9090208@coppice.org> On 04/03/2013 09:50 PM, Andrew Cassidy wrote: > (Vaguely) related... I have a potential customer that still uses dial > up banking. Which bank still lets them do this I have no idea. Could > that be done in this way or in G.711 over SIP? Can this be done with mod_spandsp? No. Can this be done with G.711 over SIP? If everything works perfectly, then kinda, maybe. Regards, Steve > > > On 3 April 2013 14:38, Alex Crow > wrote: > > Thanks Shahar (and Nestor who authored the patch)! > > Alex > > > ----Original Message---- > *Subject:* Re: [Freeswitch-users] Spandsp modems as non-root? > *From:* Shahar > *To:* FreeSWITCH Users Help > > > *CC:* > *Date:* Tue, 2 Apr 2013 19:48:14 -0400 > >> Alex, >> >> I've added another section exactly on this subject. >> you might want to have a look at >> http://wiki.freeswitch.org/wiki/HylaFax#Modem_devices_permissions_issue >> >> >> >> >> On Sat, Oct 6, 2012 at 10:17 AM, Alex Crow >> > wrote: >> >> Hi, >> >> Has anyone figured out how to get the spandsp emulated modems >> running >> when Freeswitch is running as a non-root user yet? >> >> I don't even get the /dev/pts/pts* devices added either on >> debian or Ubuntu. >> >> Thanks >> >> Alex >> From fs at tcowan.net Thu Apr 4 19:06:11 2013 From: fs at tcowan.net (fs at tcowan.net) Date: Thu, 04 Apr 2013 11:06:11 -0400 Subject: [Freeswitch-users] =?utf-8?q?record=5Fsession_wav_files_are_corru?= =?utf-8?q?pted?= In-Reply-To: References: Message-ID: On 04-02-2013 05:17, Steven Schoch wrote: > On Tue, Apr 2, 2013 at 10:28 AM, wrote: > >> ?If I try to import the >> audio as raw format in audacity you can hear some high pitched >> talking >> that plays back very quickly. > > When you import as raw data, try changing the sample rate to 8000. > > The reason Audacity can't open the file is because the header, which > specifies the encoding, channels, sample rate, etc. is damaged or > missing. > > --? > Steve? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Well why would the header to damaged or missing. I have updated the config to the following and it still doesnt work. If I look at the header info with windows by going to file properties it doesnt list anything in the header info. The file is just a bunch of high pitched noise if I dont import as raw. I can hear the audio fine when importing raw and 8000hz rate. I am not sure what else I need to do to make this work. I followed the steps listed on: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session From anthony.minessale at gmail.com Thu Apr 4 19:30:32 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 4 Apr 2013 10:30:32 -0500 Subject: [Freeswitch-users] record_session wav files are corrupted In-Reply-To: References: Message-ID: I just tested this and it's working for me. Please update to the latest code and reproduce, if you still have the issue report it as a jira to http://jira.freeswitch.org Include a full console trace of the offending call and the example file. sofia global siptrace on sofia tracelevel alert console loglevel debug On Thu, Apr 4, 2013 at 10:06 AM, wrote: > On 04-02-2013 05:17, Steven Schoch wrote: > > On Tue, Apr 2, 2013 at 10:28 AM, wrote: > > > >> If I try to import the > >> audio as raw format in audacity you can hear some high pitched > >> talking > >> that plays back very quickly. > > > > When you import as raw data, try changing the sample rate to 8000. > > > > The reason Audacity can't open the file is because the header, which > > specifies the encoding, channels, sample rate, etc. is damaged or > > missing. > > > > -- > > Steve > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > Well why would the header to damaged or missing. I have updated the > config to the following and it still doesnt work. If I look at the > header info with windows by going to file properties it doesnt list > anything in the header info. The file is just a bunch of high pitched > noise if I dont import as raw. I can hear the audio fine when importing > raw and 8000hz rate. I am not sure what else I need to do to make this > work. > > I followed the steps listed on: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session > > > > data="sip_h_X-accountcode=${accountcode}" /> > > > > data="effective_caller_id_name=${outbound_caller_id_name}" /> > data="effective_caller_id_number=${outbound_caller_id_number}" /> > > > /> > > > > > > > > data="$${base_dir}/recordings/${domain_name}/outbound10d-${strftime(%m-%d-%Y-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav" > /> > data="sofia/gateway/customer1-bandwidth/$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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130404/53ae8016/attachment.html From lists at telefaks.de Thu Apr 4 19:55:03 2013 From: lists at telefaks.de (Peter Steinbach) Date: Thu, 04 Apr 2013 17:55:03 +0200 Subject: [Freeswitch-users] Set User-to-user header Message-ID: <515DA257.50409@telefaks.de> In http://jira.freeswitch.org/browse/FS-4994 I have seen, that the User-to-User header can be passed into a channel variable. But what obout the other direction? Is there a way to set this header on an outbound call? Or do I just have to prefix with "sip_h_"? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From ashish at nms.co.in Thu Apr 4 09:27:38 2013 From: ashish at nms.co.in (Ashish gautam) Date: Thu, 4 Apr 2013 10:57:38 +0530 Subject: [Freeswitch-users] FreeTDM channel restart error Message-ID: Hi, Whenever I restart FreeSWITCH I get the following channel restart errors from mod_ftdm: 2013-04-04 10:35:43.534528 [NOTICE] ftmod_libpri.c:2159 [s1c4][1:4] -- Restart of channel completed 2013-04-04 10:35:43.534528 [WARNING] ftdm_io.c:3018 [s1c4][1:4] Channel not opened, proceeding anyway 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c5][1:5] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c6][1:6] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c7][1:7] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c8][1:8] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c9][1:9] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c10][1:10] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c11][1:11] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c12][1:12] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c13][1:13] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c14][1:14] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c15][1:15] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c17][1:17] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c18][1:18] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c19][1:19] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c20][1:20] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c21][1:21] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c22][1:22] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c23][1:23] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c24][1:24] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c25][1:25] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c26][1:26] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c27][1:27] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c28][1:28] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c29][1:29] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c30][1:30] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c31][1:31] -- T316 timed out, channel reached restart attempt limit '3' and is suspended It makes 3 attempts to restart all the 30 channels on span 1 of the PRI card and then show these errors. Please anybody help me getting out of this. Thanks in advance. Regards. -Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/50bba59b/attachment.html From bhavikpatel14388 at gmail.com Thu Apr 4 12:52:23 2013 From: bhavikpatel14388 at gmail.com (bhavik patel) Date: Wed, 3 Apr 2013 22:52:23 -1000 Subject: [Freeswitch-users] freeswitch say application currency in multipal language Message-ID: Hi all, I use free switch and i want to play sounds file like if user has credit in USD then doller.wav file play and EUR then another file will be play. Currently it play doller.wav by default in /usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play EUR so how can this possible. Is that any easy way to do this thing in multi language currency play in say application. i use this syntax in my free-switch dial plan $dialstring = ""; Thanks In advance... -- Thanks, Bhavik Patel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130403/0fd81591/attachment.html From ashish at nms.co.in Thu Apr 4 13:25:37 2013 From: ashish at nms.co.in (Ashish gautam) Date: Thu, 4 Apr 2013 14:55:37 +0530 Subject: [Freeswitch-users] Call not originated through event socket Message-ID: Hi, I am generating calls through event socket using originate action to multiple numbers to PSTN network. ES is returning '+OK' for the call not generated. Still the call is not generated. Any help is appreciated. Thanks. -Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/f5b94c3c/attachment.html From adrian.fuentes at hovanetworks.com Thu Apr 4 20:47:19 2013 From: adrian.fuentes at hovanetworks.com (Adrian Fuentes) Date: Thu, 4 Apr 2013 10:47:19 -0600 Subject: [Freeswitch-users] Can be modified switch_m_sdp? In-Reply-To: References: Message-ID: <0FDA9B76-A518-4871-9445-DF951809BFB4@hovanetworks.com> Ken Thanks for reply. I have this scenario IPPHONE --> FreeSwitch Bypass --> Other-SoftSwitch --> Gateway Internet --> Public IP --> Local IP --> IP Local & Public I want to change (to c = IN IP4 172.25.1.9) by (to c = IN IP4 200.56.93.77), to be able to bypass between IPPHONE and Gateway. This is because "Other-SoftSwitch" can not respond Gateway Public IP in SDP (to c = IN IP4 200.56.93.77). Adrian Fuentes El 04/04/2013, a las 02:03, freeswitch-users-request at lists.freeswitch.org escribi?: > > De: Ken Rice > Asunto: Re: [Freeswitch-users] Can be modified switch_m_sdp? > Fecha: 3 de abril de 2013 19:07:29 CST > Para: FreeSWITCH Users Help > Responder a: FreeSWITCH Users Help > > > what are you trying to do? you need to be specific... you can modify the sdp possibly but the end goal tells us how to advise this > > Ken > Sent from my iPad > > On Apr 3, 2013, at 19:18, Adrian Fuentes wrote: > >> Hi all >> >> Is it possible to replace the sdp c = IN IP4 172.25.1.9 through 172.25.1.9 c = IN IP4? >> >> variable_switch_m_sdp: [v=0 >> o=- 13649160524104272291 13649160524104272291 IN IP4 10.5.86.99 >> s=it-ssw >> c=IN IP4 172.25.1.9 >> t=0 0 >> m=audio 6150 RTP/AVP 8 101 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> a=silenceSupp:off - - - - >> ] >> >> Regards >> Adrian Fuentes >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/72ec2731/attachment.html From schoch+freeswitch.org at xwin32.com Thu Apr 4 21:01:50 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 4 Apr 2013 10:01:50 -0700 Subject: [Freeswitch-users] Spandsp modems as non-root? In-Reply-To: References: <50703D76.7010301@integrafin.co.uk> Message-ID: On Tue, Apr 2, 2013 at 4:48 PM, Shahar wrote: > you might want to have a look at > http://wiki.freeswitch.org/wiki/HylaFax#Modem_devices_permissions_issue > That's great! ... and better than my solution, which was: 1. Create a 'dev' group in /etc/group with this line: dev:x:11:freeswitch 2. Modify the startup script by adding this line before the 'daemon' line: chgrp dev /dev; chmod g+w /dev Now that this patch is in, I can avoid giving the 'freeswitch' user more permissions than it needs. I also modified the start script to make the modem devices work with HylaFax by adding this line after the 'daemon' line: (sleep 120; chgrp uucp /dev/FS*; chmod g+r /dev/FS*) & This allows HylaFAX, which runs in the 'uucp' group, to access the modems. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/2b147a31/attachment-0001.html From schoch+freeswitch.org at xwin32.com Thu Apr 4 21:26:08 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 4 Apr 2013 10:26:08 -0700 Subject: [Freeswitch-users] One of my extensions Caller ID says Anonymous! In-Reply-To: References: <0c0801ce3089$839fe4f0$8adfaed0$@bizfocused.com> <0e6f01ce30aa$ebb5b210$c3211630$@bizfocused.com> Message-ID: In my directory, I set the variable "effective_caller_id_name" instead of "internal_caller_id_name". (I also set the "outbound_caller_id_name" variable just like your example, although that never seems to get sent to the PSTN.) -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/c3d8de74/attachment.html From vipkilla at gmail.com Thu Apr 4 21:37:58 2013 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 4 Apr 2013 13:37:58 -0400 Subject: [Freeswitch-users] One of my extensions Caller ID says Anonymous! In-Reply-To: References: <0c0801ce3089$839fe4f0$8adfaed0$@bizfocused.com> <0e6f01ce30aa$ebb5b210$c3211630$@bizfocused.com> Message-ID: I think he resolved this in the Cisco configuration. I'm having the same problem on the Cisco, I can't figure out which configuration parameter needs to be changed though. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/46d98469/attachment.html From Hector.Geraldino at ipsoft.com Thu Apr 4 21:39:07 2013 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Thu, 4 Apr 2013 17:39:07 +0000 Subject: [Freeswitch-users] Call not originated through event socket In-Reply-To: References: Message-ID: +OK doesn't mean the call was generated, it just means that FreeSWITCH received (and processed) the command. You should probably be better served by listening to the events for each one of these calls. There are a few handful events that can tell you if the channel was created, the call was originated, if it was answered and when it was dropped with the drop reason. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ashish gautam Sent: Thursday, April 04, 2013 5:26 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Call not originated through event socket Hi, I am generating calls through event socket using originate action to multiple numbers to PSTN network. ES is returning '+OK' for the call not generated. Still the call is not generated. Any help is appreciated. Thanks. -Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/72c7d14a/attachment.html From msc at freeswitch.org Thu Apr 4 21:41:10 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Apr 2013 10:41:10 -0700 Subject: [Freeswitch-users] Regeneration of DTMF In-Reply-To: References: Message-ID: Can you supply a simple set of configs that we could plug into a default freeswitch install and see if we can lab it up ourselves? Perhaps if we can see it in a controlled environment it would be easier to diagnose. -MC On Thu, Apr 4, 2013 at 5:31 AM, Michael Lutz wrote: > Hi Avi, > > I could get a pcap of leg-a, and a p-cap of leg-c, but the leg-b is an > 'intercal' call to the same switch, by doing an originate to > external/phonenumber at localswitchIP > They don't appear on the wirehark logs as i suspect it be routed > internally as it is recognized as 'self' ... > Unless you have other ways of capturing those.. > > The problem is, I have 6 production switches, all handling traffic, and > these are the switches this specific (using inband dtmf) inbound provider > routes te traffic to. > My test switch, is not able to receive calls from this provider. I have no > other provider that can route traffic with just inband dtmf to my test > switch.... > > Mike. > > 2013/4/4 Avi Marcus > >> Can you get an fs_cli log and/or pcap of the call? Should help understand >> / see what's going on... >> -Avi >> >> >> On Thu, Apr 4, 2013 at 12:05 PM, Michael Lutz wrote: >> >>> Hi Avi, >>> >>> Thanks for you response and explanation, the thing is my leg C does't >>> have start_dtmf, I even tried forcing stop_dtmf, and stop_dtmf_generate on >>> leg B and C, but still this happens., >>> >>> It looks like whats happening: >>> Leg A, has inband DTMF and is using start_dtmf. Leg B is bridged with >>> (from) Leg-A, and is not using start_dtmf, when I eavesdrop the leg-B I can >>> only hear dtmf's once.. (so far so good) >>> Leg C dials out to external number, and is not using start_dtmf (unless >>> somewhere 'under water') , and is bridged with leg-B using >>> uuid_bridge. When I eavesdrop leg-C, I can hear dtmf's twice. >>> >>> So somewhere in Leg-B or C based on receiving the generated rfc2833, FS >>> is generating these dtmf's as 'audio'? I would expect 'stop-dtmf_generate' >>> would stop that? (though this is not documented well) >>> >>> So what I need, is either way to stop generating (converting) the inband >>> dtmf to rfc2833, or to stop converting rfc2833 back to inband in leg B ... >>> (I guess?) >>> >>> Good we both know what the issue is, now find someone that can help fix >>> it ... ;-) >>> >>> Regards, >>> Mike. >>> >>> 2013/4/3 Avi Marcus >>> >>>> I can probably explain the issue to you, but I don't really know how to >>>> fix it: >>>> >>>> 1) Leg A comes in with inband. >>>> 2) Your leg B does start_dtmf and detects the inband dtmf. >>>> 3) You bridge to leg C which negotiates rfc2833. It gets the rfc2833 >>>> events from leg B. >>>> >>>> But! start_dtmf can't remove the dtmf from the leg A. So the leg A >>>> inband dtmf is ALSO being passed along. This however is only a problem if >>>> leg C has start_dtmf too. The default dialplan only triggers start_dtmf if >>>> there is no rfc2833 negotiated. But don't count on remote parties to do the >>>> same.... >>>> >>>> >>>> >>>> -Avi Marcus >>>> BestFone >>>> >>>> >>>> On Wed, Apr 3, 2013 at 8:15 PM, Michael Lutz wrote: >>>> >>>>> Hi, >>>>> >>>>> I have a problem, which I am trying to resolve, but can not exactly >>>>> figure out where it is going wrong. >>>>> >>>>> I have an inbound call, this call comes in via SIP and uses inband >>>>> dtmf, at the begining of the dialplan I enable dtmf detection using >>>>> spandsp_start_dtmf. this works fine, and my Lua recognizes digits correctly. >>>>> The tricky part is that I bridge this call in Lua using an api call >>>>> "originate", this call is forwarded to the same switch, and is picked up by >>>>> another Lua script. >>>>> This script, is waiting for a custom event, to end the lua, and is >>>>> bridged with a 3rd call. so the 1st and 3rd call can hear each other. This >>>>> 3rd call is initiated asynchronosly by an esl server. (this al works fine, >>>>> and is not 'the issue'..) >>>>> >>>>> The problem is the receiving end (3rd leg) is receiving the DTMF >>>>> pressed by the 1st leg twice. When I eavesdrop the 2nd leg, i only hear the >>>>> dtmf once, when i eavesdrop the 3rd leg, I can hear the dtmf twice. So it >>>>> is somewhere generated along the way. >>>>> >>>>> I have tryed several different settings, using stop_dtmf_generate on >>>>> different legs, but can not seem to diable this regeneration of this extra >>>>> dtmf. >>>>> >>>>> Any help would be appreciated as this is really causing issues on my >>>>> side, >>>>> >>>>> ps, I know this '3rd leg' principle might look a bit weird, but cannot >>>>> be avoided, >>>>> ps2. When my inbound call comes in using rfc2833, everything works >>>>> perfectly. >>>>> >>>>> >>>>> Best regards, >>>>> Michael Lutz. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/9b01e225/attachment-0001.html From sdevoy at bizfocused.com Thu Apr 4 21:59:41 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 4 Apr 2013 13:59:41 -0400 Subject: [Freeswitch-users] One of my extensions Caller ID says Anonymous! In-Reply-To: References: <0c0801ce3089$839fe4f0$8adfaed0$@bizfocused.com> <0e6f01ce30aa$ebb5b210$c3211630$@bizfocused.com> Message-ID: <008e01ce315e$2e86dc00$8b949400$@bizfocused.com> I am still searching. I have not found the cofig param yet either. The single device doing it is behind a firewall so I can get in to see the configuration there! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vik Killa Sent: Thursday, April 04, 2013 1:38 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One of my extensions Caller ID says Anonymous! I think he resolved this in the Cisco configuration. I'm having the same problem on the Cisco, I can't figure out which configuration parameter needs to be changed though. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/5205e401/attachment.html From msc at freeswitch.org Thu Apr 4 22:07:23 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Apr 2013 11:07:23 -0700 Subject: [Freeswitch-users] Mod Managed Conf Call In-Reply-To: <20130403232334.a4d0f38d@mail.tritonwest.net> References: <20130403232334.a4d0f38d@mail.tritonwest.net> Message-ID: FYI, we have this tentatively scheduled for next Wed April 10. I will be sending out more information tonight or tomorrow. Thanks, Michael On Wed, Apr 3, 2013 at 4:23 PM, Dave R. Kompel wrote: > ** > I can do a shorter one "inband" but thas was going to be out of band (not > on a meeting) that would be more intense, including having ppl follow along > and build full application themself. > > --Dave > > > ------------------------------ > *From:* Ken Rice [mailto:krice at freeswitch.org] > *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > *Sent:* Wed, 03 Apr 2013 12:10:13 -0700 > *Subject:* Re: [Freeswitch-users] Mod Managed Conf Call > > Its going to be inband... Talk to Collins > > > On 4/3/13 12:50 PM, "Dave R. Kompel" > > wrote: > > Ken this was going to be out of band. Sorry for the delay. My hand had a > dispute with a knife, and it's out of comission for about a week, so It's > going to have to wait till next week, when I'm back to tying. > > Can everyoine that want's to be included, e-mail me directly > drk at drkngs.net and we can schedulel a time. I see the doctor on Friday, so I should be > able to type normal after that appointment. > > --Dave > > ------------------------------ > *From:* Ken Rice [mailto:krice at freeswitch.org ] > *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org > ] > *Cc:* > > [mailto:freeswitch-users at lists.freeswitch.org > ] > *Sent:* Tue, 02 Apr 2013 20:25:25 -0700 > *Subject:* Re: [Freeswitch-users] Mod Managed Conf Call > > it will happen soon during the regular weekly conference call > > Ken > Sent from my iPad > > On Apr 2, 2013, at 22:14, "Usama Zaidi" > > wrote: > > > Hey, > > > > Any updates on when the conference call is going to be scheduled for? The > > thread kinda died. > > > > Regards. > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] > On Behalf Of > > freeswitch-users-request at lists.freeswitch.org > > Sent: Tuesday, April 02, 2013 11:34 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: FreeSWITCH-users Digest, Vol 82, Issue 19 > > > > 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..." > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/0d19e4a5/attachment.html From msc at freeswitch.org Thu Apr 4 22:08:33 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Apr 2013 11:08:33 -0700 Subject: [Freeswitch-users] Call not originated through event socket In-Reply-To: References: Message-ID: You could also just hop on fs_cli and watch the output while you make a test call with the event socket. Most likely you'll see what's going on. -MC On Thu, Apr 4, 2013 at 10:39 AM, Hector Geraldino < Hector.Geraldino at ipsoft.com> wrote: > +OK doesn?t mean the call was generated, it just means that FreeSWITCH > received (and processed) the command.**** > > ** ** > > You should probably be better served by listening to the events for each > one of these calls. There are a few handful events that can tell you if the > channel was created, the call was originated, if it was answered and when > it was dropped with the drop reason.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish > gautam > *Sent:* Thursday, April 04, 2013 5:26 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Call not originated through event socket**** > > ** ** > > Hi,**** > > ** ** > > I am generating calls through event socket using originate action to > multiple numbers to PSTN network. ES is returning '+OK' for the call not > generated. Still the call is not generated.**** > > ** ** > > Any help is appreciated.**** > > ** ** > > Thanks.**** > > ** ** > > -Ashish**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/b291e6cb/attachment-0001.html From vipkilla at gmail.com Thu Apr 4 22:13:26 2013 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 4 Apr 2013 14:13:26 -0400 Subject: [Freeswitch-users] One of my extensions Caller ID says Anonymous! In-Reply-To: <008e01ce315e$2e86dc00$8b949400$@bizfocused.com> References: <0c0801ce3089$839fe4f0$8adfaed0$@bizfocused.com> <0e6f01ce30aa$ebb5b210$c3211630$@bizfocused.com> <008e01ce315e$2e86dc00$8b949400$@bizfocused.com> Message-ID: Please let us know if and how you get it resolved. I'll do the same if I can figure it out. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/742a610d/attachment.html From msc at freeswitch.org Thu Apr 4 22:14:52 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Apr 2013 11:14:52 -0700 Subject: [Freeswitch-users] freeswitch say application currency in multipal language In-Reply-To: References: Message-ID: I don't believe that there is currently a way to do this easily right now. We just spoke about languages on yesterday's conference call and this is a prime example of the kinds of things that we will need to overcome. Additionally I don't believe that I have any currencies other than dollar.wav and dollars.wav for the English sounds. I'll be glad to get them ordered. Could the community at large send me some ideas for units of currency? Here are a few ideas: euro, euros franc, francs Canadian, Australian, US dollar/dollars pound, pounds Send me some more ideas and I will get them added to the to-be-recorded list. -MC On Thu, Apr 4, 2013 at 1:52 AM, bhavik patel wrote: > Hi all, > I use free switch and i want to play sounds file like if user has credit > in USD then doller.wav file play and EUR then another file will be play. > > Currently it play doller.wav by default in > /usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play EUR > so how can this possible. > > Is that any easy way to do this thing in multi language currency play in > say application. > > i use this syntax in my free-switch dial plan > $dialstring = " $credit_balance\"/>"; > > Thanks In advance... > > -- > Thanks, > Bhavik Patel > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/4a62f89b/attachment.html From avi at avimarcus.net Thu Apr 4 22:16:04 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 4 Apr 2013 21:16:04 +0300 Subject: [Freeswitch-users] One of my extensions Caller ID says Anonymous! In-Reply-To: References: <0c0801ce3089$839fe4f0$8adfaed0$@bizfocused.com> <0e6f01ce30aa$ebb5b210$c3211630$@bizfocused.com> Message-ID: Indeed, outbound_caller_id_name is just a variable. To use it you have to tell the dialplan to set effective_caller_id_name=${ outbound_caller_id_name}. effective_caller_id_name IS a variable that actually gets used. (The default dialplan does this.) -Avi Marcus BestFone On Thu, Apr 4, 2013 at 8:26 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > In my directory, I set the variable "effective_caller_id_name" instead of > "internal_caller_id_name". (I also set the "outbound_caller_id_name" > variable just like your example, although that never seems to get sent to > the PSTN.) > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/61b8eb8f/attachment.html From steveayre at gmail.com Thu Apr 4 22:46:22 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 4 Apr 2013 19:46:22 +0100 Subject: [Freeswitch-users] Call not originated through event socket In-Reply-To: References: Message-ID: +1 '/log 9' and watch the debug output. It'll be very clear about what's happening. Most likely there's something wrong about your usage of originate. -Steve On 4 April 2013 19:08, Michael Collins wrote: > You could also just hop on fs_cli and watch the output while you make a > test call with the event socket. Most likely you'll see what's going on. > > -MC > > On Thu, Apr 4, 2013 at 10:39 AM, Hector Geraldino < > Hector.Geraldino at ipsoft.com> wrote: > >> +OK doesn?t mean the call was generated, it just means that FreeSWITCH >> received (and processed) the command.**** >> >> ** ** >> >> You should probably be better served by listening to the events for each >> one of these calls. There are a few handful events that can tell you if the >> channel was created, the call was originated, if it was answered and when >> it was dropped with the drop reason.**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish >> gautam >> *Sent:* Thursday, April 04, 2013 5:26 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] Call not originated through event socket*** >> * >> >> ** ** >> >> Hi,**** >> >> ** ** >> >> I am generating calls through event socket using originate action to >> multiple numbers to PSTN network. ES is returning '+OK' for the call not >> generated. Still the call is not generated.**** >> >> ** ** >> >> Any help is appreciated.**** >> >> ** ** >> >> Thanks.**** >> >> ** ** >> >> -Ashish**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.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/20130404/a0c3843d/attachment-0001.html From drk at drkngs.net Thu Apr 4 22:52:16 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Thu, 04 Apr 2013 11:52:16 -0700 Subject: [Freeswitch-users] Mod Managed Conf Call In-Reply-To: Message-ID: <20130404185216.cb7c3077@mail.tritonwest.net> Were talkiing about doing the full bore one, not a scalled down one. E-mail to follow for what you will need if you want to follow along. --Dave _____ From: Michael Collins [mailto:msc at freeswitch.org] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Thu, 04 Apr 2013 11:07:23 -0700 Subject: Re: [Freeswitch-users] Mod Managed Conf Call FYI, we have this tentatively scheduled for next Wed April 10. I will be sending out more information tonight or tomorrow. Thanks, Michael On Wed, Apr 3, 2013 at 4:23 PM, Dave R. Kompel wrote: I can do a shorter one "inband" but thas was going to be out of band (not on a meeting) that would be more intense, including having ppl follow along and build full application themself. --Dave _____ From: Ken Rice [mailto:krice at freeswitch.org] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Wed, 03 Apr 2013 12:10:13 -0700 Subject: Re: [Freeswitch-users] Mod Managed Conf Call Its going to be inband... Talk to Collins On 4/3/13 12:50 PM, "Dave R. Kompel" wrote: Ken this was going to be out of band. Sorry for the delay. My hand had a dispute with a knife, and it's out of comission for about a week, so It's going to have to wait till next week, when I'm back to tying. Can everyoine that want's to be included, e-mail me directly drk at drkngs.net and we can schedulel a time. I see the doctor on Friday, so I should be able to type normal after that appointment. --Dave _____ From: Ken Rice [mailto:krice at freeswitch.org] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Cc: [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 02 Apr 2013 20:25:25 -0700 Subject: Re: [Freeswitch-users] Mod Managed Conf Call it will happen soon during the regular weekly conference call Ken Sent from my iPad On Apr 2, 2013, at 22:14, "Usama Zaidi" wrote: > Hey, > > Any updates on when the conference call is going to be scheduled for? The > thread kinda died. > > Regards. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > freeswitch-users-request at lists.freeswitch.org > Sent: Tuesday, April 02, 2013 11:34 PM > To: freeswitch-users at lists.freeswitch.org > Subject: FreeSWITCH-users Digest, Vol 82, Issue 19 > > 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..." > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/94809255/attachment.html From shaharhd at gmail.com Thu Apr 4 23:17:25 2013 From: shaharhd at gmail.com (Shahar) Date: Thu, 4 Apr 2013 15:17:25 -0400 Subject: [Freeswitch-users] Spandsp modems as non-root? In-Reply-To: References: <50703D76.7010301@integrafin.co.uk> Message-ID: well... the directory setting always existed in spandsp - it was just undocumented :) On Thu, Apr 4, 2013 at 1:01 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Tue, Apr 2, 2013 at 4:48 PM, Shahar wrote: > >> you might want to have a look at >> http://wiki.freeswitch.org/wiki/HylaFax#Modem_devices_permissions_issue >> > > That's great! ... and better than my solution, which was: > > 1. Create a 'dev' group in /etc/group with this line: > dev:x:11:freeswitch > 2. Modify the startup script by adding this line before the 'daemon' line: > chgrp dev /dev; chmod g+w /dev > > Now that this patch is in, I can avoid giving the 'freeswitch' user more > permissions than it needs. > > I also modified the start script to make the modem devices work with > HylaFax by adding this line after the 'daemon' line: > > (sleep 120; chgrp uucp /dev/FS*; chmod g+r /dev/FS*) & > > This allows HylaFAX, which runs in the 'uucp' group, to access the modems. > > -- > Steve > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/69ef864b/attachment-0001.html From shaharhd at gmail.com Thu Apr 4 23:19:43 2013 From: shaharhd at gmail.com (Shahar) Date: Thu, 4 Apr 2013 15:19:43 -0400 Subject: [Freeswitch-users] Spandsp modems as non-root? In-Reply-To: <515D9170.9090208@coppice.org> References: <50703D76.7010301@integrafin.co.uk> <515C30EF.8020301@integrafin.co.uk> <515D9170.9090208@coppice.org> Message-ID: were not thinking about utilizing spandsp for the dialup - we just want it to create the com devices. then were passing the handling via mgetty to pppd - just like hylafax takes over the spandsp in the data handling. maybe we need to disable the T.38 for this to work (and stay in G711) ? On Thu, Apr 4, 2013 at 10:42 AM, Steve Underwood wrote: > On 04/03/2013 09:50 PM, Andrew Cassidy wrote: > > (Vaguely) related... I have a potential customer that still uses dial > > up banking. Which bank still lets them do this I have no idea. Could > > that be done in this way or in G.711 over SIP? > > Can this be done with mod_spandsp? No. > > Can this be done with G.711 over SIP? If everything works perfectly, > then kinda, maybe. > > Regards, > Steve > > > > > > On 3 April 2013 14:38, Alex Crow > > wrote: > > > > Thanks Shahar (and Nestor who authored the patch)! > > > > Alex > > > > > > ----Original Message---- > > *Subject:* Re: [Freeswitch-users] Spandsp modems as non-root? > > *From:* Shahar > > *To:* FreeSWITCH Users Help > > > > > > *CC:* > > *Date:* Tue, 2 Apr 2013 19:48:14 -0400 > > > >> Alex, > >> > >> I've added another section exactly on this subject. > >> you might want to have a look at > >> > http://wiki.freeswitch.org/wiki/HylaFax#Modem_devices_permissions_issue > >> > >> > >> > >> > >> On Sat, Oct 6, 2012 at 10:17 AM, Alex Crow > >> > wrote: > >> > >> Hi, > >> > >> Has anyone figured out how to get the spandsp emulated modems > >> running > >> when Freeswitch is running as a non-root user yet? > >> > >> I don't even get the /dev/pts/pts* devices added either on > >> debian or Ubuntu. > >> > >> Thanks > >> > >> Alex > >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/2411aa79/attachment.html From schoch+freeswitch.org at xwin32.com Thu Apr 4 23:28:21 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 4 Apr 2013 12:28:21 -0700 Subject: [Freeswitch-users] One of my extensions Caller ID says Anonymous! In-Reply-To: References: <0c0801ce3089$839fe4f0$8adfaed0$@bizfocused.com> <0e6f01ce30aa$ebb5b210$c3211630$@bizfocused.com> Message-ID: On Thu, Apr 4, 2013 at 11:16 AM, Avi Marcus wrote: > Indeed, outbound_caller_id_name is just a variable. > > To use it you have to tell the dialplan to set effective_caller_id_name=${ > outbound_caller_id_name}. > effective_caller_id_name IS a variable that actually gets used. > (The default dialplan does this.) > I'm sorry, I should have explained it better. I do use the method in the default dialplan that sets: effective_caller_id_name=${outbound_caller_id_name} and effective_caller_id_number=${outbound_caller_id_number} for outgoing calls through the gateway (Flowroute). The annoying issue is that while the outbound_caller_id_number gets sent correctly, the outbound_caller_id_name gets lost somewhere between the gateway and my POTS line, which seems to use a CNAM lookup instead. I have been told that this is because many parts of the PSTN do not pass the "NAME" part of the Caller-ID. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130404/9a380b50/attachment.html From Tim.Meade at Millicorp.com Fri Apr 5 05:14:28 2013 From: Tim.Meade at Millicorp.com (Tim Meade) Date: Fri, 5 Apr 2013 01:14:28 +0000 Subject: [Freeswitch-users] Current master 1.5.1b? Message-ID: <804D48104511D4468F0D60DF9D31003511A9EB69@MAIL.millicorp.com> I just did a git clone and built that latest master. Here is what I'm getting from version: FreeSWITCH Version 1.5.1b+git~20130405T004141Z~2cef8580a0 (git 2cef858 2013-04-05 00:41:41Z) Is this a typo? 3 weeks ago it was 1.3.17 FreeSWITCH Version 1.3.17+git~20130321T141651Z~ac0defb874 (git ac0defb 2013-03-21 14:16:51Z) Also; this build is showing these alerts which we have not seen before. 2013-04-04 21:12:00.695788 [ALERT] sofia_glue.c:2430 Reuse Unused Cached DB handle db="sofia_reg_internal" [CORE_DB] 2013-04-04 21:12:00.695788 [ALERT] sofia_glue.c:2430 Reuse Unused Cached DB handle db="sofia_reg_internal" [CORE_DB] 2013-04-04 21:12:00.755821 [ALERT] switch_console.c:253 Create Cached DB handle (null) [CORE_DB] src/switch_console.c:253 2013-04-04 21:12:30.695446 [ALERT] sofia_glue.c:2430 Reuse Unused Cached DB handle db="sofia_reg_internal" [CORE_DB] 2013-04-04 21:12:30.695446 [ALERT] sofia_glue.c:2430 Reuse Unused Cached DB handle db="sofia_reg_internal" [CORE_DB] Otherwise things are working as advertised. Thanks Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/5761628a/attachment.html From anthony.minessale at gmail.com Fri Apr 5 06:08:09 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 4 Apr 2013 21:08:09 -0500 Subject: [Freeswitch-users] Current master 1.5.1b? In-Reply-To: <804D48104511D4468F0D60DF9D31003511A9EB69@MAIL.millicorp.com> References: <804D48104511D4468F0D60DF9D31003511A9EB69@MAIL.millicorp.com> Message-ID: Nope its 1.5 now. HEAD is odd revs feeding the preceding even. 1.3 fed 1.2 until it was discontinued for 1.5. Now 1.2 runs on its own and 1.5 has the latest dev brewing releases for 1.4. The message you are seeing is because you have the debug_level higher than 0. It was changed to go on alert level so it could be distinguished from other log messages. fsctl debug_level 0 to turn it off. On Thu, Apr 4, 2013 at 8:14 PM, Tim Meade wrote: > I just did a git clone and built that latest master. Here is what I?m > getting from version:**** > > ** ** > > FreeSWITCH Version 1.5.1b+git~20130405T004141Z~2cef8580a0 (git 2cef858 > 2013-04-05 00:41:41Z)**** > > ** ** > > Is this a typo? 3 weeks ago it was 1.3.17**** > > ** ** > > FreeSWITCH Version 1.3.17+git~20130321T141651Z~ac0defb874 (git ac0defb > 2013-03-21 14:16:51Z)**** > > ** ** > > ** ** > > Also; this build is showing these alerts which we have not seen before.** > ** > > ** ** > > 2013-04-04 21:12:00.695788 [ALERT] sofia_glue.c:2430 Reuse Unused Cached > DB handle db="sofia_reg_internal" [CORE_DB]**** > > 2013-04-04 21:12:00.695788 [ALERT] sofia_glue.c:2430 Reuse Unused Cached > DB handle db="sofia_reg_internal" [CORE_DB]**** > > 2013-04-04 21:12:00.755821 [ALERT] switch_console.c:253 Create Cached DB > handle (null) [CORE_DB] src/switch_console.c:253**** > > 2013-04-04 21:12:30.695446 [ALERT] sofia_glue.c:2430 Reuse Unused Cached > DB handle db="sofia_reg_internal" [CORE_DB]**** > > 2013-04-04 21:12:30.695446 [ALERT] sofia_glue.c:2430 Reuse Unused Cached > DB handle db="sofia_reg_internal" [CORE_DB]**** > > ** ** > > Otherwise things are working as advertised.**** > > ** ** > > Thanks**** > > ** ** > > Tim **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130404/8428cb1d/attachment-0001.html From henry.houfeng at gmail.com Fri Apr 5 06:16:24 2013 From: henry.houfeng at gmail.com (henry.houfeng) Date: Thu, 4 Apr 2013 19:16:24 -0700 (PDT) Subject: [Freeswitch-users] start_tone_detect, but no event is fired In-Reply-To: <1364911596726-7589332.post@n2.nabble.com> References: <1364911596726-7589332.post@n2.nabble.com> Message-ID: <1365128184720-7589431.post@n2.nabble.com> Anyone can help me? Thanks! Regards, Henry -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/start-tone-detect-but-no-event-is-fired-tp7589332p7589431.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ashish at nms.co.in Fri Apr 5 10:24:48 2013 From: ashish at nms.co.in (Ashish gautam) Date: Fri, 5 Apr 2013 11:54:48 +0530 Subject: [Freeswitch-users] Call not originated through event socket Message-ID: Hi Steven, Thanks for responding. I am checking the console at debug level and getting [NORMAL_CIRCUIT_CONGESTION] error. This must not happen as the number of calls I am generating is only 4. What is happening is, the ES generates calls to all the numbers except the last one (it always fails for the last number and shows NORMAL_CIRCUIT_CONGESTION). If the problem would be in originate dialstring, calls to other numbers would also be unsuccessful but they are working fine. Thanks. -Ashish On Fri, Apr 5, 2013 at 12:17 AM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: One of my extensions Caller ID says Anonymous! (Vik Killa) > 2. Re: freeswitch say application currency in multipal language > (Michael Collins) > 3. Re: One of my extensions Caller ID says Anonymous! (Avi Marcus) > 4. Re: Call not originated through event socket (Steven Ayre) > > > ---------- Forwarded message ---------- > From: Vik Killa > To: FreeSWITCH Users Help > Cc: > Date: Thu, 4 Apr 2013 14:13:26 -0400 > Subject: Re: [Freeswitch-users] One of my extensions Caller ID says > Anonymous! > Please let us know if and how you get it resolved. I'll do the same if I > can figure it out. Thanks. > > > > > ---------- Forwarded message ---------- > From: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Thu, 4 Apr 2013 11:14:52 -0700 > Subject: Re: [Freeswitch-users] freeswitch say application currency in > multipal language > I don't believe that there is currently a way to do this easily right now. > We just spoke about languages on yesterday's conference call and this is a > prime example of the kinds of things that we will need to overcome. > > Additionally I don't believe that I have any currencies other than > dollar.wav and dollars.wav for the English sounds. I'll be glad to get them > ordered. Could the community at large send me some ideas for units of > currency? Here are a few ideas: > > euro, euros > franc, francs > Canadian, Australian, US dollar/dollars > pound, pounds > > Send me some more ideas and I will get them added to the to-be-recorded > list. > > -MC > > On Thu, Apr 4, 2013 at 1:52 AM, bhavik patel wrote: > >> Hi all, >> I use free switch and i want to play sounds file like if user has credit >> in USD then doller.wav file play and EUR then another file will be play. >> >> Currently it play doller.wav by default in >> /usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play EUR >> so how can this possible. >> >> Is that any easy way to do this thing in multi language currency play in >> say application. >> >> i use this syntax in my free-switch dial plan >> $dialstring = "> $credit_balance\"/>"; >> >> Thanks In advance... >> >> -- >> Thanks, >> Bhavik Patel >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > ---------- Forwarded message ---------- > From: Avi Marcus > To: FreeSWITCH Users Help > Cc: > Date: Thu, 4 Apr 2013 21:16:04 +0300 > Subject: Re: [Freeswitch-users] One of my extensions Caller ID says > Anonymous! > Indeed, outbound_caller_id_name is just a variable. > > To use it you have to tell the dialplan to set effective_caller_id_name=${ > outbound_caller_id_name}. > effective_caller_id_name IS a variable that actually gets used. > (The default dialplan does this.) > > -Avi Marcus > BestFone > > > On Thu, Apr 4, 2013 at 8:26 PM, Steven Schoch < > schoch+freeswitch.org at xwin32.com> wrote: > >> In my directory, I set the variable "effective_caller_id_name" instead of >> "internal_caller_id_name". (I also set the "outbound_caller_id_name" >> variable just like your example, although that never seems to get sent to >> the PSTN.) >> >> -- >> Steve >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > ---------- Forwarded message ---------- > From: Steven Ayre > To: FreeSWITCH Users Help > Cc: > Date: Thu, 4 Apr 2013 19:46:22 +0100 > Subject: Re: [Freeswitch-users] Call not originated through event socket > +1 > > '/log 9' and watch the debug output. It'll be very clear about what's > happening. Most likely there's something wrong about your usage of > originate. > > -Steve > > > On 4 April 2013 19:08, Michael Collins wrote: > >> You could also just hop on fs_cli and watch the output while you make a >> test call with the event socket. Most likely you'll see what's going on. >> >> -MC >> >> On Thu, Apr 4, 2013 at 10:39 AM, Hector Geraldino < >> Hector.Geraldino at ipsoft.com> wrote: >> >>> +OK doesn?t mean the call was generated, it just means that FreeSWITCH >>> received (and processed) the command.**** >>> >>> ** ** >>> >>> You should probably be better served by listening to the events for each >>> one of these calls. There are a few handful events that can tell you if the >>> channel was created, the call was originated, if it was answered and when >>> it was dropped with the drop reason.**** >>> >>> ** ** >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish >>> gautam >>> *Sent:* Thursday, April 04, 2013 5:26 AM >>> *To:* freeswitch-users at lists.freeswitch.org >>> *Subject:* [Freeswitch-users] Call not originated through event socket** >>> ** >>> >>> ** ** >>> >>> Hi,**** >>> >>> ** ** >>> >>> I am generating calls through event socket using originate action to >>> multiple numbers to PSTN network. ES is returning '+OK' for the call not >>> generated. Still the call is not generated.**** >>> >>> ** ** >>> >>> Any help is appreciated.**** >>> >>> ** ** >>> >>> Thanks.**** >>> >>> ** ** >>> >>> -Ashish**** >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/753cd7df/attachment-0001.html From ashish at nms.co.in Fri Apr 5 10:32:34 2013 From: ashish at nms.co.in (Ashish gautam) Date: Fri, 5 Apr 2013 12:02:34 +0530 Subject: [Freeswitch-users] start_tone_detect, but no event is fired Message-ID: Hi Henry, You can switch the event reporting on using this command for the event you would like to listen to "event [xml/json etc.] LIST/ALL" On Fri, Apr 5, 2013 at 11:55 AM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: start_tone_detect, but no event is fired (henry.houfeng) > 2. Re: Call not originated through event socket (Ashish gautam) > > > ---------- Forwarded message ---------- > From: "henry.houfeng" > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Thu, 4 Apr 2013 19:16:24 -0700 (PDT) > Subject: Re: [Freeswitch-users] start_tone_detect, but no event is fired > Anyone can help me? Thanks! > > Regards, > Henry > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/start-tone-detect-but-no-event-is-fired-tp7589332p7589431.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > ---------- Forwarded message ---------- > From: Ashish gautam > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Fri, 5 Apr 2013 11:54:48 +0530 > Subject: Re: [Freeswitch-users] Call not originated through event socket > Hi Steven, > > Thanks for responding. > > I am checking the console at debug level and getting > [NORMAL_CIRCUIT_CONGESTION] error. This must not happen as the number of > calls I am generating is only 4. What is happening is, the ES generates > calls to all the numbers except the last one (it always fails for the last > number and shows NORMAL_CIRCUIT_CONGESTION). If the problem would be in > originate dialstring, calls to other numbers would also be unsuccessful but > they are working fine. > > Thanks. > -Ashish > > On Fri, Apr 5, 2013 at 12:17 AM, < > freeswitch-users-request at lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: One of my extensions Caller ID says Anonymous! (Vik Killa) >> 2. Re: freeswitch say application currency in multipal language >> (Michael Collins) >> 3. Re: One of my extensions Caller ID says Anonymous! (Avi Marcus) >> 4. Re: Call not originated through event socket (Steven Ayre) >> >> >> ---------- Forwarded message ---------- >> From: Vik Killa >> To: FreeSWITCH Users Help >> Cc: >> Date: Thu, 4 Apr 2013 14:13:26 -0400 >> Subject: Re: [Freeswitch-users] One of my extensions Caller ID says >> Anonymous! >> Please let us know if and how you get it resolved. I'll do the same if I >> can figure it out. Thanks. >> >> >> >> >> ---------- Forwarded message ---------- >> From: Michael Collins >> To: FreeSWITCH Users Help >> Cc: >> Date: Thu, 4 Apr 2013 11:14:52 -0700 >> Subject: Re: [Freeswitch-users] freeswitch say application currency in >> multipal language >> I don't believe that there is currently a way to do this easily right >> now. We just spoke about languages on yesterday's conference call and this >> is a prime example of the kinds of things that we will need to overcome. >> >> Additionally I don't believe that I have any currencies other than >> dollar.wav and dollars.wav for the English sounds. I'll be glad to get them >> ordered. Could the community at large send me some ideas for units of >> currency? Here are a few ideas: >> >> euro, euros >> franc, francs >> Canadian, Australian, US dollar/dollars >> pound, pounds >> >> Send me some more ideas and I will get them added to the to-be-recorded >> list. >> >> -MC >> >> On Thu, Apr 4, 2013 at 1:52 AM, bhavik patel wrote: >> >>> Hi all, >>> I use free switch and i want to play sounds file like if user has credit >>> in USD then doller.wav file play and EUR then another file will be play. >>> >>> Currently it play doller.wav by default in >>> /usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play EUR >>> so how can this possible. >>> >>> Is that any easy way to do this thing in multi language currency play in >>> say application. >>> >>> i use this syntax in my free-switch dial plan >>> $dialstring = ">> $credit_balance\"/>"; >>> >>> Thanks In advance... >>> >>> -- >>> Thanks, >>> Bhavik Patel >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> ---------- Forwarded message ---------- >> From: Avi Marcus >> To: FreeSWITCH Users Help >> Cc: >> Date: Thu, 4 Apr 2013 21:16:04 +0300 >> Subject: Re: [Freeswitch-users] One of my extensions Caller ID says >> Anonymous! >> Indeed, outbound_caller_id_name is just a variable. >> >> To use it you have to tell the dialplan to set >> effective_caller_id_name=${outbound_caller_id_name}. >> effective_caller_id_name IS a variable that actually gets used. >> (The default dialplan does this.) >> >> -Avi Marcus >> BestFone >> >> >> On Thu, Apr 4, 2013 at 8:26 PM, Steven Schoch < >> schoch+freeswitch.org at xwin32.com> wrote: >> >>> In my directory, I set the variable "effective_caller_id_name" instead >>> of "internal_caller_id_name". (I also set the "outbound_caller_id_name" >>> variable just like your example, although that never seems to get sent to >>> the PSTN.) >>> >>> -- >>> Steve >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> ---------- Forwarded message ---------- >> From: Steven Ayre >> To: FreeSWITCH Users Help >> Cc: >> Date: Thu, 4 Apr 2013 19:46:22 +0100 >> Subject: Re: [Freeswitch-users] Call not originated through event socket >> +1 >> >> '/log 9' and watch the debug output. It'll be very clear about what's >> happening. Most likely there's something wrong about your usage of >> originate. >> >> -Steve >> >> >> On 4 April 2013 19:08, Michael Collins wrote: >> >>> You could also just hop on fs_cli and watch the output while you make a >>> test call with the event socket. Most likely you'll see what's going on. >>> >>> -MC >>> >>> On Thu, Apr 4, 2013 at 10:39 AM, Hector Geraldino < >>> Hector.Geraldino at ipsoft.com> wrote: >>> >>>> +OK doesn?t mean the call was generated, it just means that >>>> FreeSWITCH received (and processed) the command.**** >>>> >>>> ** ** >>>> >>>> You should probably be better served by listening to the events for >>>> each one of these calls. There are a few handful events that can tell you >>>> if the channel was created, the call was originated, if it was answered and >>>> when it was dropped with the drop reason.**** >>>> >>>> ** ** >>>> >>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish >>>> gautam >>>> *Sent:* Thursday, April 04, 2013 5:26 AM >>>> *To:* freeswitch-users at lists.freeswitch.org >>>> *Subject:* [Freeswitch-users] Call not originated through event socket* >>>> *** >>>> >>>> ** ** >>>> >>>> Hi,**** >>>> >>>> ** ** >>>> >>>> I am generating calls through event socket using originate action to >>>> multiple numbers to PSTN network. ES is returning '+OK' for the call not >>>> generated. Still the call is not generated.**** >>>> >>>> ** ** >>>> >>>> Any help is appreciated.**** >>>> >>>> ** ** >>>> >>>> Thanks.**** >>>> >>>> ** ** >>>> >>>> -Ashish**** >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.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 >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/0a7b7fcf/attachment-0001.html From steveayre at gmail.com Fri Apr 5 11:03:40 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 5 Apr 2013 08:03:40 +0100 Subject: [Freeswitch-users] Call not originated through event socket In-Reply-To: References: Message-ID: NORMAL_CIRCUIT_CONGESTION is an error that can be returned by the SIP server you're calling out through. Show us the debug log and enable sip trace (sofia global siptrace on). Almost certainly that error is coming from your SIP provider and you'll need to take it up with them to see why they're returning that. Check for the SIP code in the final response they return (eg 503 Service Unavailable) and the Reason header in that that message (which is where you may see NORMAL_CIRCUIT_CONGESTION). -Steve On 5 April 2013 07:24, Ashish gautam wrote: > Hi Steven, > > Thanks for responding. > > I am checking the console at debug level and getting > [NORMAL_CIRCUIT_CONGESTION] error. This must not happen as the number of > calls I am generating is only 4. What is happening is, the ES generates > calls to all the numbers except the last one (it always fails for the last > number and shows NORMAL_CIRCUIT_CONGESTION). If the problem would be in > originate dialstring, calls to other numbers would also be unsuccessful but > they are working fine. > > Thanks. > -Ashish > > On Fri, Apr 5, 2013 at 12:17 AM, < > freeswitch-users-request at lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: One of my extensions Caller ID says Anonymous! (Vik Killa) >> 2. Re: freeswitch say application currency in multipal language >> (Michael Collins) >> 3. Re: One of my extensions Caller ID says Anonymous! (Avi Marcus) >> 4. Re: Call not originated through event socket (Steven Ayre) >> >> >> ---------- Forwarded message ---------- >> From: Vik Killa >> To: FreeSWITCH Users Help >> Cc: >> Date: Thu, 4 Apr 2013 14:13:26 -0400 >> Subject: Re: [Freeswitch-users] One of my extensions Caller ID says >> Anonymous! >> Please let us know if and how you get it resolved. I'll do the same if I >> can figure it out. Thanks. >> >> >> >> >> ---------- Forwarded message ---------- >> From: Michael Collins >> To: FreeSWITCH Users Help >> Cc: >> Date: Thu, 4 Apr 2013 11:14:52 -0700 >> Subject: Re: [Freeswitch-users] freeswitch say application currency in >> multipal language >> I don't believe that there is currently a way to do this easily right >> now. We just spoke about languages on yesterday's conference call and this >> is a prime example of the kinds of things that we will need to overcome. >> >> Additionally I don't believe that I have any currencies other than >> dollar.wav and dollars.wav for the English sounds. I'll be glad to get them >> ordered. Could the community at large send me some ideas for units of >> currency? Here are a few ideas: >> >> euro, euros >> franc, francs >> Canadian, Australian, US dollar/dollars >> pound, pounds >> >> Send me some more ideas and I will get them added to the to-be-recorded >> list. >> >> -MC >> >> On Thu, Apr 4, 2013 at 1:52 AM, bhavik patel wrote: >> >>> Hi all, >>> I use free switch and i want to play sounds file like if user has credit >>> in USD then doller.wav file play and EUR then another file will be play. >>> >>> Currently it play doller.wav by default in >>> /usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play EUR >>> so how can this possible. >>> >>> Is that any easy way to do this thing in multi language currency play in >>> say application. >>> >>> i use this syntax in my free-switch dial plan >>> $dialstring = ">> $credit_balance\"/>"; >>> >>> Thanks In advance... >>> >>> -- >>> Thanks, >>> Bhavik Patel >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> ---------- Forwarded message ---------- >> From: Avi Marcus >> To: FreeSWITCH Users Help >> Cc: >> Date: Thu, 4 Apr 2013 21:16:04 +0300 >> Subject: Re: [Freeswitch-users] One of my extensions Caller ID says >> Anonymous! >> Indeed, outbound_caller_id_name is just a variable. >> >> To use it you have to tell the dialplan to set >> effective_caller_id_name=${outbound_caller_id_name}. >> effective_caller_id_name IS a variable that actually gets used. >> (The default dialplan does this.) >> >> -Avi Marcus >> BestFone >> >> >> On Thu, Apr 4, 2013 at 8:26 PM, Steven Schoch < >> schoch+freeswitch.org at xwin32.com> wrote: >> >>> In my directory, I set the variable "effective_caller_id_name" instead >>> of "internal_caller_id_name". (I also set the "outbound_caller_id_name" >>> variable just like your example, although that never seems to get sent to >>> the PSTN.) >>> >>> -- >>> Steve >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> ---------- Forwarded message ---------- >> From: Steven Ayre >> To: FreeSWITCH Users Help >> Cc: >> Date: Thu, 4 Apr 2013 19:46:22 +0100 >> Subject: Re: [Freeswitch-users] Call not originated through event socket >> +1 >> >> '/log 9' and watch the debug output. It'll be very clear about what's >> happening. Most likely there's something wrong about your usage of >> originate. >> >> -Steve >> >> >> On 4 April 2013 19:08, Michael Collins wrote: >> >>> You could also just hop on fs_cli and watch the output while you make a >>> test call with the event socket. Most likely you'll see what's going on. >>> >>> -MC >>> >>> On Thu, Apr 4, 2013 at 10:39 AM, Hector Geraldino < >>> Hector.Geraldino at ipsoft.com> wrote: >>> >>>> +OK doesn?t mean the call was generated, it just means that >>>> FreeSWITCH received (and processed) the command.**** >>>> >>>> ** ** >>>> >>>> You should probably be better served by listening to the events for >>>> each one of these calls. There are a few handful events that can tell you >>>> if the channel was created, the call was originated, if it was answered and >>>> when it was dropped with the drop reason.**** >>>> >>>> ** ** >>>> >>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish >>>> gautam >>>> *Sent:* Thursday, April 04, 2013 5:26 AM >>>> *To:* freeswitch-users at lists.freeswitch.org >>>> *Subject:* [Freeswitch-users] Call not originated through event socket* >>>> *** >>>> >>>> ** ** >>>> >>>> Hi,**** >>>> >>>> ** ** >>>> >>>> I am generating calls through event socket using originate action to >>>> multiple numbers to PSTN network. ES is returning '+OK' for the call not >>>> generated. Still the call is not generated.**** >>>> >>>> ** ** >>>> >>>> Any help is appreciated.**** >>>> >>>> ** ** >>>> >>>> Thanks.**** >>>> >>>> ** ** >>>> >>>> -Ashish**** >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.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 >>> >>> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/e3032279/attachment-0001.html From khuenm at vega.com.vn Fri Apr 5 11:24:39 2013 From: khuenm at vega.com.vn (Khue Nguyen Minh) Date: Fri, 5 Apr 2013 14:24:39 +0700 Subject: [Freeswitch-users] CDR Message-ID: Hi all, I want create cdr (filetype: csv) file in freeswitch. But, I don't known how I can do it from xml dialplan. Please guide me. Thanks & Best regards, Khue. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/0ef2cbb7/attachment.html From ashish at nms.co.in Fri Apr 5 11:30:39 2013 From: ashish at nms.co.in (Ashish gautam) Date: Fri, 5 Apr 2013 13:00:39 +0530 Subject: [Freeswitch-users] Call not originated through event socket In-Reply-To: References: Message-ID: Hi Steve, I am generating outbound calls to the PSTN network connected to my FreeSWITCH box via PRI line. Below is a piece of debug log for the same. There is some more but I am pasting the error lines out of that: 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1242 Connect outbound channel FreeTDM/1:3/9891605603 ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5 2013-04-05 12:10:13.234533 [NOTICE] switch_channel.c:976 New Channel FreeTDM/1:3/9891605603 [ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5] ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1251 (FreeTDM/1:3/9891605603) State Change CS_NEW -> CS_INIT ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5 2013-04-05 12:10:13.234533 [DEBUG] switch_core_session.c:1310 Send signal FreeTDM/1:3/9891605603 [BREAK] 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1270 Attached session ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5 to channel 1:3 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1758 [s1c3][1:3] Changed state from DOWN to DIALING 2013-04-05 12:10:13.234533 [DEBUG] switch_ivr_originate.c:2039 Parsing global variables 2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [voiceMessageID]=[95] 2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [respreqd]=[1] 2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [mobnum]=[9467797613] 2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [lang]=[en] 2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true] 2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [continue_on_fail]=[true] 2013-04-05 12:10:13.234533 [INFO] ftmod_zt.c:671 Setting echo cancel to 64 taps for 1:4 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:403 Set codec PCMA 20ms 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1242 Connect outbound channel FreeTDM/1:4/9467797613 ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5 2013-04-05 12:10:13.234533 [NOTICE] switch_channel.c:976 New Channel FreeTDM/1:4/9467797613 [ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5] ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1251 (FreeTDM/1:4/9467797613) State Change CS_NEW -> CS_INIT ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5 2013-04-05 12:10:13.234533 [DEBUG] switch_core_session.c:1310 Send signal FreeTDM/1:4/9467797613 [BREAK] 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1270 Attached session ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5 to channel 1:4 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1758 [s1c4][1:4] Changed state from DOWN to DIALING 2013-04-05 12:10:13.254533 [DEBUG] switch_ivr_originate.c:2039 Parsing global variables 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [voiceMessageID]=[96] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [respreqd]=[1] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [mobnum]=[8802865008] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [lang]=[en] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [continue_on_fail]=[true] ab47f0c0-9dbb-11e2-b0c4-6fb3d8274ed5 2013-04-05 12:10:13.254533 [NOTICE] mod_freetdm.c:1766 Close Channel N/A [CS_NEW] ab47f0c0-9dbb-11e2-b0c4-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:568 () Running State Change CS_DESTROY ab47f0c0-9dbb-11e2-b0c4-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY ab47f0c0-9dbb-11e2-b0c4-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY going to sleep 2013-04-05 12:10:13.254533 [NOTICE] switch_ivr_originate.c:2636 Cannot create outgoing channel of type [freetdm] cause: [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 12:10:13.254533 [DEBUG] switch_ivr_originate.c:3601 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 12:10:13.254533 [DEBUG] switch_ivr_originate.c:2039 Parsing global variables 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [voiceMessageID]=[97] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [respreqd]=[1] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [mobnum]=[8826454579] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [lang]=[en] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [continue_on_fail]=[true] ab49c22e-9dbb-11e2-b0c8-6fb3d8274ed5 2013-04-05 12:10:13.254533 [NOTICE] mod_freetdm.c:1766 Close Channel N/A [CS_NEW] ab49c22e-9dbb-11e2-b0c8-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:568 () Running State Change CS_DESTROY ab49c22e-9dbb-11e2-b0c8-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY ab49c22e-9dbb-11e2-b0c8-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY going to sleep 2013-04-05 12:10:13.254533 [NOTICE] switch_ivr_originate.c:2636 Cannot create outgoing channel of type [freetdm] cause: [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 12:10:13.254533 [DEBUG] switch_ivr_originate.c:3601 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 12:10:13.274533 [DEBUG] switch_ivr_originate.c:2039 Parsing global variables 2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [voiceMessageID]=[98] 2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [respreqd]=[1] 2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [mobnum]=[9868599566] 2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [lang]=[en] 2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true] 2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [continue_on_fail]=[true] ab4b9752-9dbb-11e2-b0cc-6fb3d8274ed5 2013-04-05 12:10:13.274533 [NOTICE] mod_freetdm.c:1766 Close Channel N/A [CS_NEW] ab4b9752-9dbb-11e2-b0cc-6fb3d8274ed5 2013-04-05 12:10:13.274533 [DEBUG] switch_core_state_machine.c:568 () Running State Change CS_DESTROY ab4b9752-9dbb-11e2-b0cc-6fb3d8274ed5 2013-04-05 12:10:13.274533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY ab4b9752-9dbb-11e2-b0cc-6fb3d8274ed5 2013-04-05 12:10:13.274533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY going to sleep 2013-04-05 12:10:13.274533 [NOTICE] switch_ivr_originate.c:2636 Cannot create outgoing channel of type [freetdm] cause: [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 12:10:13.274533 [DEBUG] switch_ivr_originate.c:3601 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 12:10:13.274533 [DEBUG] ftdm_state.c:541 [s1c1][1:1] Executing state processor for DIALING 2013-04-05 12:10:13.274533 [DEBUG] ftmod_libpri.c:935 -- 1:1 STATE [DIALING] 2013-04-05 12:10:13.274533 [DEBUG] mod_freetdm.c:2657 got clear channel sig [DIALING] On Fri, Apr 5, 2013 at 12:33 PM, Steven Ayre wrote: > NORMAL_CIRCUIT_CONGESTION is an error that can be returned by the SIP > server you're calling out through. Show us the debug log and enable sip > trace (sofia global siptrace on). > > Almost certainly that error is coming from your SIP provider and you'll > need to take it up with them to see why they're returning that. > > Check for the SIP code in the final response they return (eg 503 Service > Unavailable) and the Reason header in that that message (which is where you > may see NORMAL_CIRCUIT_CONGESTION). > > -Steve > > > > On 5 April 2013 07:24, Ashish gautam wrote: > >> Hi Steven, >> >> Thanks for responding. >> >> I am checking the console at debug level and getting >> [NORMAL_CIRCUIT_CONGESTION] error. This must not happen as the number of >> calls I am generating is only 4. What is happening is, the ES generates >> calls to all the numbers except the last one (it always fails for the last >> number and shows NORMAL_CIRCUIT_CONGESTION). If the problem would be in >> originate dialstring, calls to other numbers would also be unsuccessful but >> they are working fine. >> >> Thanks. >> -Ashish >> >> On Fri, Apr 5, 2013 at 12:17 AM, < >> freeswitch-users-request at lists.freeswitch.org> wrote: >> >>> Send FreeSWITCH-users mailing list submissions to >>> freeswitch-users at lists.freeswitch.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> or, via email, send a message with subject or body 'help' to >>> freeswitch-users-request at lists.freeswitch.org >>> >>> You can reach the person managing the list at >>> freeswitch-users-owner at lists.freeswitch.org >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of FreeSWITCH-users digest..." >>> >>> Today's Topics: >>> >>> 1. Re: One of my extensions Caller ID says Anonymous! (Vik Killa) >>> 2. Re: freeswitch say application currency in multipal language >>> (Michael Collins) >>> 3. Re: One of my extensions Caller ID says Anonymous! (Avi Marcus) >>> 4. Re: Call not originated through event socket (Steven Ayre) >>> >>> >>> ---------- Forwarded message ---------- >>> From: Vik Killa >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Thu, 4 Apr 2013 14:13:26 -0400 >>> Subject: Re: [Freeswitch-users] One of my extensions Caller ID says >>> Anonymous! >>> Please let us know if and how you get it resolved. I'll do the same if I >>> can figure it out. Thanks. >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Michael Collins >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Thu, 4 Apr 2013 11:14:52 -0700 >>> Subject: Re: [Freeswitch-users] freeswitch say application currency in >>> multipal language >>> I don't believe that there is currently a way to do this easily right >>> now. We just spoke about languages on yesterday's conference call and this >>> is a prime example of the kinds of things that we will need to overcome. >>> >>> Additionally I don't believe that I have any currencies other than >>> dollar.wav and dollars.wav for the English sounds. I'll be glad to get them >>> ordered. Could the community at large send me some ideas for units of >>> currency? Here are a few ideas: >>> >>> euro, euros >>> franc, francs >>> Canadian, Australian, US dollar/dollars >>> pound, pounds >>> >>> Send me some more ideas and I will get them added to the to-be-recorded >>> list. >>> >>> -MC >>> >>> On Thu, Apr 4, 2013 at 1:52 AM, bhavik patel >> > wrote: >>> >>>> Hi all, >>>> I use free switch and i want to play sounds file like if user has >>>> credit in USD then doller.wav file play and EUR then another file will be >>>> play. >>>> >>>> Currently it play doller.wav by default in >>>> /usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play EUR >>>> so how can this possible. >>>> >>>> Is that any easy way to do this thing in multi language currency play >>>> in say application. >>>> >>>> i use this syntax in my free-switch dial plan >>>> $dialstring = ">>> PRONOUNCED $credit_balance\"/>"; >>>> >>>> Thanks In advance... >>>> >>>> -- >>>> Thanks, >>>> Bhavik Patel >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Avi Marcus >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Thu, 4 Apr 2013 21:16:04 +0300 >>> Subject: Re: [Freeswitch-users] One of my extensions Caller ID says >>> Anonymous! >>> Indeed, outbound_caller_id_name is just a variable. >>> >>> To use it you have to tell the dialplan to set >>> effective_caller_id_name=${outbound_caller_id_name}. >>> effective_caller_id_name IS a variable that actually gets used. >>> (The default dialplan does this.) >>> >>> -Avi Marcus >>> BestFone >>> >>> >>> On Thu, Apr 4, 2013 at 8:26 PM, Steven Schoch < >>> schoch+freeswitch.org at xwin32.com> wrote: >>> >>>> In my directory, I set the variable "effective_caller_id_name" instead >>>> of "internal_caller_id_name". (I also set the "outbound_caller_id_name" >>>> variable just like your example, although that never seems to get sent to >>>> the PSTN.) >>>> >>>> -- >>>> Steve >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Steven Ayre >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Thu, 4 Apr 2013 19:46:22 +0100 >>> Subject: Re: [Freeswitch-users] Call not originated through event socket >>> +1 >>> >>> '/log 9' and watch the debug output. It'll be very clear about what's >>> happening. Most likely there's something wrong about your usage of >>> originate. >>> >>> -Steve >>> >>> >>> On 4 April 2013 19:08, Michael Collins wrote: >>> >>>> You could also just hop on fs_cli and watch the output while you make a >>>> test call with the event socket. Most likely you'll see what's going on. >>>> >>>> -MC >>>> >>>> On Thu, Apr 4, 2013 at 10:39 AM, Hector Geraldino < >>>> Hector.Geraldino at ipsoft.com> wrote: >>>> >>>>> +OK doesn?t mean the call was generated, it just means that >>>>> FreeSWITCH received (and processed) the command.**** >>>>> >>>>> ** ** >>>>> >>>>> You should probably be better served by listening to the events for >>>>> each one of these calls. There are a few handful events that can tell you >>>>> if the channel was created, the call was originated, if it was answered and >>>>> when it was dropped with the drop reason.**** >>>>> >>>>> ** ** >>>>> >>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish >>>>> gautam >>>>> *Sent:* Thursday, April 04, 2013 5:26 AM >>>>> *To:* freeswitch-users at lists.freeswitch.org >>>>> *Subject:* [Freeswitch-users] Call not originated through event socket >>>>> **** >>>>> >>>>> ** ** >>>>> >>>>> Hi,**** >>>>> >>>>> ** ** >>>>> >>>>> I am generating calls through event socket using originate action to >>>>> multiple numbers to PSTN network. ES is returning '+OK' for the call not >>>>> generated. Still the call is not generated.**** >>>>> >>>>> ** ** >>>>> >>>>> Any help is appreciated.**** >>>>> >>>>> ** ** >>>>> >>>>> Thanks.**** >>>>> >>>>> ** ** >>>>> >>>>> -Ashish**** >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.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 >>>> >>>> >>> >>> _______________________________________________ >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/aa1ed3f1/attachment-0001.html From ashish at nms.co.in Fri Apr 5 11:38:47 2013 From: ashish at nms.co.in (Ashish gautam) Date: Fri, 5 Apr 2013 13:08:47 +0530 Subject: [Freeswitch-users] CDR In-Reply-To: References: Message-ID: Hello, I think here you will find the information that will solve your problem: http://wiki.freeswitch.org/wiki/Mod_cdr_csv#Default_configuration - Ashish On Fri, Apr 5, 2013 at 12:54 PM, Khue Nguyen Minh wrote: > Hi all, > > I want create cdr (filetype: csv) file in freeswitch. But, I don't known > how I can do it from xml dialplan. Please guide me. > > Thanks & Best regards, > Khue. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/71ebe474/attachment.html From khuenm at vega.com.vn Fri Apr 5 12:09:51 2013 From: khuenm at vega.com.vn (Khue Nguyen Minh) Date: Fri, 5 Apr 2013 15:09:51 +0700 Subject: [Freeswitch-users] CDR In-Reply-To: References: Message-ID: I see that in this link is guide about all parameters can export to cdr file. But, I don't see how I can create cdr file from dialplan xml. example, I have a javascript, from dialplan I can execute it: . If I write from fs_cli I receive error: Invalid Application cdr_csv 2013/4/5 Ashish gautam > Hello, > > I think here you will find the information that will solve your problem: > > http://wiki.freeswitch.org/wiki/Mod_cdr_csv#Default_configuration > > - Ashish > > On Fri, Apr 5, 2013 at 12:54 PM, Khue Nguyen Minh wrote: > >> Hi all, >> >> I want create cdr (filetype: csv) file in freeswitch. But, I don't known >> how I can do it from xml dialplan. Please guide me. >> >> Thanks & Best regards, >> Khue. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20130405/fca6f4ed/attachment.html From henry.houfeng at gmail.com Fri Apr 5 12:50:50 2013 From: henry.houfeng at gmail.com (henry.houfeng) Date: Fri, 5 Apr 2013 01:50:50 -0700 (PDT) Subject: [Freeswitch-users] start_tone_detect, but no event is fired In-Reply-To: References: <1364911596726-7589332.post@n2.nabble.com> Message-ID: <1365151850437-7589439.post@n2.nabble.com> Thanks for your reply. Yes, I already turned on ALL events list this: $con->events('plain','all'); And I am watching the log as well. I have set loglevel to 9 like this: sofia loglevel all 9 I captured IP packets with wireshark. The voice codec is G711U. And the RTP stream of early media can be replayed by wireshark. But no event is fired. Regards, Henry -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/start-tone-detect-but-no-event-is-fired-tp7589332p7589439.html Sent from the freeswitch-users mailing list archive at Nabble.com. From vitaliy.davudov at vts24.ru Fri Apr 5 13:06:55 2013 From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=) Date: Fri, 05 Apr 2013 13:06:55 +0400 Subject: [Freeswitch-users] make current error In-Reply-To: References: <515D7601.8030708@vts24.ru> Message-ID: <515E942F.60004@vts24.ru> Thanks a lot! This helped! 04.04.2013 16:56, Steven Ayre ?????: > Try checking out a fresh copy with 'git clone' > > > On 4 April 2013 13:45, ??????? ??????? > wrote: > > Hi, list! > > I've installed FreeSWITCH Version > 1.2.3+git~20120920T220849Z~f718a5e8e6 > (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z) > > Now, I try to update to the latest tree. While running "make > current" in > FS source directory, there is an error: > > ... > make[1]: Leaving directory `/usr/src/freeswitch' > make all > make[1]: Entering directory `/usr/src/freeswitch' > CDPATH="${ZSH_VERSION+.}:" && cd . && /bin/sh > /usr/src/freeswitch/build/config/missing --run aclocal-1.11 > cd . && /bin/sh /usr/src/freeswitch/build/config/missing --run > automake-1.11 --foreign > CDPATH="${ZSH_VERSION+.}:" && cd . && /bin/sh > /usr/src/freeswitch/build/config/missing --run autoconf > configure.in:141 : error: possibly > undefined macro: AC_PROG_LIBTOOL > If this token and others are legitimate, please use > m4_pattern_allow. > See the Autoconf documentation. > make[1]: *** [configure] Error 1 > make[1]: Leaving directory `/usr/src/freeswitch' > make: *** [current] Error 2 > > > In addition, already installed: > > # rpm -qa | grep autoconf > autoconf-2.62-12.3 > # rpm -qa | grep libtool > libtool-1.5.22-7.el5_4 > > What might be the reasons of this error? May be new versions of > libtool/autoconf is required? > > -- > Best regards, > Vitaly. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 -- Best regards, Vitaly. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/391df08c/attachment.html From steveayre at gmail.com Fri Apr 5 13:22:04 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 5 Apr 2013 10:22:04 +0100 Subject: [Freeswitch-users] CDR In-Reply-To: References: Message-ID: The simple answer is from from dialplan you don't. Not all calls might hit dialplan, and if the caller hangs up it might not reach the end of the dialplan. CDRs generated from the dialplan would not be reliable. There is a specific reporting call stage that exists specifically for CDRs. mod_cdr_csv (as with the other CDR modules) install a hook into the reporting stage and automatically record a CDR if loaded. You don't need to call anything from the dialplan. If you want to set information in the CDR from your dialplan, set a variable on the channel (set application) and then add that variable to the list of fields logged by mod_cdr_csv. -Steve On 5 April 2013 09:09, Khue Nguyen Minh wrote: > I see that in this link is guide about all parameters can export to cdr > file. But, I don't see how I can create cdr file from dialplan xml. > example, I have a javascript, from dialplan I can execute it: application="javascript" data="hello.js"/>. > If I write > from fs_cli I receive error: Invalid Application cdr_csv > > > 2013/4/5 Ashish gautam > >> Hello, >> >> I think here you will find the information that will solve your problem: >> >> http://wiki.freeswitch.org/wiki/Mod_cdr_csv#Default_configuration >> >> - Ashish >> >> On Fri, Apr 5, 2013 at 12:54 PM, Khue Nguyen Minh wrote: >> >>> Hi all, >>> >>> I want create cdr (filetype: csv) file in freeswitch. But, I don't known >>> how I can do it from xml dialplan. Please guide me. >>> >>> Thanks & Best regards, >>> Khue. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20130405/e2cd4b3d/attachment-0001.html From alex at digitalmail.com Fri Apr 5 17:21:20 2013 From: alex at digitalmail.com (Alex Lake) Date: Fri, 05 Apr 2013 14:21:20 +0100 Subject: [Freeswitch-users] Creating a User account with no SIP login Message-ID: <515ECFD0.9020706@digitalmail.com> I want to have a freeswitch user that can have voicemail but can't be logged into from a SIP handset. How can I do this? From steveu at coppice.org Fri Apr 5 17:44:25 2013 From: steveu at coppice.org (Steve Underwood) Date: Fri, 05 Apr 2013 21:44:25 +0800 Subject: [Freeswitch-users] Spandsp modems as non-root? In-Reply-To: References: <50703D76.7010301@integrafin.co.uk> <515C30EF.8020301@integrafin.co.uk> <515D9170.9090208@coppice.org> Message-ID: <515ED539.8030105@coppice.org> On 04/05/2013 03:19 AM, Shahar wrote: > were not thinking about utilizing spandsp for the dialup - we just > want it to create the com devices. If it creates the com devices, what will be your modem? > > then were passing the handling via mgetty to pppd - just like hylafax > takes over the spandsp in the data handling. > maybe we need to disable the T.38 for this to work (and stay in G711) ? Are you under the impression there are duplex data modems in spandsp? If so, its a false impression. There is a V.22bis modem, but it is not exposed in the AT interface provided for the FAX modems. Steve From avi at avimarcus.net Fri Apr 5 18:07:21 2013 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 5 Apr 2013 17:07:21 +0300 Subject: [Freeswitch-users] Creating a User account with no SIP login In-Reply-To: <515ECFD0.9020706@digitalmail.com> References: <515ECFD0.9020706@digitalmail.com> Message-ID: Set a very long and complicated password, and don't tell it to them. -Avi On Fri, Apr 5, 2013 at 4:21 PM, Alex Lake wrote: > I want to have a freeswitch user that can have voicemail but can't be > logged into from a SIP handset. How can I do this? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/15177ffc/attachment.html From jason.holden at start.ca Fri Apr 5 18:17:52 2013 From: jason.holden at start.ca (Jason Holden) Date: Fri, 5 Apr 2013 10:17:52 -0400 Subject: [Freeswitch-users] error 606 user not registered Message-ID: <6FD7A17C91074F82B3563A36E8261D5E@bob> Anyone have a suggestion on how to correct this? When I do a show registrations I do see the extension registered. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/44e86f48/attachment.html From krice at freeswitch.org Fri Apr 5 18:17:22 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 05 Apr 2013 09:17:22 -0500 Subject: [Freeswitch-users] CDR In-Reply-To: Message-ID: You don?t do that in the dialplan, mod_cdr_csv does that for you On 4/5/13 2:24 AM, "Khue Nguyen Minh" wrote: > Hi all, > > I want create cdr (filetype: csv) file in freeswitch. But, I don't known how I > can do it from xml dialplan. Please guide me. > > Thanks & Best regards, > Khue. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/d1558dc2/attachment.html From jnvines at gmail.com Fri Apr 5 18:23:29 2013 From: jnvines at gmail.com (Nick Vines) Date: Fri, 5 Apr 2013 07:23:29 -0700 Subject: [Freeswitch-users] error 606 user not registered In-Reply-To: <6FD7A17C91074F82B3563A36E8261D5E@bob> References: <6FD7A17C91074F82B3563A36E8261D5E@bob> Message-ID: Can you post the dialplan and log? Check the domain of the user and context of the dialplan too (606 at domain1.com is different than 606 at domain2.com). On Fri, Apr 5, 2013 at 7:17 AM, Jason Holden wrote: > Anyone have a suggestion on how to correct this?**** > > When I do a show registrations I do see the extension registered.**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/5e77caa8/attachment.html From nneul at mst.edu Fri Apr 5 18:37:39 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Fri, 05 Apr 2013 09:37:39 -0500 Subject: [Freeswitch-users] Anyone have a good dialplan recipe for a long distance forced authorization code? Message-ID: <515EE1B3.2050702@mst.edu> I.e. to require a pin entry prior to bridging an outbound long distance call? I was thinking I could duplicate my outbound gateway bridging rules with an answer, start_dtmf, play_and_get_digits, followed by the actual bridging instruction. Is that the ideal way to do this? Intent is for open access phones that have restricted long distance unless the line-specific authorization code is entered. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From vipkilla at gmail.com Fri Apr 5 19:15:56 2013 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 5 Apr 2013 11:15:56 -0400 Subject: [Freeswitch-users] One of my extensions Caller ID says Anonymous! In-Reply-To: References: <0c0801ce3089$839fe4f0$8adfaed0$@bizfocused.com> <0e6f01ce30aa$ebb5b210$c3211630$@bizfocused.com> Message-ID: For those of you wondering, I figured this out on the Cisco SPAXXX First set the SIP parameter like this: RPID-FROM Then for each extension set this: Yes Now the Cisco sends the extension name instead of 'Anonymous' -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/9f2d3cbc/attachment.html From steveayre at gmail.com Fri Apr 5 19:23:55 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 5 Apr 2013 16:23:55 +0100 Subject: [Freeswitch-users] Anyone have a good dialplan recipe for a long distance forced authorization code? In-Reply-To: <515EE1B3.2050702@mst.edu> References: <515EE1B3.2050702@mst.edu> Message-ID: <4231C2C3-6D1A-4DB6-9A7F-93C759CECDE7@gmail.com> Don't use start_dtmf unless you are receiving inband dtmf (audible tones in the audio without rfc2833 or sip info dtmf). Answer and play_and_get_digits probably is the way to do this yes... Whether in dialplan or from lua etc. Rather than duplicate anything, how about having a context that checks the PIN and transfers the call into the existing context for routing? That also means they couldn't get to that context without the correct PIN, so would be an extra layer of security. You can then either send calls for an entire SIP profile into the pin check context (context profile param) or for specific users (user_context user param). Steve On 5 Apr 2013, at 15:37, Nathan Neulinger wrote: > I.e. to require a pin entry prior to bridging an outbound long distance call? > > I was thinking I could duplicate my outbound gateway bridging rules with an answer, start_dtmf, play_and_get_digits, > followed by the actual bridging instruction. > > Is that the ideal way to do this? Intent is for open access phones that have restricted long distance unless the > line-specific authorization code is entered. > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Apr 5 19:26:48 2013 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 5 Apr 2013 18:26:48 +0300 Subject: [Freeswitch-users] Anyone have a good dialplan recipe for a long distance forced authorization code? In-Reply-To: <515EE1B3.2050702@mst.edu> References: <515EE1B3.2050702@mst.edu> Message-ID: Sounds *sorta* like DISA, but for users already authorized just not authorized for long distance. For examples, look up DISA on the wiki or perhaps fusionpbx's lua scripts and take a look... -Avi Marcus BestFone -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/ee07dc3d/attachment.html From steveayre at gmail.com Fri Apr 5 19:27:35 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 5 Apr 2013 16:27:35 +0100 Subject: [Freeswitch-users] Anyone have a good dialplan recipe for a long distance forced authorization code? In-Reply-To: <515EE1B3.2050702@mst.edu> References: <515EE1B3.2050702@mst.edu> Message-ID: <085E3C1B-A82C-400A-8CE2-B69826E4CB7C@gmail.com> Ah sorry, this is for specific outbound calls rather than incoming... I might approach this by setting a variable if the call is allowed... For local destinations always set it, and for long distance only if the PIN is correct. Then allow the call to continue processing further into the dialplan, and add an extra condition to the bridging extension that checks the variable is set. Bear in mind that unless you're using TLS+SRTP/ZRTP it may be possible to eavesdrop on the PIN. Steve On 5 Apr 2013, at 15:37, Nathan Neulinger wrote: > I.e. to require a pin entry prior to bridging an outbound long distance call? > > I was thinking I could duplicate my outbound gateway bridging rules with an answer, start_dtmf, play_and_get_digits, > followed by the actual bridging instruction. > > Is that the ideal way to do this? Intent is for open access phones that have restricted long distance unless the > line-specific authorization code is entered. > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveayre at gmail.com Fri Apr 5 19:31:57 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 5 Apr 2013 16:31:57 +0100 Subject: [Freeswitch-users] start_tone_detect, but no event is fired In-Reply-To: <1365151850437-7589439.post@n2.nabble.com> References: <1364911596726-7589332.post@n2.nabble.com> <1365151850437-7589439.post@n2.nabble.com> Message-ID: How are you invoking tone_detect? Perhaps the frequencies you're after are t present in the call, or its too noisy Steve On 5 Apr 2013, at 09:50, "henry.houfeng" wrote: > Thanks for your reply. > > Yes, I already turned on ALL events list this: > $con->events('plain','all'); > > And I am watching the log as well. I have set loglevel to 9 like this: > sofia loglevel all 9 > > I captured IP packets with wireshark. The voice codec is G711U. And the RTP > stream of early media can be replayed by wireshark. But no event is fired. > > Regards, > Henry > > > > > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/start-tone-detect-but-no-event-is-fired-tp7589332p7589439.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 nneul at mst.edu Fri Apr 5 19:35:40 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Fri, 05 Apr 2013 10:35:40 -0500 Subject: [Freeswitch-users] Anyone have a good dialplan recipe for a long distance forced authorization code? In-Reply-To: <085E3C1B-A82C-400A-8CE2-B69826E4CB7C@gmail.com> References: <515EE1B3.2050702@mst.edu> <085E3C1B-A82C-400A-8CE2-B69826E4CB7C@gmail.com> Message-ID: <515EEF4C.8030102@mst.edu> Ah, so effectively, similar to duplicating the rulesets for outbound long distance, but fall through after setting the variable instead of fully duplicating. That may save some additional trouble later on as well. -- Nathan On 04/05/2013 10:27 AM, Steven Ayre wrote: > Ah sorry, this is for specific outbound calls rather than incoming... > > I might approach this by setting a variable if the call is allowed... For local destinations always set it, and for long distance only if the PIN is correct. > > Then allow the call to continue processing further into the dialplan, and add an extra condition to the bridging extension that checks the variable is set. > > Bear in mind that unless you're using TLS+SRTP/ZRTP it may be possible to eavesdrop on the PIN. > > Steve > > > > On 5 Apr 2013, at 15:37, Nathan Neulinger wrote: > >> I.e. to require a pin entry prior to bridging an outbound long distance call? >> >> I was thinking I could duplicate my outbound gateway bridging rules with an answer, start_dtmf, play_and_get_digits, >> followed by the actual bridging instruction. >> >> Is that the ideal way to do this? Intent is for open access phones that have restricted long distance unless the >> line-specific authorization code is entered. >> >> -- Nathan >> >> ------------------------------------------------------------ >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From steveayre at gmail.com Fri Apr 5 19:37:01 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 5 Apr 2013 16:37:01 +0100 Subject: [Freeswitch-users] record_session wav files are corrupted In-Reply-To: References: Message-ID: Do you have mod_sndfile loaded? Steve On 4 Apr 2013, at 16:06, fs at tcowan.net wrote: > On 04-02-2013 05:17, Steven Schoch wrote: >> On Tue, Apr 2, 2013 at 10:28 AM, wrote: >> >>> If I try to import the >>> audio as raw format in audacity you can hear some high pitched >>> talking >>> that plays back very quickly. >> >> When you import as raw data, try changing the sample rate to 8000. >> >> The reason Audacity can't open the file is because the header, which >> specifies the encoding, channels, sample rate, etc. is damaged or >> missing. >> >> -- >> Steve >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > Well why would the header to damaged or missing. I have updated the > config to the following and it still doesnt work. If I look at the > header info with windows by going to file properties it doesnt list > anything in the header info. The file is just a bunch of high pitched > noise if I dont import as raw. I can hear the audio fine when importing > raw and 8000hz rate. I am not sure what else I need to do to make this > work. > > I followed the steps listed on: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session > > > > data="sip_h_X-accountcode=${accountcode}" /> > > > > data="effective_caller_id_name=${outbound_caller_id_name}" /> > data="effective_caller_id_number=${outbound_caller_id_number}" /> > > > /> > > > > > > > data="$${base_dir}/recordings/${domain_name}/outbound10d-${strftime(%m-%d-%Y-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav" > /> > data="sofia/gateway/customer1-bandwidth/$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 From zhnupy at gmail.com Fri Apr 5 21:50:25 2013 From: zhnupy at gmail.com (Zhnupy Gonzalez) Date: Fri, 5 Apr 2013 11:50:25 -0600 Subject: [Freeswitch-users] pstn call control from computer Message-ID: Hi I'm a total newbie looking a way to controll PSTN calls from my computer, by control I mean: see caller id on screen, answering with softh phone, pressing a button to play a song, that kind of stuff. Is it possible with hardware like grandstream Ht503 or obihai OBi110 or do I need more (and more expensive) hardware? regards zhnu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/8b9a7e2d/attachment.html From dantavious313 at gmail.com Fri Apr 5 22:34:55 2013 From: dantavious313 at gmail.com (Derrick Dantavious Edwards) Date: Fri, 05 Apr 2013 14:34:55 -0400 Subject: [Freeswitch-users] mod_python question Message-ID: <1628059.cgEIOWz82P@zeus> Hi, It seems that mod_python defaults to Python version 2.7. I would like for it to use verison 3.3. I have version 3.3 installed on system and attempted to edit the mod_python Makefile to point to installed Python verison 3.3 but it failed on the compile. Any ideas ? V/r Derrick From mike at jerris.com Fri Apr 5 23:35:35 2013 From: mike at jerris.com (Michael Jerris) Date: Fri, 5 Apr 2013 15:35:35 -0400 Subject: [Freeswitch-users] B64 audio codec? In-Reply-To: <1364998586.10160.194.camel@luna.madrid.commsmundi.com> References: <1364917233.10160.90.camel@luna.madrid.commsmundi.com> <003801ce2fc3$0f3a99f0$2dafcdd0$@v10networks.ca> <7847442903959552389@unknownmsgid> <1364998586.10160.194.camel@luna.madrid.commsmundi.com> Message-ID: Coppice- Do you think we should have added packet redundancy and FEC too? On Apr 3, 2013, at 10:16 AM, Fran?ois wrote: > Hehe, wasn't quite sure as this was commited the 29th.. Still, I'll give > it a try, base64 audio sounds like lots of fun! > > > On Tue, 2013-04-02 at 12:23 -0600, Michael Jerris wrote: >> Good jokes are functional. >> >> On Apr 2, 2013, at 12:00 PM, Jeff Leung wrote: >> >>> Personally I still have a hunch that mod_b64 is a subtle April Fool's Day joke. Even though it's a joke, it's still a very functional codec. ;) >>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- >>>> users-bounces at lists.freeswitch.org] On Behalf Of Fran?ois >>>> Sent: Tuesday, April 2, 2013 8:41 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: [Freeswitch-users] B64 audio codec? >>>> >>>> Hi all, >>>> >>>> Anyone has some info about the recently commited mod_b64 codec? >>>> >>>> The source refers to it as "The B64 ultra-low delay audio codec", and links to a >>>> webpage www.b64-codec.org which does not exist! >>>> >>>> Anyway, it sounds interesting and doesn't seem to be an april fools joke! >>>> >>>> Thanks, >>>> Fran?ois. >>>> From anthony.minessale at gmail.com Sat Apr 6 00:52:15 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 5 Apr 2013 15:52:15 -0500 Subject: [Freeswitch-users] B64 audio codec? In-Reply-To: References: <1364917233.10160.90.camel@luna.madrid.commsmundi.com> <003801ce2fc3$0f3a99f0$2dafcdd0$@v10networks.ca> <7847442903959552389@unknownmsgid> <1364998586.10160.194.camel@luna.madrid.commsmundi.com> Message-ID: The proposed method was sending the last 5 packets in their entirely also b64 encoded so you could easily recover dropped packets. That would probably put us over the MTU so we may have to consider gzipping the data too. On Fri, Apr 5, 2013 at 2:35 PM, Michael Jerris wrote: > Coppice- Do you think we should have added packet redundancy and FEC too? > > On Apr 3, 2013, at 10:16 AM, Fran?ois > wrote: > > > Hehe, wasn't quite sure as this was commited the 29th.. Still, I'll give > > it a try, base64 audio sounds like lots of fun! > > > > > > On Tue, 2013-04-02 at 12:23 -0600, Michael Jerris wrote: > >> Good jokes are functional. > >> > >> On Apr 2, 2013, at 12:00 PM, Jeff Leung wrote: > >> > >>> Personally I still have a hunch that mod_b64 is a subtle April Fool's > Day joke. Even though it's a joke, it's still a very functional codec. ;) > >>> > >>>> -----Original Message----- > >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch- > >>>> users-bounces at lists.freeswitch.org] On Behalf Of Fran?ois > >>>> Sent: Tuesday, April 2, 2013 8:41 AM > >>>> To: freeswitch-users at lists.freeswitch.org > >>>> Subject: [Freeswitch-users] B64 audio codec? > >>>> > >>>> Hi all, > >>>> > >>>> Anyone has some info about the recently commited mod_b64 codec? > >>>> > >>>> The source refers to it as "The B64 ultra-low delay audio codec", and > links to a > >>>> webpage www.b64-codec.org which does not exist! > >>>> > >>>> Anyway, it sounds interesting and doesn't seem to be an april fools > joke! > >>>> > >>>> Thanks, > >>>> Fran?ois. > >>>> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130405/416a343b/attachment.html From steveayre at gmail.com Sat Apr 6 01:55:46 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 5 Apr 2013 22:55:46 +0100 Subject: [Freeswitch-users] B64 audio codec? In-Reply-To: References: <1364917233.10160.90.camel@luna.madrid.commsmundi.com> <003801ce2fc3$0f3a99f0$2dafcdd0$@v10networks.ca> <7847442903959552389@unknownmsgid> <1364998586.10160.194.camel@luna.madrid.commsmundi.com> Message-ID: We could also add optional encryption support, for platforms that don't support SRTP/ZRTP. I suggest adopting ROT-13 for light encryption and ROT-26 for heavier encryption (twice as good!) -Steve On 5 April 2013 21:52, Anthony Minessale wrote: > The proposed method was sending the last 5 packets in their entirely also > b64 encoded so you could easily recover dropped packets. That would > probably put us over the MTU so we may have to consider gzipping the data > too. > > > > On Fri, Apr 5, 2013 at 2:35 PM, Michael Jerris wrote: > >> Coppice- Do you think we should have added packet redundancy and FEC too? >> >> On Apr 3, 2013, at 10:16 AM, Fran?ois >> wrote: >> >> > Hehe, wasn't quite sure as this was commited the 29th.. Still, I'll give >> > it a try, base64 audio sounds like lots of fun! >> > >> > >> > On Tue, 2013-04-02 at 12:23 -0600, Michael Jerris wrote: >> >> Good jokes are functional. >> >> >> >> On Apr 2, 2013, at 12:00 PM, Jeff Leung wrote: >> >> >> >>> Personally I still have a hunch that mod_b64 is a subtle April Fool's >> Day joke. Even though it's a joke, it's still a very functional codec. ;) >> >>> >> >>>> -----Original Message----- >> >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch- >> >>>> users-bounces at lists.freeswitch.org] On Behalf Of Fran?ois >> >>>> Sent: Tuesday, April 2, 2013 8:41 AM >> >>>> To: freeswitch-users at lists.freeswitch.org >> >>>> Subject: [Freeswitch-users] B64 audio codec? >> >>>> >> >>>> Hi all, >> >>>> >> >>>> Anyone has some info about the recently commited mod_b64 codec? >> >>>> >> >>>> The source refers to it as "The B64 ultra-low delay audio codec", >> and links to a >> >>>> webpage www.b64-codec.org which does not exist! >> >>>> >> >>>> Anyway, it sounds interesting and doesn't seem to be an april fools >> joke! >> >>>> >> >>>> Thanks, >> >>>> Fran?ois. >> >>>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at 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/20130405/351e4e47/attachment-0001.html From steveayre at gmail.com Sat Apr 6 01:57:08 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 5 Apr 2013 22:57:08 +0100 Subject: [Freeswitch-users] CRIT switch_ivr_bridge in log file In-Reply-To: References: <0D1C698866F66045A6201FD0F59CAC900147722B8F@EX.frontier.local> Message-ID: Well, harmless as long as your log files aren't too large anyway. On 4 April 2013 00:43, Anthony Minessale wrote: > That was an extra debug line, its been removed in latest code in the repo. > Its harmless to you and it will be gone when you update. > > > > > On Wed, Apr 3, 2013 at 6:09 PM, Colin Mason wrote: > >> Hello,**** >> >> ** ** >> >> I?m using FreeSWITCH version:**** >> >> FreeSWITCH Version 1.2.8+git~20130402T040229Z~b72d2c32d7 (git b72d2c3 >> 2013-04-02 04:02:29Z)**** >> >> ** ** >> >> An inbound call is sent to my box and after the far end (caller) ACKs my >> 200 OK, FreeSWITCH prints this to the log and then sends off a SIP INFO >> packet to the callee with caller id information. In the log CALLERNAME and >> CALLERNUM are replaced by the actual caller details:**** >> >> ** ** >> >> ** ** >> >> 2013-04-03 19:01:30.899243 [CRIT] switch_ivr_bridge.c:133 SEND >> sofia/carrier/XXXXXXXXXX [CALLERNAME][CALLERNUM]**** >> >> 2013-04-03 19:01:30.899243 [CRIT] switch_ivr_bridge.c:133 SEND >> sofia/internet/XXXXXXXXXX at X.X.X.X:5060 [CALLERNAME][CALLERNUM]**** >> >> ** ** >> >> ** ** >> >> It?s a critical error and it is showing up for every inbound call. >> Everything appears to be operating normally. Is this error just printing >> information to the log or should I be concerned about it?**** >> >> ** ** >> >> Thanks,**** >> >> Colin**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at 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/20130405/75197b0f/attachment.html From jleung at v10networks.ca Sat Apr 6 02:03:01 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Fri, 5 Apr 2013 15:03:01 -0700 Subject: [Freeswitch-users] B64 audio codec? In-Reply-To: References: <1364917233.10160.90.camel@luna.madrid.commsmundi.com> <003801ce2fc3$0f3a99f0$2dafcdd0$@v10networks.ca> <7847442903959552389@unknownmsgid> <1364998586.10160.194.camel@luna.madrid.commsmundi.com> Message-ID: <000c01ce3249$5805a750$0810f5f0$@v10networks.ca> Problem is with ROT13 or ROT26, anybody can read it given they have the appropriate decoder ;) From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Friday, April 5, 2013 2:56 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] B64 audio codec? We could also add optional encryption support, for platforms that don't support SRTP/ZRTP. I suggest adopting ROT-13 for light encryption and ROT-26 for heavier encryption (twice as good!) -Steve On 5 April 2013 21:52, Anthony Minessale wrote: The proposed method was sending the last 5 packets in their entirely also b64 encoded so you could easily recover dropped packets. That would probably put us over the MTU so we may have to consider gzipping the data too. On Fri, Apr 5, 2013 at 2:35 PM, Michael Jerris wrote: Coppice- Do you think we should have added packet redundancy and FEC too? On Apr 3, 2013, at 10:16 AM, Fran?ois wrote: > Hehe, wasn't quite sure as this was commited the 29th.. Still, I'll give > it a try, base64 audio sounds like lots of fun! > > > On Tue, 2013-04-02 at 12:23 -0600, Michael Jerris wrote: >> Good jokes are functional. >> >> On Apr 2, 2013, at 12:00 PM, Jeff Leung wrote: >> >>> Personally I still have a hunch that mod_b64 is a subtle April Fool's Day joke. Even though it's a joke, it's still a very functional codec. ;) >>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- >>>> users-bounces at lists.freeswitch.org] On Behalf Of Fran?ois >>>> Sent: Tuesday, April 2, 2013 8:41 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: [Freeswitch-users] B64 audio codec? >>>> >>>> Hi all, >>>> >>>> Anyone has some info about the recently commited mod_b64 codec? >>>> >>>> The source refers to it as "The B64 ultra-low delay audio codec", and links to a >>>> webpage www.b64-codec.org which does not exist! >>>> >>>> Anyway, it sounds interesting and doesn't seem to be an april fools joke! >>>> >>>> Thanks, >>>> Fran?ois. >>>> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130405/123755e6/attachment.html From krice at freeswitch.org Sat Apr 6 02:40:09 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 5 Apr 2013 17:40:09 -0500 Subject: [Freeswitch-users] B64 audio codec? In-Reply-To: <000c01ce3249$5805a750$0810f5f0$@v10networks.ca> References: <1364917233.10160.90.camel@luna.madrid.commsmundi.com> <003801ce2fc3$0f3a99f0$2dafcdd0$@v10networks.ca> <7847442903959552389@unknownmsgid> <1364998586.10160.194.camel@luna.madrid.commsmundi.com> <000c01ce3249$5805a750$0810f5f0$@v10networks.ca> Message-ID: <21E2C4C9-4F6E-491A-B489-0DF703D7E182@freeswitch.org> it already includes ROT13 encryption Ken Sent from my iPad On Apr 5, 2013, at 17:03, "Jeff Leung" wrote: > Problem is with ROT13 or ROT26, anybody can read it given they have the appropriate decoder ;) > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre > Sent: Friday, April 5, 2013 2:56 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] B64 audio codec? > > We could also add optional encryption support, for platforms that don't support SRTP/ZRTP. I suggest adopting ROT-13 for light encryption and ROT-26 for heavier encryption (twice as good!) > > -Steve > > > On 5 April 2013 21:52, Anthony Minessale wrote: > The proposed method was sending the last 5 packets in their entirely also b64 encoded so you could easily recover dropped packets. That would probably put us over the MTU so we may have to consider gzipping the data too. > > > > On Fri, Apr 5, 2013 at 2:35 PM, Michael Jerris wrote: > Coppice- Do you think we should have added packet redundancy and FEC too? > > On Apr 3, 2013, at 10:16 AM, Fran?ois wrote: > > > Hehe, wasn't quite sure as this was commited the 29th.. Still, I'll give > > it a try, base64 audio sounds like lots of fun! > > > > > > On Tue, 2013-04-02 at 12:23 -0600, Michael Jerris wrote: > >> Good jokes are functional. > >> > >> On Apr 2, 2013, at 12:00 PM, Jeff Leung wrote: > >> > >>> Personally I still have a hunch that mod_b64 is a subtle April Fool's Day joke. Even though it's a joke, it's still a very functional codec. ;) > >>> > >>>> -----Original Message----- > >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > >>>> users-bounces at lists.freeswitch.org] On Behalf Of Fran?ois > >>>> Sent: Tuesday, April 2, 2013 8:41 AM > >>>> To: freeswitch-users at lists.freeswitch.org > >>>> Subject: [Freeswitch-users] B64 audio codec? > >>>> > >>>> Hi all, > >>>> > >>>> Anyone has some info about the recently commited mod_b64 codec? > >>>> > >>>> The source refers to it as "The B64 ultra-low delay audio codec", and links to a > >>>> webpage www.b64-codec.org which does not exist! > >>>> > >>>> Anyway, it sounds interesting and doesn't seem to be an april fools joke! > >>>> > >>>> Thanks, > >>>> Fran?ois. > >>>> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/ff74bbe7/attachment-0001.html From schoch+freeswitch.org at xwin32.com Sat Apr 6 03:55:13 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Fri, 5 Apr 2013 16:55:13 -0700 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: <20130119191943.ca09ab6d@mail.tritonwest.net> <5102AF93.2000201@coppice.org> Message-ID: I'm still having problems. Is there something I've overlooked? I'm using HylaFAX talking to FreeSWITCH through a mod_spandsp modem, connected to the gateway Flowroute, sending to a FAX number hosted by eFax.com. HylaFAX reports "no carrier". Here is the log of the call, with SIP tracing (the IP address of my PBX redacted): http://pastebin.freeswitch.org/20769 -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/e15d4b6d/attachment.html From schoch+freeswitch.org at xwin32.com Sat Apr 6 04:08:30 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Fri, 5 Apr 2013 17:08:30 -0700 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: <20130119191943.ca09ab6d@mail.tritonwest.net> <5102AF93.2000201@coppice.org> Message-ID: Sorry, false alarm. The FAX was actually received successfully. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130405/e79cd574/attachment.html From shayne.alone at gmail.com Sat Apr 6 08:56:38 2013 From: shayne.alone at gmail.com (shayne.alone at gmail.com) Date: Sat, 6 Apr 2013 09:26:38 +0430 Subject: [Freeswitch-users] central Configuration via curl XML Message-ID: Hi all I'm looking for a way for get FS configuration out of it's server as much as I can... * I'm not looking for dynamic configuration via DB and just need to have one instance of configuration files in static XML on a web server to serve all freeswitch instances. due to curl_XML wiki: we can bind a section which name is "configuration"! but we have not any corresponding folder to serve it... what dose it mean? and how can I find more about the way curl_xml works? in deep... -- Regards, Ali R. Taleghani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130406/41331509/attachment.html From ashish at nms.co.in Sat Apr 6 09:20:44 2013 From: ashish at nms.co.in (Ashish gautam) Date: Sat, 6 Apr 2013 10:50:44 +0530 Subject: [Freeswitch-users] Normal Circuit congestion Message-ID: I am getting this error while making a PSTN outgoing call through PRI card using FreeTDM Module. 2013-04-05 15:04:03.454536 [DEBUG] switch_event.c:1608 Parsing variable [voiceMessageID]=[109] 2013-04-05 15:04:03.454536 [DEBUG] switch_event.c:1608 Parsing variable [respreqd]=[1] 2013-04-05 15:04:03.454536 [DEBUG] switch_event.c:1608 Parsing variable [mobnum]=[9818324290] 2013-04-05 15:04:03.454536 [DEBUG] switch_event.c:1608 Parsing variable [lang]=[en] 2013-04-05 15:04:03.454536 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true] 2013-04-05 15:04:03.454536 [DEBUG] switch_event.c:1608 Parsing variable [continue_on_fail]=[true] f42a6418-9dd3-11e2-b11d-6fb3d8274ed5 2013-04-05 15:04:03.454536 [NOTICE] mod_freetdm.c:1766 Close Channel N/A [CS_NEW] f42a6418-9dd3-11e2-b11d-6fb3d8274ed5 2013-04-05 15:04:03.454536 [DEBUG] switch_core_state_machine.c:568 () Running State Change CS_DESTROY f42a6418-9dd3-11e2-b11d-6fb3d8274ed5 2013-04-05 15:04:03.454536 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY f42a6418-9dd3-11e2-b11d-6fb3d8274ed5 2013-04-05 15:04:03.454536 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY going to sleep 2013-04-05 15:04:03.454536 [NOTICE] switch_ivr_originate.c:2636 Cannot create outgoing channel of type [freetdm] cause: [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 15:04:03.454536 [DEBUG] switch_ivr_originate.c:3601 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 15:04:03.454536 [DEBUG] ftdm_state.c:541 [s1c1][1:1] Executing state processor for DIALING The calls are not being dialled properly despite of the channels being free to use. Please help me getting out of this issue. -Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130406/9ea6f728/attachment.html From shaheryarkh at gmail.com Sat Apr 6 11:04:39 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Sat, 6 Apr 2013 08:04:39 +0100 Subject: [Freeswitch-users] central Configuration via curl XML In-Reply-To: References: Message-ID: Did you tried reading the wiki? http://wiki.freeswitch.org/wiki/Mod_xml_curl It has everything from very basic to very advance configuration as well as samples in many programming languages. Thank you. On Sat, Apr 6, 2013 at 5:56 AM, shayne.alone at gmail.com < shayne.alone at gmail.com> wrote: > Hi all > > I'm looking for a way for get FS configuration out of it's server as much > as I can... > > * I'm not looking for dynamic configuration via DB and just need to have > one instance of configuration files in static XML on a web server to serve > all freeswitch instances. > > due to curl_XML wiki: > we can bind a section which name is "configuration"! but we have not any > corresponding folder to serve it... > what dose it mean? and how can I find more about the way curl_xml works? > in deep... > > -- > Regards, > Ali R. Taleghani > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130406/0871f4ec/attachment.html From ashish at nms.co.in Sat Apr 6 11:18:36 2013 From: ashish at nms.co.in (Ashish gautam) Date: Sat, 6 Apr 2013 12:48:36 +0530 Subject: [Freeswitch-users] Default maximum number of simultaneous outgoing PSTN calls Message-ID: Hi, I am making simultaneous outgoing calls to PSTN network through freeTDM and PRI card via Event socket. FS only generates at max. four calls and for the rest it shows NORMAL_CIRCUIT_CONGESTION. What could be the possible reason for this? Is there any default maximum limit on the number of simultaneous outgoing calls through freeTDM/libpri/DAHDI stack.? Kindly throw some light on this. Regards. -- Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130406/479cc70c/attachment-0001.html From steveayre at gmail.com Sat Apr 6 15:16:55 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 6 Apr 2013 12:16:55 +0100 Subject: [Freeswitch-users] central Configuration via curl XML In-Reply-To: References: Message-ID: In particular read http://wiki.freeswitch.org/wiki/Xml_curl#.3Cbinding.3E_options (the settings are common to all binding types) And this part for examples: http://wiki.freeswitch.org/wiki/Xml_curl#Section:_configuration_2 On 6 April 2013 08:04, Muhammad Shahzad wrote: > Did you tried reading the wiki? > > http://wiki.freeswitch.org/wiki/Mod_xml_curl > > It has everything from very basic to very advance configuration as well as > samples in many programming languages. > > Thank you. > > > > > > On Sat, Apr 6, 2013 at 5:56 AM, shayne.alone at gmail.com < > shayne.alone at gmail.com> wrote: > >> Hi all >> >> I'm looking for a way for get FS configuration out of it's server as much >> as I can... >> >> * I'm not looking for dynamic configuration via DB and just need to have >> one instance of configuration files in static XML on a web server to serve >> all freeswitch instances. >> >> due to curl_XML wiki: >> we can bind a section which name is "configuration"! but we have not any >> corresponding folder to serve it... >> what dose it mean? and how can I find more about the way curl_xml works? >> in deep... >> >> -- >> Regards, >> Ali R. Taleghani >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Mit freundlichen Gr??en > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.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/20130406/bf191abf/attachment.html From ira at connectmevoice.com Sat Apr 6 17:28:48 2013 From: ira at connectmevoice.com (Ira Tessler) Date: Sat, 6 Apr 2013 09:28:48 -0400 Subject: [Freeswitch-users] Presence Sanity Check Message-ID: I just need a little guidance with the way presence works. Forgive me if I am asking novice questions. Background (simple version) We run Freeswitch in a hosted/cloud environment in a data center. We have IP phones in our office on our LAN. That way I am understanding how Presence works, I am just learning this, is that when a BLF button is programmed on a phone, that phone will send a "Subscribe" message to Freeswitch. The subscriptions are stored in the sip_subscriptions table (i think) in the sofia database for the sip profile. When calls come in for that subscription, Freeswitch will send out a NOTIFY message to the phone that subscribed in order to change the state of the BLF Light. He is my questions/issue/confusion. All our phones use UDP which has a maximum packet size of 1500 bytes. When doing a sofia global siptrace on, I notice that most of the NOTIFY messages are greater then 1500 bytes. That will cause packet fragmentation. So if the NOTIFY message is fragmented, will it get to the phone correctly? (all the time, some of the time, never??) If the the answer is other then ("all the time"), how do I fix this? The only solution I can come up with is having my phones use TCP instead of UDP. Is that the correctly solution? Did anyone else out there run into this issue and if so, what is the "best practice" solution (if there is one)? Thank you in advance! Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130406/00053bd2/attachment.html From fvillarroel at yahoo.com Sat Apr 6 18:10:04 2013 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Sat, 6 Apr 2013 07:10:04 -0700 (PDT) Subject: [Freeswitch-users] mod_python question In-Reply-To: <1628059.cgEIOWz82P@zeus> Message-ID: <1365257404.83299.YahooMailClassic@web162004.mail.bf1.yahoo.com> I am not sure But there are many apis and librarys that have not supported for Python 3.x So i think you must use Python 2.x for the moment. --- On Fri, 4/5/13, Derrick Dantavious Edwards wrote: > From: Derrick Dantavious Edwards > Subject: [Freeswitch-users] mod_python question > To: freeswitch-users at lists.freeswitch.org > Date: Friday, April 5, 2013, 3:34 PM > ??? Hi, > It seems that mod_python defaults to Python version 2.7. I > would like for it > to use verison 3.3. I have version 3.3 installed on system > and attempted to > edit the mod_python Makefile to point to installed Python > verison 3.3 but it > failed on the compile. Any ideas ? > > V/r > Derrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Tim.Meade at Millicorp.com Sat Apr 6 19:08:25 2013 From: Tim.Meade at Millicorp.com (Tim Meade) Date: Sat, 6 Apr 2013 15:08:25 +0000 Subject: [Freeswitch-users] Current master 1.5.1b? In-Reply-To: References: <804D48104511D4468F0D60DF9D31003511A9EB69@MAIL.millicorp.com> Message-ID: <804D48104511D4468F0D60DF9D31003511ADA8A5@MAIL.millicorp.com> Thanks. Just a shock to the grey matter to see the version jump From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, April 04, 2013 10:08 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Current master 1.5.1b? Nope its 1.5 now. HEAD is odd revs feeding the preceding even. 1.3 fed 1.2 until it was discontinued for 1.5. Now 1.2 runs on its own and 1.5 has the latest dev brewing releases for 1.4. The message you are seeing is because you have the debug_level higher than 0. It was changed to go on alert level so it could be distinguished from other log messages. fsctl debug_level 0 to turn it off. On Thu, Apr 4, 2013 at 8:14 PM, Tim Meade > wrote: I just did a git clone and built that latest master. Here is what I'm getting from version: FreeSWITCH Version 1.5.1b+git~20130405T004141Z~2cef8580a0 (git 2cef858 2013-04-05 00:41:41Z) Is this a typo? 3 weeks ago it was 1.3.17 FreeSWITCH Version 1.3.17+git~20130321T141651Z~ac0defb874 (git ac0defb 2013-03-21 14:16:51Z) Also; this build is showing these alerts which we have not seen before. 2013-04-04 21:12:00.695788 [ALERT] sofia_glue.c:2430 Reuse Unused Cached DB handle db="sofia_reg_internal" [CORE_DB] 2013-04-04 21:12:00.695788 [ALERT] sofia_glue.c:2430 Reuse Unused Cached DB handle db="sofia_reg_internal" [CORE_DB] 2013-04-04 21:12:00.755821 [ALERT] switch_console.c:253 Create Cached DB handle (null) [CORE_DB] src/switch_console.c:253 2013-04-04 21:12:30.695446 [ALERT] sofia_glue.c:2430 Reuse Unused Cached DB handle db="sofia_reg_internal" [CORE_DB] 2013-04-04 21:12:30.695446 [ALERT] sofia_glue.c:2430 Reuse Unused Cached DB handle db="sofia_reg_internal" [CORE_DB] Otherwise things are working as advertised. Thanks Tim _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130406/ed338dc7/attachment-0001.html From anouarabrik at gmail.com Sun Apr 7 02:14:49 2013 From: anouarabrik at gmail.com (Anouar Abrik) Date: Sun, 7 Apr 2013 03:14:49 +0500 Subject: [Freeswitch-users] Skypopen: A suggestion and some random things to point out Message-ID: Hi, I have been trying to actually try to run skypopen on my ubuntu for quite a while, and while I had loads of problems trying to, I finally managed to make it work, the wiki guide written by Mr. Giovanni has been a great help, yet I'd like to point some things out, to help people who might ever face the same problem I faced. I could not load the skypopen.ko module into the kernel, I realized that not just SND_OSS should be removed, but any SND driver in general, I had to remove each one individually to finally be able to insert the module. I even think now that a desktop distro won't actually be a problem to run the mod in, I am yet to try it myself, but I do believe it possible easily, if the guide is followed properly and if the module is loaded. Finally, I'd like to thank Mr.Giovanni for the help and support. -- Regards, Anouar Abrik *Graphics Designer* Direct Number(PK): +92-336-229-4557 Skype: donnymaniac * * * * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130407/6eda5fa3/attachment.html From ebrahim.bararian at gmail.com Sun Apr 7 02:36:20 2013 From: ebrahim.bararian at gmail.com (Ebrahim Bararian) Date: Sun, 7 Apr 2013 03:06:20 +0430 Subject: [Freeswitch-users] Problem With Running Mod_pocketsphinx Message-ID: Hi, I've just tried to run the pocket_sphinx according to what the link below described: http://wiki.freeswitch.org/wiki/Mod_pocketsphinx But when I try to run by using command "pa call 74992" in the freeswitch console, the following happens: *[ERR] mod_pocketsphinx.c:99 In valid rate 48000. Only 8000 and 16000 are supported. * * * *[ERR] mod_pocketsphinx.c:142 Can't open dictionary C: /Program Files/FreeSWITCH/grammar/default.dic. * * * but how can I fix it? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130407/b1a5f09c/attachment.html From ebrahim.bararian at gmail.com Sun Apr 7 03:29:02 2013 From: ebrahim.bararian at gmail.com (Ebrahim Bararian) Date: Sun, 7 Apr 2013 03:59:02 +0430 Subject: [Freeswitch-users] Problem With Running Mod_pocketsphinx In-Reply-To: References: Message-ID: I put the pocketsphinx dictionary and acoustic model in the place it needed it. So that error have been solved. But the first error is lasting yet. How can I change the sampling rate from 48k to 16k or 8k? On Sun, Apr 7, 2013 at 3:06 AM, Ebrahim Bararian wrote: > Hi, > > I've just tried to run the pocket_sphinx according to what the link below > described: > > http://wiki.freeswitch.org/wiki/Mod_pocketsphinx > > But when I try to run by using command "pa call 74992" in the freeswitch > console, the following happens: > > *[ERR] mod_pocketsphinx.c:99 In valid rate 48000. Only 8000 and 16000 are > supported. > * > * > * > *[ERR] mod_pocketsphinx.c:142 Can't open dictionary C: /Program > Files/FreeSWITCH/grammar/default.dic. * > * > * > but how can I fix it? > > Thanks in advance. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130407/20bc8de0/attachment.html From shaharhd at gmail.com Sun Apr 7 03:59:28 2013 From: shaharhd at gmail.com (Shahar) Date: Sat, 6 Apr 2013 19:59:28 -0400 Subject: [Freeswitch-users] Spandsp modems as non-root? In-Reply-To: <515ED539.8030105@coppice.org> References: <50703D76.7010301@integrafin.co.uk> <515C30EF.8020301@integrafin.co.uk> <515D9170.9090208@coppice.org> <515ED539.8030105@coppice.org> Message-ID: Steve, I was under the impression that spandsp does implement a V.29 (9.6K) with V.42bis (compression and error control). But I might be wrong here. On Fri, Apr 5, 2013 at 9:44 AM, Steve Underwood wrote: > On 04/05/2013 03:19 AM, Shahar wrote: > > were not thinking about utilizing spandsp for the dialup - we just > > want it to create the com devices. > If it creates the com devices, what will be your modem? > > > > then were passing the handling via mgetty to pppd - just like hylafax > > takes over the spandsp in the data handling. > > maybe we need to disable the T.38 for this to work (and stay in G711) ? > Are you under the impression there are duplex data modems in spandsp? If > so, its a false impression. There is a V.22bis modem, but it is not > exposed in the AT interface provided for the FAX modems. > > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130406/4af7ba12/attachment.html From ebrahim.bararian at gmail.com Sun Apr 7 04:17:04 2013 From: ebrahim.bararian at gmail.com (Ebrahim Bararian) Date: Sun, 7 Apr 2013 04:47:04 +0430 Subject: [Freeswitch-users] Problem With Running Mod_pocketsphinx In-Reply-To: References: Message-ID: I found it. It was in the file portaudio.conf.xml. I changed the sample rate to 16000. On Sun, Apr 7, 2013 at 3:59 AM, Ebrahim Bararian wrote: > I put the pocketsphinx dictionary and acoustic model in the place it > needed it. So that error have been solved. > > But the first error is lasting yet. How can I change the sampling rate > from 48k to 16k or 8k? > > > On Sun, Apr 7, 2013 at 3:06 AM, Ebrahim Bararian < > ebrahim.bararian at gmail.com> wrote: > >> Hi, >> >> I've just tried to run the pocket_sphinx according to what the link below >> described: >> >> http://wiki.freeswitch.org/wiki/Mod_pocketsphinx >> >> But when I try to run by using command "pa call 74992" in the freeswitch >> console, the following happens: >> >> *[ERR] mod_pocketsphinx.c:99 In valid rate 48000. Only 8000 and 16000 >> are supported. >> * >> * >> * >> *[ERR] mod_pocketsphinx.c:142 Can't open dictionary C: /Program >> Files/FreeSWITCH/grammar/default.dic. * >> * >> * >> but how can I fix it? >> >> Thanks in advance. >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130407/7155297e/attachment.html From abefried1 at yahoo.com Sat Apr 6 14:11:49 2013 From: abefried1 at yahoo.com (jow) Date: Sat, 6 Apr 2013 03:11:49 -0700 (PDT) Subject: [Freeswitch-users] Do I need a new dns resource records for each tenant? Message-ID: <1365243109613-7589470.post@n2.nabble.com> The subject says it all. do I need a new dns resource records for each new tenant? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Do-I-need-a-new-dns-resource-records-for-each-tenant-tp7589470.html Sent from the freeswitch-users mailing list archive at Nabble.com. From hajime9ma at gmail.com Sun Apr 7 00:20:47 2013 From: hajime9ma at gmail.com (hajime) Date: Sat, 6 Apr 2013 13:20:47 -0700 (PDT) Subject: [Freeswitch-users] Presence in ODBC/PgSQL? Message-ID: <1365279647653-7589475.post@n2.nabble.com> My first question in the mailing list, so hope you can bear over with me :-) Been trawling through the mailing list and google alas not yet found a sound solution to get presence data stored in a database - say I'd like to expose the users' online status via a web page ( I'd like to avoid having to install Kamailio which could manage this for me ). Is the simplest option to subscribe to events (Mod event socket) and write them myself into a database? Or is there an easier way to achieve this. If you could provide me some pointers I'd really appreciate it. Thanks, Hajime -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Presence-in-ODBC-PgSQL-tp7589475.html Sent from the freeswitch-users mailing list archive at Nabble.com. From kashif at kashifbukhari.com Sun Apr 7 00:41:33 2013 From: kashif at kashifbukhari.com (Kashif Ali) Date: Sun, 7 Apr 2013 01:41:33 +0500 Subject: [Freeswitch-users] Dynamic specify the outbound GW within source code In-Reply-To: References: Message-ID: On Mon, Jan 9, 2012 at 11:50 AM, fieldpeak wrote: > Hi Avi, > > Thanks so much for your kindly reply. > > Actually, now i'm using mod_nibble for billing, i write a function > "check_billing_before_routing" in nibble_state_handler, in this > func("check_billing_before_routing"), it will call an external command, > this command will query the backend database if the caller has enough money > to contiue the call, the mod_nibblebill will contiue the call or hangup the > call according to the result of the external command. i have realize all > above, it works well. > > switch_state_handler_table_t nibble_state_handler = { > /* on_init */ NULL, > /* on_routing */ check_billing_before_routing, /* Need to add a check > here for anything in their account before routing */ > /* on_execute */ sched_billing, /* Turn on heartbeat for this > session and do an initial account check */ > /* on_hangup */ process_hangup, /* On hangup - most important > place to go bill */ > /* on_exch_media */ NULL, > /* on_soft_exec */ NULL, > /* on_consume_med */ NULL, > /* on_hibernate */ NULL, > /* on_reset */ NULL, > /* on_park */ NULL, > /* on_reporting */ NULL, > /* on_destroy */ NULL > }; > > For PSTN call, i use dial plan below, "1.2.3.4" is the PSTN-GW > > > > > > > > > Now, as we add one more PSTN-GW for outbound call, and the FS have to > route call to the specific GW accoring to result of the external command > (the external command will return the IP address of GW as well), > > i can think out the FS own function like > "switch_channel_set_variable(channel, "caller_id_number")" can configure > the value of variable, however, what variable should i use for this case, > could you please advise, thank you very much! > > Regards, > Charels > > 2012/1/8 Avi Marcus > >> I'm not quite sure of the use case. Do any of these help? >> 1) specify a server, not an IP, and then let DNS determine where it goes. >> 2) use a small lua script to set the channel variable based on whatever >> you need - an sql query, some logic.. and then use that variable in the >> bridge string. >> >> Those help? If not, please explain more what problem you are trying to >> solve. >> >> -Avi >> >> >> On Sun, Jan 8, 2012 at 3:34 PM, fieldpeak wrote: >> >>> Dear friends, >>> >>> i have FS for PSTN outbound call using below dial plan, >>> >>> >>> >>> >>> >>> >>> >>> While, now i need dynamically specify the outbound GW?s IP address >>> according to the return result of the external command before routing in >>> the source code , e.g. if the external command return FS the IP address of >>> OB GW 6.7.8.9, then >>> >>> >>> >>> however, i don't know which function i should call within the source >>> code to realize it, could anybody help to advise, >>> >>> P.S. i know there is existing module ?mod_xml_curl? can realize similar >>> function, however, I could not use it for this case? >>> >>> >>> thanks a lot! >>> >>> Regards, >>> Charles >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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, > Charles > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130407/d1eb0d63/attachment-0001.html From ml88888 at hotmail.com Sun Apr 7 02:00:12 2013 From: ml88888 at hotmail.com (FSX) Date: Sat, 6 Apr 2013 15:00:12 -0700 (PDT) Subject: [Freeswitch-users] Build errors in 2010.express solution Message-ID: <1365285612276-7589476.post@n2.nabble.com> I'm trying to build "FreeSWITCH.2010.express" solution with latest sources from git and getting lot of errors (see below). Git [2013-04-06]: 4ccfb47 --------------------------------------- Creating library C:\Dev\FS\Win32\Release\mod\mod_fifo.lib and object C:\Dev\FS\Win32\Release\mod\mod_fifo.exp mod_fifo.obj : error LNK2001: unresolved external symbol __imp__switch_channel_invert_cid at 4 C:\Dev\FS\Win32\Release\mod\mod_fifo.dll : fatal error LNK1120: 1 unresolved externals Creating library C:\Dev\FS\Win32\Release\mod\mod_sofia.lib and object C:\Dev\FS\Win32\Release\mod\mod_sofia.exp mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_pass_zrtp_hash2 at 8 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_codec_chosen at 8 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_prepare_codecs at 8 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_proxy_remote_addr at 8 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_start_udptl at 8 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_kill_socket at 8 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_read_frame at 20 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_session_set_ice at 4 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_media_handle_destroy at 4 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_set_local_sdp at 12 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_check_outgoing_proxy at 8 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_get_jb at 8 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_break at 8 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_set_udptl_image_sdp at 12 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_reset_autofix_timing at 8 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_toggle_hold at 8 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_choose_port at 12 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_patch_sdp at 4 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_queue_rfc2833 at 12 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_check_dtmf_type at 4 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_gen_local_sdp at 20 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_udptl_mode at 8 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_write_frame at 20 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_absorb_sdp at 4 mod_sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_ready at 8 rtp.obj : error LNK2001: unresolved external symbol __imp__switch_rtp_clear_flags at 8 rtp.obj : error LNK2001: unresolved external symbol __imp__switch_rtp_set_flags at 8 sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_process_t38_passthru at 12 sofia.obj : error LNK2001: unresolved external symbol _nutag_wss_url sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_parse_rtp_bugs at 8 sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_set_sdp_codec_string at 8 sofia.obj : error LNK2001: unresolved external symbol _nutag_ws_url sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_extract_t38_options at 8 sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_gen_certs at 4 sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_session_local_crypto_key at 8 sofia.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_clear_rtp_flag at 12 sofia_glue.obj : error LNK2001: unresolved external symbol __imp__switch_media_handle_set_media_flags at 8 sofia_glue.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_get_stats at 12 sofia_glue.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_add_ice_acl at 12 sofia_glue.obj : error LNK2001: unresolved external symbol __imp__switch_core_session_get_recovery_crypto_key at 12 sofia_glue.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_recover_session at 4 sofia_glue.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_check_video_codecs at 4 sofia_glue.obj : error LNK2001: unresolved external symbol __imp__switch_media_handle_create at 12 sofia_media.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_negotiate_sdp at 12 sofia_media.obj : error LNK2001: unresolved external symbol __imp__switch_core_media_activate_rtp at 4 C:\Dev\FS\Win32\Release\mod\mod_sofia.dll : fatal error LNK1120: 45 unresolved externals Creating library C:\Dev\FS\Win32\Release\mod\mod_dingaling.lib and object C:\Dev\FS\Win32\Release\mod\mod_dingaling.exp mod_dingaling.obj : error LNK2001: unresolved external symbol __imp__switch_rtp_activate_rtcp at 16 mod_dingaling.obj : error LNK2001: unresolved external symbol __imp__switch_rtp_activate_ice at 32 C:\Dev\FS\Win32\Release\mod\mod_dingaling.dll : fatal error LNK1120: 2 unresolved externals --------------------------------------- Is there any chance to get this fixed? I can't build it for the past week, every time getting similar errors... Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Build-errors-in-2010-express-solution-tp7589476.html Sent from the freeswitch-users mailing list archive at Nabble.com. From shaheryarkh at gmail.com Sun Apr 7 10:09:50 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Sun, 7 Apr 2013 07:09:50 +0100 Subject: [Freeswitch-users] Skypopen: A suggestion and some random things to point out In-Reply-To: References: Message-ID: Good, You can always update the module wiki with your observations and workarounds. Thank you. On Sat, Apr 6, 2013 at 11:14 PM, Anouar Abrik wrote: > Hi, > > I have been trying to actually try to run skypopen on my ubuntu for quite > a while, and while I had loads of problems trying to, I finally managed to > make it work, the wiki guide written by Mr. Giovanni has been a great help, > yet I'd like to point some things out, to help people who might ever face > the same problem I faced. > I could not load the skypopen.ko module into the kernel, I realized that > not just SND_OSS should be removed, but any SND driver in general, I had to > remove each one individually to finally be able to insert the module. I > even think now that a desktop distro won't actually be a problem to run the > mod in, I am yet to try it myself, but I do believe it possible easily, if > the guide is followed properly and if the module is loaded. > > Finally, I'd like to thank Mr.Giovanni for the help and support. > > -- > Regards, > > Anouar Abrik > > *Graphics Designer* > > Direct Number(PK): +92-336-229-4557 > Skype: donnymaniac > > > * > * > * * > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130407/8cdd420c/attachment.html From gerald.weber at besharp.at Sun Apr 7 12:33:37 2013 From: gerald.weber at besharp.at (Gerald Weber) Date: Sun, 7 Apr 2013 08:33:37 +0000 Subject: [Freeswitch-users] Presence in ODBC/PgSQL? In-Reply-To: <1365279647653-7589475.post@n2.nabble.com> References: <1365279647653-7589475.post@n2.nabble.com> Message-ID: FreeSWITCH offers presence out oft he box, if you have it enabled in the sofia profile, the tables are already there. Table names starting with sip_ like sip_subscriptions or sip_presence (the table you are looking for) The problem is that the sip_presence table is always empty for me, because of the code checking for various things like multiple reigstrations, etc. I'm currenlty working on an patch/alternative solution. When i figured this out, i plan to put presence into the list_users command so you have an overview of all your users in the directory at a glance. -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von hajime Gesendet: Samstag, 06. April 2013 22:21 An: freeswitch-users at lists.freeswitch.org Betreff: [Freeswitch-users] Presence in ODBC/PgSQL? My first question in the mailing list, so hope you can bear over with me :-) Been trawling through the mailing list and google alas not yet found a sound solution to get presence data stored in a database - say I'd like to expose the users' online status via a web page ( I'd like to avoid having to install Kamailio which could manage this for me ). Is the simplest option to subscribe to events (Mod event socket) and write them myself into a database? Or is there an easier way to achieve this. If you could provide me some pointers I'd really appreciate it. Thanks, Hajime -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Presence-in-ODBC-PgSQL-tp7589475.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 sherifomran2000 at yahoo.com Sun Apr 7 13:09:47 2013 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 7 Apr 2013 02:09:47 -0700 (PDT) Subject: [Freeswitch-users] How to: Setup IVR message menu In-Reply-To: Message-ID: <1365325787.28691.YahooMailClassic@web141204.mail.bf1.yahoo.com> Hello guys, I am trying to play an ivr menu but seem to be missing a point, i saw the demo ivr main menu example and now changed its name into salah_ivr but i get switch_ivr_play_say.c:142 Can't find macro salah_ivr_main_menu. where could the error be? It does not play any sound. I don't know what should i call the sound file, in case i don't use flite tts? ????? greet-long="say: welcome welcome" ????? greet-short="phrase:salah_ivr_main_menu_short" ????? invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" ????? exit-sound="voicemail/vm-goodbye.wav" ????? confirm-macro="" ????? confirm-key="" ????? tts-engine="flite" ????? tts-voice="rms" ????? confirm-attempts="3" ????? timeout="10000" ????? inter-digit-timeout="2000" ????? max-failures="3" ????? max-timeouts="3" ????? digit-len="1"> ??? > greet-long="say: welcome welcome" > greet-short="phrase:salah_ivr_main_menu_short" > invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" > exit-sound="voicemail/vm-goodbye.wav" > confirm-macro="" > confirm-key="" > tts-engine="flite" > tts-voice="rms" > confirm-attempts="3" > timeout="10000" > inter-digit-timeout="2000" > max-failures="3" > max-timeouts="3" > digit-len="1"> > corresponding folder to serve it... > what dose it mean? and how can I find more about the way curl_xml works? > in deep... > > -- > Regards, > Ali R. Taleghani > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130407/940c345a/attachment.html From avi at avimarcus.net Sun Apr 7 20:47:23 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 7 Apr 2013 19:47:23 +0300 Subject: [Freeswitch-users] central Configuration via curl XML In-Reply-To: References: Message-ID: Where's the part about "if you just want central configuration, you can use pre-process wget"? I can't find it... -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130407/71e6da4c/attachment.html From cal.leeming at simplicitymedialtd.co.uk Sun Apr 7 20:49:45 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 7 Apr 2013 17:49:45 +0100 Subject: [Freeswitch-users] Presence Sanity Check In-Reply-To: References: Message-ID: In regards to the UDP fragmentation, this is an extremely good question. Only yesterday I started to build a simple forwarding SBC in Python using UDP sockets, however I came up against the same theoretical problem of packets being larger than 1500 bytes. I've had a read through various documentation; http://www.rfc-ref.org/RFC-TEXTS/3261/chapter18.html http://www.ietf.org/rfc/rfc3428.txt https://lists.cs.columbia.edu/pipermail/sip-implementors/2006-August/013857.html http://lists.freeswitch.org/pipermail/freeswitch-users/2011-February/068372.html http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05912.html Anthony has stated the following; *"the only reliable answer is use TCP."* There was also the option of enabling compact headers; http://lists.freeswitch.org/pipermail/freeswitch-users/2011-December/078633.html >From what I can tell, there is no way to guarantee this problem won't happen unless you use TCP. You could reduce the packet size by compacting headers or removing codecs, but this would be on the assumption that every hop is running at 1500 MTU. Hope this helps! Cal On Sat, Apr 6, 2013 at 2:28 PM, Ira Tessler wrote: > I just need a little guidance with the way presence works. Forgive me if I > am asking novice questions. > > Background (simple version) > We run Freeswitch in a hosted/cloud environment in a data center. We have > IP phones in our office on our LAN. > > That way I am understanding how Presence works, I am just learning this, > is that when a BLF button is programmed on a phone, that phone will send a > "Subscribe" message to Freeswitch. The subscriptions are stored in the > sip_subscriptions table (i think) in the sofia database for the sip > profile. When calls come in for that subscription, Freeswitch will send out > a NOTIFY message to the phone that subscribed in order to change the state > of the BLF Light. > > He is my questions/issue/confusion. > All our phones use UDP which has a maximum packet size of 1500 bytes. When > doing a sofia global siptrace on, I notice that most of the NOTIFY messages > are greater then 1500 bytes. That will cause packet fragmentation. So if > the NOTIFY message is fragmented, will it get to the phone correctly? (all > the time, some of the time, never??) > > If the the answer is other then ("all the time"), how do I fix this? The > only solution I can come up with is having my phones use TCP instead of > UDP. Is that the correctly solution? Did anyone else out there run into > this issue and if so, what is the "best practice" solution (if there is > one)? > > Thank you in advance! > > Ira Tessler > Lead Software Engineer > ConnectMe > (732) 490-9007 x2 > ira at connectmevoice.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/20130407/2b4344eb/attachment.html From cal.leeming at simplicitymedialtd.co.uk Sun Apr 7 20:54:40 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 7 Apr 2013 17:54:40 +0100 Subject: [Freeswitch-users] central Configuration via curl XML In-Reply-To: References: Message-ID: Curious, didn't seem to make it into the final cut. I've re-added it back in. http://wiki.freeswitch.org/wiki/Mod_xml_curl#Alternative_ways_of_storing_static_configurations Cal On Sun, Apr 7, 2013 at 5:47 PM, Avi Marcus wrote: > Where's the part about "if you just want central configuration, you can > use pre-process wget"? I can't find it... > -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/20130407/042a4d92/attachment-0001.html From brian at freeswitch.org Mon Apr 8 05:41:19 2013 From: brian at freeswitch.org (Brian West) Date: Sun, 7 Apr 2013 20:41:19 -0500 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: <20130119191943.ca09ab6d@mail.tritonwest.net> <5102AF93.2000201@coppice.org> Message-ID: Not too sure you need t38_gateway on this call flow. I'll have to double check that. On Feb 15, 2013, at 7:51 PM, Steven Schoch wrote: > -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://freeswitchcookbook.com http://freeswitchbook.com T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 From brian at freeswitch.org Mon Apr 8 05:42:04 2013 From: brian at freeswitch.org (Brian West) Date: Sun, 7 Apr 2013 20:42:04 -0500 Subject: [Freeswitch-users] Default maximum number of simultaneous outgoing PSTN calls In-Reply-To: References: Message-ID: <3E944195-31F2-4239-A9FB-D9F9D5758557@freeswitch.org> How many calls are you trying to start at once? You could just be pissing off the switch on the other side by bring up too many calls quickly. On Apr 6, 2013, at 2:18 AM, Ashish gautam wrote: > Hi, > > I am making simultaneous outgoing calls to PSTN network through freeTDM and > PRI card via Event socket. FS only generates at max. four calls and for the > rest it shows NORMAL_CIRCUIT_CONGESTION. > > What could be the possible reason for this? Is there any default maximum > limit on the number of simultaneous outgoing calls through > freeTDM/libpri/DAHDI stack.? > > Kindly throw some light on this. > > Regards. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://freeswitchcookbook.com http://freeswitchbook.com T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 From brian at freeswitch.org Mon Apr 8 05:43:10 2013 From: brian at freeswitch.org (Brian West) Date: Sun, 7 Apr 2013 20:43:10 -0500 Subject: [Freeswitch-users] Problem With Running Mod_pocketsphinx In-Reply-To: References: Message-ID: The 16k acoustical model isn't as robust as the 8k version. Just an FYI. On Apr 6, 2013, at 7:17 PM, Ebrahim Bararian wrote: > I found it. It was in the file portaudio.conf.xml. I changed the sample > rate to 16000. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://freeswitchcookbook.com http://freeswitchbook.com T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 From khuenm at vega.com.vn Mon Apr 8 07:47:21 2013 From: khuenm at vega.com.vn (Khue Nguyen Minh) Date: Mon, 8 Apr 2013 10:47:21 +0700 Subject: [Freeswitch-users] CDR In-Reply-To: References: Message-ID: I was export cdr file success. Thank you very much. Now, I want write all dtmf into cdr file, how I can do that? Please help me. Brs, Khue 2013/4/5 Ken Rice > You don?t do that in the dialplan, mod_cdr_csv does that for you > > > > On 4/5/13 2:24 AM, "Khue Nguyen Minh" wrote: > > Hi all, > > I want create cdr (filetype: csv) file in freeswitch. But, I don't known > how I can do it from xml dialplan. Please guide me. > > Thanks & Best regards, > Khue. > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #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/20130408/8e3aaa8b/attachment.html From ashish at nms.co.in Mon Apr 8 08:40:49 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 8 Apr 2013 10:10:49 +0530 Subject: [Freeswitch-users] Default maximum number of simultaneous outgoing PSTN calls In-Reply-To: <3E944195-31F2-4239-A9FB-D9F9D5758557@freeswitch.org> References: <3E944195-31F2-4239-A9FB-D9F9D5758557@freeswitch.org> Message-ID: Thanks Brian, I am generating not more than 10 calls at a time from FreeSWITCH. There is a script that reads for any new entries from database and as soon as the rows are inserted and it generates calls through event socket simultaneously to all the new numbers. I have also tried to put a sleep of 250 ms in the script generating request to event socket. Below is the piece of code that executes whenever a new row is generated: my $dialstring = "api originate {voiceMessageID=$vmID,respreqd=$respReqd,mobnum=$mobileNum,lang=$language,ignore_early_media=true}freetdm/1/A/$mobileNum 47673501 XML public\n\n"; # initialize host and port $socket = IO::Socket::INET->new(PeerAddr => "127.0.0.1", PeerPort => 8021, Proto => "tcp", Type => SOCK_STREAM) or die "Couldn't connect to 10.1.30.229:8021 : $@\n"; print $socket "auth ClueCon\n\n\n"; $answer = <$socket>; print $answer; print $socket $dialstring; $socket->recv($data,1024); #$answer = <$socket>; print $data; close($socket); -- Ashish On Mon, Apr 8, 2013 at 7:12 AM, Brian West wrote: > How many calls are you trying to start at once? You could just be pissing > off the switch on the other side by bring up too many calls quickly. > On Apr 6, 2013, at 2:18 AM, Ashish gautam wrote: > > > Hi, > > > > I am making simultaneous outgoing calls to PSTN network through freeTDM > and > > PRI card via Event socket. FS only generates at max. four calls and for > the > > rest it shows NORMAL_CIRCUIT_CONGESTION. > > > > What could be the possible reason for this? Is there any default maximum > > limit on the number of simultaneous outgoing calls through > > freeTDM/libpri/DAHDI stack.? > > > > Kindly throw some light on this. > > > > Regards. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > http://freeswitchcookbook.com > http://freeswitchbook.com > > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130408/2dbe9804/attachment-0001.html From ashish at nms.co.in Mon Apr 8 09:09:53 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 8 Apr 2013 10:39:53 +0530 Subject: [Freeswitch-users] T316 timed out, resending RESTART request Message-ID: Hi, I am getting this warning: "2013-03-15 12:25:15.816238 [WARNING] ftmod_libpri.c:1954 [s1c29][1:29] -- T316 timed out, resending RESTART request" when I start my FS box. After these warnings with three restart attempts I get this error: 2013-03-15 12:25:45.816239 [ERR] ftmod_libpri.c:1950 [s1c31][1:31] -- T316 timed out, channel reached restart attempt limit '3' and is suspended I am also facing NORMAL_CIRCUIT_CONGESTION error when generation outgoing calls through FreeTDM. Is this a bug? Kindly help. -- Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130408/7ceac596/attachment.html From POlsson at enghouse.com Mon Apr 8 09:47:29 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 8 Apr 2013 05:47:29 +0000 Subject: [Freeswitch-users] Default maximum number of simultaneous outgoing PSTN calls In-Reply-To: References: <3E944195-31F2-4239-A9FB-D9F9D5758557@freeswitch.org>, Message-ID: You probably need to contact you PSTN provider to help you out here. You can also dump the FreeTDM traffic and analyze what happens on the signalling level. /Peter 8 apr 2013 kl. 06:48 skrev "Ashish gautam" >: Thanks Brian, I am generating not more than 10 calls at a time from FreeSWITCH. There is a script that reads for any new entries from database and as soon as the rows are inserted and it generates calls through event socket simultaneously to all the new numbers. I have also tried to put a sleep of 250 ms in the script generating request to event socket. Below is the piece of code that executes whenever a new row is generated: my $dialstring = "api originate {voiceMessageID=$vmID,respreqd=$respReqd,mobnum=$mobileNum,lang=$language,ignore_early_media=true}freetdm/1/A/$mobileNum 47673501 XML public\n\n"; # initialize host and port $socket = IO::Socket::INET->new(PeerAddr => "127.0.0.1", PeerPort => 8021, Proto => "tcp", Type => SOCK_STREAM) or die "Couldn't connect to 10.1.30.229:8021 : $@\n"; print $socket "auth ClueCon\n\n\n"; $answer = <$socket>; print $answer; print $socket $dialstring; $socket->recv($data,1024); #$answer = <$socket>; print $data; close($socket); -- Ashish On Mon, Apr 8, 2013 at 7:12 AM, Brian West > wrote: How many calls are you trying to start at once? You could just be pissing off the switch on the other side by bring up too many calls quickly. On Apr 6, 2013, at 2:18 AM, Ashish gautam wrote: > Hi, > > I am making simultaneous outgoing calls to PSTN network through freeTDM and > PRI card via Event socket. FS only generates at max. four calls and for the > rest it shows NORMAL_CIRCUIT_CONGESTION. > > What could be the possible reason for this? Is there any default maximum > limit on the number of simultaneous outgoing calls through > freeTDM/libpri/DAHDI stack.? > > Kindly throw some light on this. > > Regards. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://freeswitchcookbook.com http://freeswitchbook.com T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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:5162473332765878717253! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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:5162473332765878717253! From ashish at nms.co.in Mon Apr 8 11:20:26 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 8 Apr 2013 12:50:26 +0530 Subject: [Freeswitch-users] Default maximum number of simultaneous outgoing PSTN calls In-Reply-To: References: <3E944195-31F2-4239-A9FB-D9F9D5758557@freeswitch.org> Message-ID: Peter, The PSTN provider is saying there is no issue from their side, its on my side. On Mon, Apr 8, 2013 at 11:17 AM, Peter Olsson wrote: > You probably need to contact you PSTN provider to help you out here. You > can also dump the FreeTDM traffic and analyze what happens on the > signalling level. > > /Peter > > 8 apr 2013 kl. 06:48 skrev "Ashish gautam" ashish at nms.co.in>>: > > Thanks Brian, > > I am generating not more than 10 calls at a time from FreeSWITCH. There is > a script that reads for any new entries from database and as soon as the > rows are inserted and it generates calls through event socket > simultaneously to all the new numbers. I have also tried to put a sleep of > 250 ms in the script generating request to event socket. Below is the piece > of code that executes whenever a new row is generated: > > my $dialstring = "api originate > {voiceMessageID=$vmID,respreqd=$respReqd,mobnum=$mobileNum,lang=$language,ignore_early_media=true}freetdm/1/A/$mobileNum > 47673501 XML public\n\n"; > # initialize host and port > $socket = IO::Socket::INET->new(PeerAddr => "127.0.0.1", > PeerPort => 8021, > Proto => "tcp", > Type => SOCK_STREAM) > or die "Couldn't connect to 10.1.30.229:8021 : > $@\n"; > print $socket "auth ClueCon\n\n\n"; > $answer = <$socket>; > print $answer; > print $socket $dialstring; > $socket->recv($data,1024); > #$answer = <$socket>; > print $data; > close($socket); > -- > Ashish > > On Mon, Apr 8, 2013 at 7:12 AM, Brian West brian at freeswitch.org>> wrote: > How many calls are you trying to start at once? You could just be pissing > off the switch on the other side by bring up too many calls quickly. > On Apr 6, 2013, at 2:18 AM, Ashish gautam wrote: > > > Hi, > > > > I am making simultaneous outgoing calls to PSTN network through freeTDM > and > > PRI card via Event socket. FS only generates at max. four calls and for > the > > rest it shows NORMAL_CIRCUIT_CONGESTION. > > > > What could be the possible reason for this? Is there any default maximum > > limit on the number of simultaneous outgoing calls through > > freeTDM/libpri/DAHDI stack.? > > > > Kindly throw some light on this. > > > > Regards. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > http://freeswitchcookbook.com > http://freeswitchbook.com > > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official 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:5162473332765878717253! > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official 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://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org > > > !DSPAM:5162473332765878717253! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130408/7cf12e64/attachment.html From avi at avimarcus.net Mon Apr 8 11:52:38 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 8 Apr 2013 10:52:38 +0300 Subject: [Freeswitch-users] CDR In-Reply-To: References: Message-ID: Try logging: digits_dialed -Avi Marcus BestFone On Mon, Apr 8, 2013 at 6:47 AM, Khue Nguyen Minh wrote: > I was export cdr file success. Thank you very much. > > Now, I want write all dtmf into cdr file, how I can do that? Please help > me. > > Brs, > Khue > > > 2013/4/5 Ken Rice > >> You don?t do that in the dialplan, mod_cdr_csv does that for you >> >> >> >> On 4/5/13 2:24 AM, "Khue Nguyen Minh" wrote: >> >> Hi all, >> >> I want create cdr (filetype: csv) file in freeswitch. But, I don't known >> how I can do it from xml dialplan. Please guide me. >> >> Thanks & Best regards, >> Khue. >> >> ------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> Ken >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> *irc.freenode.net #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/20130408/a8ccdbd0/attachment-0001.html From ashish at nms.co.in Mon Apr 8 12:03:02 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 8 Apr 2013 13:33:02 +0530 Subject: [Freeswitch-users] Default maximum number of simultaneous outgoing PSTN calls In-Reply-To: References: <3E944195-31F2-4239-A9FB-D9F9D5758557@freeswitch.org> Message-ID: ftdm dump on span 1 from channels 5 to 30 shows this output: span_id: 1 chan_id: 25 physical_span_id: 1 physical_chan_id: 25 physical_status: ok physical_status_red: 0 physical_status_yellow: 0 physical_status_rai: 0 physical_status_blue: 0 physical_status_ais: 0 physical_status_general: 0 signaling_status: UP type: B state: SUSPENDED last_state: RESTART txgain: 0.00 rxgain: 0.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NORMAL_UNSPECIFIED session: (none) -- States -- -- Function -- -- Location -- -- Time Offset -- DOWN => RESTART [on_dchan_up] [ftmod_libpri.c:2372] 0ms RESTART => SUSPENDED [on_timeout_t316] [ftmod_libpri.c:1952] 120023ms Time since last state change: 4205443ms On Mon, Apr 8, 2013 at 11:17 AM, Peter Olsson wrote: > You probably need to contact you PSTN provider to help you out here. You > can also dump the FreeTDM traffic and analyze what happens on the > signalling level. > > /Peter > > 8 apr 2013 kl. 06:48 skrev "Ashish gautam" ashish at nms.co.in>>: > > Thanks Brian, > > I am generating not more than 10 calls at a time from FreeSWITCH. There is > a script that reads for any new entries from database and as soon as the > rows are inserted and it generates calls through event socket > simultaneously to all the new numbers. I have also tried to put a sleep of > 250 ms in the script generating request to event socket. Below is the piece > of code that executes whenever a new row is generated: > > my $dialstring = "api originate > {voiceMessageID=$vmID,respreqd=$respReqd,mobnum=$mobileNum,lang=$language,ignore_early_media=true}freetdm/1/A/$mobileNum > 47673501 XML public\n\n"; > # initialize host and port > $socket = IO::Socket::INET->new(PeerAddr => "127.0.0.1", > PeerPort => 8021, > Proto => "tcp", > Type => SOCK_STREAM) > or die "Couldn't connect to 10.1.30.229:8021 : > $@\n"; > print $socket "auth ClueCon\n\n\n"; > $answer = <$socket>; > print $answer; > print $socket $dialstring; > $socket->recv($data,1024); > #$answer = <$socket>; > print $data; > close($socket); > -- > Ashish > > On Mon, Apr 8, 2013 at 7:12 AM, Brian West brian at freeswitch.org>> wrote: > How many calls are you trying to start at once? You could just be pissing > off the switch on the other side by bring up too many calls quickly. > On Apr 6, 2013, at 2:18 AM, Ashish gautam wrote: > > > Hi, > > > > I am making simultaneous outgoing calls to PSTN network through freeTDM > and > > PRI card via Event socket. FS only generates at max. four calls and for > the > > rest it shows NORMAL_CIRCUIT_CONGESTION. > > > > What could be the possible reason for this? Is there any default maximum > > limit on the number of simultaneous outgoing calls through > > freeTDM/libpri/DAHDI stack.? > > > > Kindly throw some light on this. > > > > Regards. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > http://freeswitchcookbook.com > http://freeswitchbook.com > > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official 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:5162473332765878717253! > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official 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://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org > > > !DSPAM:5162473332765878717253! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130408/cb0593e2/attachment.html From khuenm at vega.com.vn Mon Apr 8 12:23:05 2013 From: khuenm at vega.com.vn (Khue Nguyen Minh) Date: Mon, 8 Apr 2013 15:23:05 +0700 Subject: [Freeswitch-users] CDR In-Reply-To: References: Message-ID: I added this line into cdr_csv.conf.xml but, in output file I don't receive dtmf. This is output file "khuenm", "984713985", "301", "" "khuenm", "984713985", "302", "" 2013/4/8 Avi Marcus > Try logging: digits_dialed > > -Avi Marcus > BestFone > > > On Mon, Apr 8, 2013 at 6:47 AM, Khue Nguyen Minh wrote: > >> I was export cdr file success. Thank you very much. >> >> Now, I want write all dtmf into cdr file, how I can do that? Please help >> me. >> >> Brs, >> Khue >> >> >> 2013/4/5 Ken Rice >> >>> You don?t do that in the dialplan, mod_cdr_csv does that for you >>> >>> >>> >>> On 4/5/13 2:24 AM, "Khue Nguyen Minh" wrote: >>> >>> Hi all, >>> >>> I want create cdr (filetype: csv) file in freeswitch. But, I don't known >>> how I can do it from xml dialplan. Please guide me. >>> >>> Thanks & Best regards, >>> Khue. >>> >>> ------------------------------ >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> -- >>> Ken >>> *http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> *irc.freenode.net #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/20130408/0be46a2e/attachment-0001.html From steveayre at gmail.com Mon Apr 8 15:46:51 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 8 Apr 2013 12:46:51 +0100 Subject: [Freeswitch-users] Default maximum number of simultaneous outgoing PSTN calls In-Reply-To: <3E944195-31F2-4239-A9FB-D9F9D5758557@freeswitch.org> References: <3E944195-31F2-4239-A9FB-D9F9D5758557@freeswitch.org> Message-ID: Also what kind of line do you have? PRI/BRI sets a limits on the number of channels. Even though PRI gives 32 the provider might be limiting it to 10. On 8 April 2013 02:42, Brian West wrote: > How many calls are you trying to start at once? You could just be pissing > off the switch on the other side by bring up too many calls quickly. > On Apr 6, 2013, at 2:18 AM, Ashish gautam wrote: > > > Hi, > > > > I am making simultaneous outgoing calls to PSTN network through freeTDM > and > > PRI card via Event socket. FS only generates at max. four calls and for > the > > rest it shows NORMAL_CIRCUIT_CONGESTION. > > > > What could be the possible reason for this? Is there any default maximum > > limit on the number of simultaneous outgoing calls through > > freeTDM/libpri/DAHDI stack.? > > > > Kindly throw some light on this. > > > > Regards. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > http://freeswitchcookbook.com > http://freeswitchbook.com > > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130408/bb3dccb7/attachment.html From ashish at nms.co.in Mon Apr 8 15:57:15 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 8 Apr 2013 17:27:15 +0530 Subject: [Freeswitch-users] Default maximum number of simultaneous outgoing PSTN calls In-Reply-To: References: <3E944195-31F2-4239-A9FB-D9F9D5758557@freeswitch.org> Message-ID: Steve, I have a PRI line and I have also checked with the provider that all the channels are open. Thanks. -- Ashish On Mon, Apr 8, 2013 at 5:16 PM, Steven Ayre wrote: > Also what kind of line do you have? PRI/BRI sets a limits on the number of > channels. Even though PRI gives 32 the provider might be limiting it to 10. > > > On 8 April 2013 02:42, Brian West wrote: > >> How many calls are you trying to start at once? You could just be >> pissing off the switch on the other side by bring up too many calls quickly. >> On Apr 6, 2013, at 2:18 AM, Ashish gautam wrote: >> >> > Hi, >> > >> > I am making simultaneous outgoing calls to PSTN network through freeTDM >> and >> > PRI card via Event socket. FS only generates at max. four calls and for >> the >> > rest it shows NORMAL_CIRCUIT_CONGESTION. >> > >> > What could be the possible reason for this? Is there any default maximum >> > limit on the number of simultaneous outgoing calls through >> > freeTDM/libpri/DAHDI stack.? >> > >> > Kindly throw some light on this. >> > >> > Regards. >> >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX PO BOX 2531 >> Brookfield, WI 53008-2531 >> >> Twitter: @FreeSWITCH_Wire >> http://freeswitchcookbook.com >> http://freeswitchbook.com >> >> T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST >> iNUM: +883 5100 1286 0410 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20130408/1260d0f0/attachment.html From martyn at magiccow.co.uk Mon Apr 8 16:00:40 2013 From: martyn at magiccow.co.uk (Martyn Davies) Date: Mon, 8 Apr 2013 13:00:40 +0100 Subject: [Freeswitch-users] T316 timed out, resending RESTART request In-Reply-To: References: Message-ID: The RESTART is sent out for PRI (and ISDN) when the signalling is first established, or the link to the switch needs to be restarted. If the switch does not respond to RESTART (and the timer T316 expires - IIRC this is a long timeout), then no channels will be available for calls. The first thing to check I think is that your PRI board is definitely set to the right variant of signalling protocol, so that it matches that at the switch/provider. There are many variations of protocol (NI-1, EuroISDN etc), and if there's a mismatch it could be that one end or the other is using RESTART to try to recover the link after some kind of failure. Tell me, do any calls go out at all? Or is it that some calls go out, then you get congestion? Regards, Martyn On 8 April 2013 06:09, Ashish gautam wrote: > Hi, > > I am getting this warning: "2013-03-15 12:25:15.816238 [WARNING] > ftmod_libpri.c:1954 [s1c29][1:29] -- T316 timed out, resending RESTART > request" when I start my FS box. > > After these warnings with three restart attempts I get this error: > > 2013-03-15 12:25:45.816239 [ERR] ftmod_libpri.c:1950 [s1c31][1:31] -- T316 > timed out, channel reached restart attempt limit '3' and is suspended > > I am also facing NORMAL_CIRCUIT_CONGESTION error when generation outgoing > calls through FreeTDM. Is this a bug? > > Kindly help. > > -- > Ashish > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ashish at nms.co.in Mon Apr 8 16:23:01 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 8 Apr 2013 17:53:01 +0530 Subject: [Freeswitch-users] T316 timed out, resending RESTART request In-Reply-To: References: Message-ID: Thanks Martyn for your response. Yes some calls go out ( nearly 4 simultaneously ) and for the rest, it shows congestion. I have checked the signalling protocol its fine. Out of the 30 channels 4 get restarted normally (probably this is the reason why only four calls can go out) and rest are not, they remain in the suspended state. --Ashish On Mon, Apr 8, 2013 at 5:30 PM, Martyn Davies wrote: > The RESTART is sent out for PRI (and ISDN) when the signalling is > first established, or the link to the switch needs to be restarted. > If the switch does not respond to RESTART (and the timer T316 expires > - IIRC this is a long timeout), then no channels will be available for > calls. > > The first thing to check I think is that your PRI board is definitely > set to the right variant of signalling protocol, so that it matches > that at the switch/provider. There are many variations of protocol > (NI-1, EuroISDN etc), and if there's a mismatch it could be that one > end or the other is using RESTART to try to recover the link after > some kind of failure. > > Tell me, do any calls go out at all? Or is it that some calls go out, > then you get congestion? > > Regards, > Martyn > > > On 8 April 2013 06:09, Ashish gautam wrote: > > Hi, > > > > I am getting this warning: "2013-03-15 12:25:15.816238 [WARNING] > > ftmod_libpri.c:1954 [s1c29][1:29] -- T316 timed out, resending RESTART > > request" when I start my FS box. > > > > After these warnings with three restart attempts I get this error: > > > > 2013-03-15 12:25:45.816239 [ERR] ftmod_libpri.c:1950 [s1c31][1:31] -- > T316 > > timed out, channel reached restart attempt limit '3' and is suspended > > > > I am also facing NORMAL_CIRCUIT_CONGESTION error when generation outgoing > > calls through FreeTDM. Is this a bug? > > > > Kindly help. > > > > -- > > Ashish > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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/20130408/0ab658f2/attachment-0001.html From jeff at jefflenk.com Mon Apr 8 16:41:33 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 8 Apr 2013 05:41:33 -0700 (PDT) Subject: [Freeswitch-users] Build errors in 2010.express solution In-Reply-To: <1365285612276-7589476.post@n2.nabble.com> References: <1365285612276-7589476.post@n2.nabble.com> Message-ID: <1365424893457-7589513.post@n2.nabble.com> If you can reproduce this from a fresh clone please open a Jira for this. There should not have been any specific changes required for express that I can think of. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Build-errors-in-2010-express-solution-tp7589476p7589513.html Sent from the freeswitch-users mailing list archive at Nabble.com. From rnbrady at gmail.com Mon Apr 8 18:45:34 2013 From: rnbrady at gmail.com (Richard Brady) Date: Mon, 8 Apr 2013 15:45:34 +0100 Subject: [Freeswitch-users] [Solved] adding In-Reply-To in sip header not working In-Reply-To: <3B6503C1-BA0A-46BD-B48F-DEA3F2AD5FBB@mgtech.com> References: <0F4F8E19-73FC-460F-A5AB-B7A2CF8CD78B@mgtech.com> <1769A24D-BC79-4CFD-AA88-0A1E81E346BA@mgtech.com> <3B6503C1-BA0A-46BD-B48F-DEA3F2AD5FBB@mgtech.com> Message-ID: Hey Mario Thanks for sharing the solution. > notice it was close to Richards idea but uses _h_ instead of _rh_ I was definitely recommending _h_ not _rh_ but glad you got there in the end! Richard On 18 February 2013 18:05, Mario G wrote: > > Finally! Sharing with anyone else wanting to pass the calling party ID from an external SIP account into FS then back to the standard PSTN on some ITSPs (Callcentric in this case), this is what works: > > Anytime before the bridge: (notice it was close to Richards idea but uses _h_ instead of _rh_) > > > In my bridge below remove the effective caller id number like this: > > > > From peter at olssononline.se Mon Apr 8 15:54:14 2013 From: peter at olssononline.se (Peter Olsson) Date: Mon, 8 Apr 2013 13:54:14 +0200 Subject: [Freeswitch-users] Testmail - I've changed my email address Message-ID: Just confirming that the list works from my new email. /Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130408/eb3b81ed/attachment.html From krice at freeswitch.org Mon Apr 8 19:10:44 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 08 Apr 2013 10:10:44 -0500 Subject: [Freeswitch-users] Testmail - I've changed my email address In-Reply-To: Message-ID: It works On 4/8/13 6:54 AM, "Peter Olsson" wrote: > Just confirming that the list works from my new email. > > /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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130408/de64cf7f/attachment.html From andrew at cassidywebservices.co.uk Mon Apr 8 19:11:41 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 8 Apr 2013 16:11:41 +0100 Subject: [Freeswitch-users] Testmail - I've changed my email address In-Reply-To: References: Message-ID: No, it doesn't. :P On 8 April 2013 12:54, Peter Olsson wrote: > Just confirming that the list works from my new email. > > /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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130408/fc733397/attachment.html From mike at jerris.com Mon Apr 8 19:16:14 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 8 Apr 2013 11:16:14 -0400 Subject: [Freeswitch-users] Testmail - I've changed my email address In-Reply-To: References: Message-ID: Nope. On Apr 8, 2013, at 7:54 AM, Peter Olsson wrote: > Just confirming that the list works from my new email. > > /Peter > _________________________________________________________________________ From peter at olssononline.se Mon Apr 8 19:41:02 2013 From: peter at olssononline.se (Peter Olsson) Date: Mon, 8 Apr 2013 17:41:02 +0200 Subject: [Freeswitch-users] Testmail - I've changed my email address In-Reply-To: References: Message-ID: <-8543436962080882999@unknownmsgid> That's too bad :) /Peter 8 apr 2013 kl. 17:19 skrev Michael Jerris : > Nope. > > On Apr 8, 2013, at 7:54 AM, Peter Olsson wrote: > >> Just confirming that the list works from my new email. >> >> /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 From schoch+freeswitch.org at xwin32.com Mon Apr 8 21:13:52 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 8 Apr 2013 10:13:52 -0700 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: <20130119191943.ca09ab6d@mail.tritonwest.net> <5102AF93.2000201@coppice.org> Message-ID: On Sun, Apr 7, 2013 at 6:41 PM, Brian West wrote: > Not too sure you need t38_gateway on this call flow. I was under the impression that the /dev/FS FAX pseudo-modems did not output T.38, but just a regular audio (G.711) stream. Do I not understand that correctly? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130408/4cee42ec/attachment-0001.html From jason.holden at start.ca Mon Apr 8 21:29:11 2013 From: jason.holden at start.ca (Jason Holden) Date: Mon, 8 Apr 2013 13:29:11 -0400 Subject: [Freeswitch-users] remove half second of moh when transfering to vm if phone is not answered Message-ID: <1B52F0571E944AEA98DB5E2046D29263@bob> Any suggestions on this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130408/0510d398/attachment.html From ebrahim.bararian at gmail.com Mon Apr 8 21:34:39 2013 From: ebrahim.bararian at gmail.com (Ebrahim Bararian) Date: Mon, 8 Apr 2013 22:04:39 +0430 Subject: [Freeswitch-users] Connecting Internal IP phones to PSTN Message-ID: Hi, I have some soft IP phones in an internal local network. I want to connect these IP phones to PSTN via 2 PSTN lines. How many ways are there to solve this problem? Which hardwares should I buy? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130408/2f0970c3/attachment.html From ebrahim.bararian at gmail.com Mon Apr 8 21:38:42 2013 From: ebrahim.bararian at gmail.com (Ebrahim Bararian) Date: Mon, 8 Apr 2013 22:08:42 +0430 Subject: [Freeswitch-users] Problem With Running Mod_pocketsphinx In-Reply-To: References: Message-ID: Thank you so much. I didn't know that before. I know how to set mod_portaudio to retrieve voice in 8k or 16k. But how can I set the mod_pocketsphinx to use 8k acoustical model? On Mon, Apr 8, 2013 at 6:13 AM, Brian West wrote: > The 16k acoustical model isn't as robust as the 8k version. Just an FYI. > > On Apr 6, 2013, at 7:17 PM, Ebrahim Bararian wrote: > > > I found it. It was in the file portaudio.conf.xml. I changed the sample > > rate to 16000. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > http://freeswitchcookbook.com > http://freeswitchbook.com > > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130408/13fb7471/attachment.html From hajime9ma at gmail.com Mon Apr 8 21:27:59 2013 From: hajime9ma at gmail.com (hajime) Date: Mon, 8 Apr 2013 10:27:59 -0700 (PDT) Subject: [Freeswitch-users] Presence in ODBC/PgSQL? In-Reply-To: References: <1365279647653-7589475.post@n2.nabble.com> Message-ID: <1365442079710-7589521.post@n2.nabble.com> sip_subscriptions or sip_presence tables - am I mistaken, or are they only maintained in the sqlite db? I have the XML configuration param "core-db-dsn" set to my local PostgreSQL db but only see tables such as aliases, calls, channels, complete, interfaces, nat, recovery, registrations, and tasks. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Presence-in-ODBC-PgSQL-tp7589475p7589521.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ira at connectmevoice.com Tue Apr 9 01:02:15 2013 From: ira at connectmevoice.com (Ira Tessler) Date: Mon, 8 Apr 2013 17:02:15 -0400 Subject: [Freeswitch-users] Presence Sanity Check In-Reply-To: References: Message-ID: Thank you for your information. It did help! Ira Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com On Sun, Apr 7, 2013 at 12:49 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > In regards to the UDP fragmentation, this is an extremely good question. > > Only yesterday I started to build a simple forwarding SBC in Python using > UDP sockets, however I came up against the same theoretical problem of > packets being larger than 1500 bytes. > > I've had a read through various documentation; > http://www.rfc-ref.org/RFC-TEXTS/3261/chapter18.html > http://www.ietf.org/rfc/rfc3428.txt > > https://lists.cs.columbia.edu/pipermail/sip-implementors/2006-August/013857.html > > http://lists.freeswitch.org/pipermail/freeswitch-users/2011-February/068372.html > > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05912.html > > Anthony has stated the following; > *"the only reliable answer is use TCP."* > > There was also the option of enabling compact headers; > > http://lists.freeswitch.org/pipermail/freeswitch-users/2011-December/078633.html > > From what I can tell, there is no way to guarantee this problem won't > happen unless you use TCP. You could reduce the packet size by compacting > headers or removing codecs, but this would be on the assumption that every > hop is running at 1500 MTU. > > Hope this helps! > > Cal > > On Sat, Apr 6, 2013 at 2:28 PM, Ira Tessler wrote: > >> I just need a little guidance with the way presence works. Forgive me if >> I am asking novice questions. >> >> Background (simple version) >> We run Freeswitch in a hosted/cloud environment in a data center. We have >> IP phones in our office on our LAN. >> >> That way I am understanding how Presence works, I am just learning this, >> is that when a BLF button is programmed on a phone, that phone will send a >> "Subscribe" message to Freeswitch. The subscriptions are stored in the >> sip_subscriptions table (i think) in the sofia database for the sip >> profile. When calls come in for that subscription, Freeswitch will send out >> a NOTIFY message to the phone that subscribed in order to change the state >> of the BLF Light. >> >> He is my questions/issue/confusion. >> All our phones use UDP which has a maximum packet size of 1500 bytes. >> When doing a sofia global siptrace on, I notice that most of the NOTIFY >> messages are greater then 1500 bytes. That will cause packet fragmentation. >> So if the NOTIFY message is fragmented, will it get to the phone correctly? >> (all the time, some of the time, never??) >> >> If the the answer is other then ("all the time"), how do I fix this? The >> only solution I can come up with is having my phones use TCP instead of >> UDP. Is that the correctly solution? Did anyone else out there run into >> this issue and if so, what is the "best practice" solution (if there is >> one)? >> >> Thank you in advance! >> >> Ira Tessler >> Lead Software Engineer >> ConnectMe >> (732) 490-9007 x2 >> ira at connectmevoice.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/20130408/7739549c/attachment.html From anthony.minessale at gmail.com Tue Apr 9 01:17:31 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 8 Apr 2013 16:17:31 -0500 Subject: [Freeswitch-users] Presence Sanity Check In-Reply-To: References: Message-ID: Its actually somewhat ridiculous and one of my personal favorites in terms of RFC smoke and mirrors in SIP to try and cover up a flaw with more specs. They basically say: If the total packet including the sip headers and the payload exceeds the MTU and you are using the UDP transport, you MUST try sending the packet over TCP instead. If that times out or fails then you SHOULD send it over UDP anyway. THEREFORE by virtue of this decree: You MUST implement your sip stack to accept UDP packets of up to 65536 bytes. AND You MUST implement both TCP and UDP transports. So in short, you are not supposed to send anything over udp that exceeds the mtu yet you are required to implement it so its possible. Many stacks, including Asterisk for many of the first half of FS existence, did not implement TCP so with this rule being enforced, the packets would sit there for 2-5 min then give up and change to UDP. How's that for PDD. Anyway, we choose to ignore this rule intentionally and just stick with the negotiated protocol. If you find yourself in this situation the solution is to use TCP. P.S. If they did not use 2k of XML to transmit about 12 bytes worth of useful info regarding the state of the presence, we would not have this problem to begin with ;) On Mon, Apr 8, 2013 at 4:02 PM, Ira Tessler wrote: > Thank you for your information. It did help! > > Ira > > Ira Tessler > Lead Software Engineer > ConnectMe > (732) 490-9007 x2 > ira at connectmevoice.com > > > On Sun, Apr 7, 2013 at 12:49 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> In regards to the UDP fragmentation, this is an extremely good question. >> >> Only yesterday I started to build a simple forwarding SBC in Python using >> UDP sockets, however I came up against the same theoretical problem of >> packets being larger than 1500 bytes. >> >> I've had a read through various documentation; >> http://www.rfc-ref.org/RFC-TEXTS/3261/chapter18.html >> http://www.ietf.org/rfc/rfc3428.txt >> >> https://lists.cs.columbia.edu/pipermail/sip-implementors/2006-August/013857.html >> >> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-February/068372.html >> >> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05912.html >> >> Anthony has stated the following; >> *"the only reliable answer is use TCP."* >> >> There was also the option of enabling compact headers; >> >> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-December/078633.html >> >> From what I can tell, there is no way to guarantee this problem won't >> happen unless you use TCP. You could reduce the packet size by compacting >> headers or removing codecs, but this would be on the assumption that every >> hop is running at 1500 MTU. >> >> Hope this helps! >> >> Cal >> >> On Sat, Apr 6, 2013 at 2:28 PM, Ira Tessler wrote: >> >>> I just need a little guidance with the way presence works. Forgive me if >>> I am asking novice questions. >>> >>> Background (simple version) >>> We run Freeswitch in a hosted/cloud environment in a data center. We >>> have IP phones in our office on our LAN. >>> >>> That way I am understanding how Presence works, I am just learning this, >>> is that when a BLF button is programmed on a phone, that phone will send a >>> "Subscribe" message to Freeswitch. The subscriptions are stored in the >>> sip_subscriptions table (i think) in the sofia database for the sip >>> profile. When calls come in for that subscription, Freeswitch will send out >>> a NOTIFY message to the phone that subscribed in order to change the state >>> of the BLF Light. >>> >>> He is my questions/issue/confusion. >>> All our phones use UDP which has a maximum packet size of 1500 bytes. >>> When doing a sofia global siptrace on, I notice that most of the NOTIFY >>> messages are greater then 1500 bytes. That will cause packet fragmentation. >>> So if the NOTIFY message is fragmented, will it get to the phone correctly? >>> (all the time, some of the time, never??) >>> >>> If the the answer is other then ("all the time"), how do I fix this? The >>> only solution I can come up with is having my phones use TCP instead of >>> UDP. Is that the correctly solution? Did anyone else out there run into >>> this issue and if so, what is the "best practice" solution (if there is >>> one)? >>> >>> Thank you in advance! >>> >>> Ira Tessler >>> Lead Software Engineer >>> ConnectMe >>> (732) 490-9007 x2 >>> ira at connectmevoice.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130408/29e437e0/attachment-0001.html From msc at freeswitch.org Tue Apr 9 01:33:12 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 8 Apr 2013 14:33:12 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: A belated happy Monday to all! Last week on our conference callwe spent some time discussing various topics such as Jira system improvements and updates, sound prompts, ClueCon 2013 and several other topics. The recordings can be found in the usual place. On this week's call we are going to have a community discussion and a few minutes of Q&A from the audience. If possible, please add your questions to the agenda page and we'll research them prior to the call. Where applicable we'll ask members of the audience to update the wiki to reflect any undocumented knowledge that has been discussed. Things have been busy with the advent of ClueCon 2013season but we're on top of things and we'll keep everyone posted on all the particulars. Feel free to register at any time. Also, please contact us at this email address if you have any questions about being a speaker, sponsor, or attendee. We'll be glad to assist. One last item: I wanted to personally say thank you to Steven Ayre for all of his hard work with answering questions on the mailing list. He has done a great job of helping lots of people with a variety of questions. Many thanks to Steven and all the others who make FreeSWITCH such a great FOSS community. Thanks and have a great week! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130408/8c234c93/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue Apr 9 01:50:44 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 8 Apr 2013 22:50:44 +0100 Subject: [Freeswitch-users] Presence Sanity Check In-Reply-To: References: Message-ID: Hear hear! On Mon, Apr 8, 2013 at 10:17 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Its actually somewhat ridiculous and one of my personal favorites in terms > of RFC smoke and mirrors in SIP to try and cover up a flaw with more specs. > They basically say: > > > If the total packet including the sip headers and the payload exceeds the > MTU and you are using the UDP transport, you MUST try sending the packet > over TCP instead. If that times out or fails then you SHOULD send it over > UDP anyway. > > > THEREFORE by virtue of this decree: > > You MUST implement your sip stack to accept UDP packets of up to 65536 > bytes. > > AND > > You MUST implement both TCP and UDP transports. > > > So in short, you are not supposed to send anything over udp that exceeds > the mtu yet you are required to implement it so its possible. > > Many stacks, including Asterisk for many of the first half of FS > existence, did not implement TCP so with this rule being enforced, the > packets would sit there for 2-5 min then give up and change to UDP. How's > that for PDD. > > Anyway, we choose to ignore this rule intentionally and just stick with > the negotiated protocol. If you find yourself in this situation the > solution is to use TCP. > > > P.S. > > If they did not use 2k of XML to transmit about 12 bytes worth of useful > info regarding the state of the presence, we would not have this problem to > begin with ;) > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Mon, Apr 8, 2013 at 4:02 PM, Ira Tessler wrote: > >> Thank you for your information. It did help! >> >> Ira >> >> Ira Tessler >> Lead Software Engineer >> ConnectMe >> (732) 490-9007 x2 >> ira at connectmevoice.com >> >> >> On Sun, Apr 7, 2013 at 12:49 PM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> In regards to the UDP fragmentation, this is an extremely good question. >>> >>> Only yesterday I started to build a simple forwarding SBC in Python >>> using UDP sockets, however I came up against the same theoretical problem >>> of packets being larger than 1500 bytes. >>> >>> I've had a read through various documentation; >>> http://www.rfc-ref.org/RFC-TEXTS/3261/chapter18.html >>> http://www.ietf.org/rfc/rfc3428.txt >>> >>> https://lists.cs.columbia.edu/pipermail/sip-implementors/2006-August/013857.html >>> >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-February/068372.html >>> >>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05912.html >>> >>> Anthony has stated the following; >>> *"the only reliable answer is use TCP."* >>> >>> There was also the option of enabling compact headers; >>> >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-December/078633.html >>> >>> From what I can tell, there is no way to guarantee this problem won't >>> happen unless you use TCP. You could reduce the packet size by compacting >>> headers or removing codecs, but this would be on the assumption that every >>> hop is running at 1500 MTU. >>> >>> Hope this helps! >>> >>> Cal >>> >>> On Sat, Apr 6, 2013 at 2:28 PM, Ira Tessler wrote: >>> >>>> I just need a little guidance with the way presence works. Forgive me >>>> if I am asking novice questions. >>>> >>>> Background (simple version) >>>> We run Freeswitch in a hosted/cloud environment in a data center. We >>>> have IP phones in our office on our LAN. >>>> >>>> That way I am understanding how Presence works, I am just learning >>>> this, is that when a BLF button is programmed on a phone, that phone will >>>> send a "Subscribe" message to Freeswitch. The subscriptions are stored in >>>> the sip_subscriptions table (i think) in the sofia database for the sip >>>> profile. When calls come in for that subscription, Freeswitch will send out >>>> a NOTIFY message to the phone that subscribed in order to change the state >>>> of the BLF Light. >>>> >>>> He is my questions/issue/confusion. >>>> All our phones use UDP which has a maximum packet size of 1500 bytes. >>>> When doing a sofia global siptrace on, I notice that most of the NOTIFY >>>> messages are greater then 1500 bytes. That will cause packet fragmentation. >>>> So if the NOTIFY message is fragmented, will it get to the phone correctly? >>>> (all the time, some of the time, never??) >>>> >>>> If the the answer is other then ("all the time"), how do I fix this? >>>> The only solution I can come up with is having my phones use TCP instead of >>>> UDP. Is that the correctly solution? Did anyone else out there run into >>>> this issue and if so, what is the "best practice" solution (if there is >>>> one)? >>>> >>>> Thank you in advance! >>>> >>>> Ira Tessler >>>> Lead Software Engineer >>>> ConnectMe >>>> (732) 490-9007 x2 >>>> ira at connectmevoice.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at 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/20130408/a66bc6cd/attachment-0001.html From steveayre at gmail.com Tue Apr 9 02:45:55 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 8 Apr 2013 23:45:55 +0100 Subject: [Freeswitch-users] Default maximum number of simultaneous outgoing PSTN calls In-Reply-To: References: <3E944195-31F2-4239-A9FB-D9F9D5758557@freeswitch.org> Message-ID: Compare CDRs with the provider to see whether they see the calls. If they do the problem isn't at your end. They could be generating that cause for a number of reasons, or it could be coming from even further upstream. Steve On 8 Apr 2013, at 12:57, Ashish gautam wrote: > Steve, > > I have a PRI line and I have also checked with the provider that all the channels are open. > > Thanks. > > -- Ashish > > On Mon, Apr 8, 2013 at 5:16 PM, Steven Ayre wrote: >> Also what kind of line do you have? PRI/BRI sets a limits on the number of channels. Even though PRI gives 32 the provider might be limiting it to 10. >> >> >> On 8 April 2013 02:42, Brian West wrote: >>> How many calls are you trying to start at once? You could just be pissing off the switch on the other side by bring up too many calls quickly. >>> On Apr 6, 2013, at 2:18 AM, Ashish gautam wrote: >>> >>> > Hi, >>> > >>> > I am making simultaneous outgoing calls to PSTN network through freeTDM and >>> > PRI card via Event socket. FS only generates at max. four calls and for the >>> > rest it shows NORMAL_CIRCUIT_CONGESTION. >>> > >>> > What could be the possible reason for this? Is there any default maximum >>> > limit on the number of simultaneous outgoing calls through >>> > freeTDM/libpri/DAHDI stack.? >>> > >>> > Kindly throw some light on this. >>> > >>> > Regards. >>> >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> >>> Twitter: @FreeSWITCH_Wire >>> http://freeswitchcookbook.com >>> http://freeswitchbook.com >>> >>> T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST >>> iNUM: +883 5100 1286 0410 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20130408/8908c5c2/attachment.html From anthony.minessale at gmail.com Tue Apr 9 02:51:02 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 8 Apr 2013 17:51:02 -0500 Subject: [Freeswitch-users] remove half second of moh when transfering to vm if phone is not answered In-Reply-To: <1B52F0571E944AEA98DB5E2046D29263@bob> References: <1B52F0571E944AEA98DB5E2046D29263@bob> Message-ID: Needs a bit more verbose description and some usage context. On Mon, Apr 8, 2013 at 12:29 PM, Jason Holden wrote: > Any suggestions on this?**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130408/db04647f/attachment.html From anthony.minessale at gmail.com Tue Apr 9 02:53:59 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 8 Apr 2013 17:53:59 -0500 Subject: [Freeswitch-users] Presence in ODBC/PgSQL? In-Reply-To: <1365442079710-7589521.post@n2.nabble.com> References: <1365279647653-7589475.post@n2.nabble.com> <1365442079710-7589521.post@n2.nabble.com> Message-ID: Sofia profiles also have their own dsn params for connection to RDBMS systems. On Mon, Apr 8, 2013 at 12:27 PM, hajime wrote: > sip_subscriptions or sip_presence tables - am I mistaken, or are they only > maintained in the sqlite db? > I have the XML configuration param "core-db-dsn" set to my local > PostgreSQL > db but only see tables such as aliases, calls, channels, complete, > interfaces, nat, recovery, registrations, and tasks. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Presence-in-ODBC-PgSQL-tp7589475p7589521.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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130408/84cea9ab/attachment.html From chang33.tw at gmail.com Tue Apr 9 08:55:32 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Tue, 09 Apr 2013 12:55:32 +0800 Subject: [Freeswitch-users] get agent status Message-ID: <51639F44.7060804@gmail.com> Hi, I try to maintain the status(state) of the FS agents from these two commands. list_users callcenter_config agent list I'm confused with their results. says the agent is error/user_not_registered. says this agent is Available|Waiting. Does this normal? Should I change some other commands? Thanks. Jimmy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/3fe49d41/attachment-0001.html From gerald.weber at besharp.at Tue Apr 9 09:57:34 2013 From: gerald.weber at besharp.at (Gerald Weber) Date: Tue, 9 Apr 2013 05:57:34 +0000 Subject: [Freeswitch-users] get agent status In-Reply-To: <51639F44.7060804@gmail.com> References: <51639F44.7060804@gmail.com> Message-ID: Hi, those 2 commands have nothing to do with each other. list_users goes through the directory listing each configured user that can be used to register a sip device. error/user_not_registered means that no sip device used the username to register with. callcenter_config agent list lists the ?agents? defined in mod_callcenter. They can be Available/Waiting even when their contact (the user/xxx thing) is not registered. mod_callcenter agent != user Hope this helps?or maybe i got your question wrong ? Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jimmy Chang Gesendet: Dienstag, 09. April 2013 06:56 An: freeswitch-users at lists.freeswitch.org Betreff: [Freeswitch-users] get agent status Hi, I try to maintain the status(state) of the FS agents from these two commands. list_users callcenter_config agent list I'm confused with their results. says the agent is error/user_not_registered. says this agent is Available|Waiting. Does this normal? Should I change some other commands? Thanks. Jimmy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/41b87fc0/attachment.html From chang33.tw at gmail.com Tue Apr 9 10:39:38 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Tue, 09 Apr 2013 14:39:38 +0800 Subject: [Freeswitch-users] get agent status In-Reply-To: References: <51639F44.7060804@gmail.com> Message-ID: <5163B7AA.2030601@gmail.com> Hi, That's what I want to know. Thanks. If I want to get lastest state of agent, any suggestions? Thanks in advance. Jimmy ? 2013/4/9 ?? 01:57, Gerald Weber ??: > > Hi, > > those 2 commands have nothing to do with each other. > > list_users goes through the directory listing each configured user > that can be used to register a sip device. > > error/user_not_registered means that no sip device used the username > to register with. > > callcenter_config agent list lists the "agents" defined in mod_callcenter. > > They can be Available/Waiting even when their contact (the user/xxx > thing) is not registered. > > mod_callcenter agent != user > > Hope this helps...or maybe i got your question wrong ? > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von > *Jimmy Chang > *Gesendet:* Dienstag, 09. April 2013 06:56 > *An:* freeswitch-users at lists.freeswitch.org > *Betreff:* [Freeswitch-users] get agent status > > Hi, > > I try to maintain the status(state) of the FS agents from these two > commands. > list_users > callcenter_config agent list > > I'm confused with their results. > says the agent is error/user_not_registered. > says this agent is Available|Waiting. > > Does this normal? > Should I change some other commands? > > Thanks. > Jimmy > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/45969602/attachment.html From gerald.weber at besharp.at Tue Apr 9 10:48:13 2013 From: gerald.weber at besharp.at (Gerald Weber) Date: Tue, 9 Apr 2013 06:48:13 +0000 Subject: [Freeswitch-users] get agent status In-Reply-To: <5163B7AA.2030601@gmail.com> References: <51639F44.7060804@gmail.com> <5163B7AA.2030601@gmail.com> Message-ID: Either use callcenter_config agent get state xxx for a specific agent or callcenter_config agent list for all agents Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jimmy Chang Gesendet: Dienstag, 09. April 2013 08:40 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] get agent status Hi, That's what I want to know. Thanks. If I want to get lastest state of agent, any suggestions? Thanks in advance. Jimmy ? 2013/4/9 ?? 01:57, Gerald Weber ??: Hi, those 2 commands have nothing to do with each other. list_users goes through the directory listing each configured user that can be used to register a sip device. error/user_not_registered means that no sip device used the username to register with. callcenter_config agent list lists the ?agents? defined in mod_callcenter. They can be Available/Waiting even when their contact (the user/xxx thing) is not registered. mod_callcenter agent != user Hope this helps?or maybe i got your question wrong ? Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jimmy Chang Gesendet: Dienstag, 09. April 2013 06:56 An: freeswitch-users at lists.freeswitch.org Betreff: [Freeswitch-users] get agent status Hi, I try to maintain the status(state) of the FS agents from these two commands. list_users callcenter_config agent list I'm confused with their results. says the agent is error/user_not_registered. says this agent is Available|Waiting. Does this normal? Should I change some other commands? Thanks. Jimmy _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/dce72221/attachment-0001.html From ashish at nms.co.in Tue Apr 9 11:00:32 2013 From: ashish at nms.co.in (Ashish gautam) Date: Tue, 9 Apr 2013 12:30:32 +0530 Subject: [Freeswitch-users] Default maximum number of simultaneous outgoing PSTN calls In-Reply-To: References: <3E944195-31F2-4239-A9FB-D9F9D5758557@freeswitch.org> Message-ID: I am currently on FS version 1.3. Should I switch back to 1.2, will it help? Also the Dahdi version is 2.4 and the libpri version is 1.4.12. Are all the versions I am using fine? On Tue, Apr 9, 2013 at 4:15 AM, Steven Ayre wrote: > Compare CDRs with the provider to see whether they see the calls. If they > do the problem isn't at your end. They could be generating that cause for a > number of reasons, or it could be coming from even further upstream. > > Steve > > > > On 8 Apr 2013, at 12:57, Ashish gautam wrote: > > Steve, > > I have a PRI line and I have also checked with the provider that all the > channels are open. > > Thanks. > > -- Ashish > > On Mon, Apr 8, 2013 at 5:16 PM, Steven Ayre wrote: > >> Also what kind of line do you have? PRI/BRI sets a limits on the number >> of channels. Even though PRI gives 32 the provider might be limiting it to >> 10. >> >> >> On 8 April 2013 02:42, Brian West wrote: >> >>> How many calls are you trying to start at once? You could just be >>> pissing off the switch on the other side by bring up too many calls quickly. >>> On Apr 6, 2013, at 2:18 AM, Ashish gautam wrote: >>> >>> > Hi, >>> > >>> > I am making simultaneous outgoing calls to PSTN network through >>> freeTDM and >>> > PRI card via Event socket. FS only generates at max. four calls and >>> for the >>> > rest it shows NORMAL_CIRCUIT_CONGESTION. >>> > >>> > What could be the possible reason for this? Is there any default >>> maximum >>> > limit on the number of simultaneous outgoing calls through >>> > freeTDM/libpri/DAHDI stack.? >>> > >>> > Kindly throw some light on this. >>> > >>> > Regards. >>> >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> >>> Twitter: @FreeSWITCH_Wire >>> http://freeswitchcookbook.com >>> http://freeswitchbook.com >>> >>> T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST >>> iNUM: +883 5100 1286 0410 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20130409/f0a883b5/attachment.html From chang33.tw at gmail.com Tue Apr 9 11:39:40 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Tue, 09 Apr 2013 15:39:40 +0800 Subject: [Freeswitch-users] get agent status In-Reply-To: References: <51639F44.7060804@gmail.com> <5163B7AA.2030601@gmail.com> Message-ID: <5163C5BC.4080606@gmail.com> Hi, callcenter_config agent list lists the ?agents" defined in mod_callcenter. They can be Available/Waiting even when their contact (the user/xxx thing) is not registered. In this situation, if I use callcenter_config command, I don't really know how many agents registered. If one agent doesn't register, he or she should be unavailable in our monitor screen. Any suggestions? Jimmy ? 2013/4/9 ?? 02:48, Gerald Weber ??: > > Either use > > callcenter_config agent get state xxx for a specific agent > > or > > callcenter_config agent list for all agents > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von > *Jimmy Chang > *Gesendet:* Dienstag, 09. April 2013 08:40 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] get agent status > > Hi, > > That's what I want to know. > Thanks. > > If I want to get lastest state of agent, any suggestions? > Thanks in advance. > > Jimmy > > ? 2013/4/9 ?? 01:57, Gerald Weber ??: > > Hi, > > those 2 commands have nothing to do with each other. > > list_users goes through the directory listing each configured user > that can be used to register a sip device. > > error/user_not_registered means that no sip device used the > username to register with. > > callcenter_config agent list lists the "agents" defined in > mod_callcenter. > > They can be Available/Waiting even when their contact (the > user/xxx thing) is not registered. > > mod_callcenter agent != user > > Hope this helps...or maybe i got your question wrong ? > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag > von *Jimmy Chang > *Gesendet:* Dienstag, 09. April 2013 06:56 > *An:* freeswitch-users at lists.freeswitch.org > > *Betreff:* [Freeswitch-users] get agent status > > Hi, > > I try to maintain the status(state) of the FS agents from these > two commands. > list_users > callcenter_config agent list > > I'm confused with their results. > says the agent is error/user_not_registered. > says this agent is Available|Waiting. > > Does this normal? > Should I change some other commands? > > Thanks. > Jimmy > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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/20130409/771a5b56/attachment-0001.html From gerald.weber at besharp.at Tue Apr 9 11:52:58 2013 From: gerald.weber at besharp.at (Gerald Weber) Date: Tue, 9 Apr 2013 07:52:58 +0000 Subject: [Freeswitch-users] get agent status In-Reply-To: <5163C5BC.4080606@gmail.com> References: <51639F44.7060804@gmail.com> <5163B7AA.2030601@gmail.com> <5163C5BC.4080606@gmail.com> Message-ID: Then i guess you have to call sofia_contact in your app for each of you agents which either returns the sip contact (sofia/internal/sip:2001 at 192.168.20.150:53372) or error/user_not_registered e.g.: freeswitch at fstest> sofia_contact [call_timeout=10]user/2001 sofia/internal/sip:2001 at 192.168.20.150:53372 freeswitch at fstest> another solution ist o patch mod_callcenter.c to support this in the agent list command Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jimmy Chang Gesendet: Dienstag, 09. April 2013 09:40 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] get agent status Hi, callcenter_config agent list lists the ?agents? defined in mod_callcenter. They can be Available/Waiting even when their contact (the user/xxx thing) is not registered. In this situation, if I use callcenter_config command, I don't really know how many agents registered. If one agent doesn't register, he or she should be unavailable in our monitor screen. Any suggestions? Jimmy ? 2013/4/9 ?? 02:48, Gerald Weber ??: Either use callcenter_config agent get state xxx for a specific agent or callcenter_config agent list for all agents Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jimmy Chang Gesendet: Dienstag, 09. April 2013 08:40 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] get agent status Hi, That's what I want to know. Thanks. If I want to get lastest state of agent, any suggestions? Thanks in advance. Jimmy ? 2013/4/9 ?? 01:57, Gerald Weber ??: Hi, those 2 commands have nothing to do with each other. list_users goes through the directory listing each configured user that can be used to register a sip device. error/user_not_registered means that no sip device used the username to register with. callcenter_config agent list lists the ?agents? defined in mod_callcenter. They can be Available/Waiting even when their contact (the user/xxx thing) is not registered. mod_callcenter agent != user Hope this helps?or maybe i got your question wrong ? Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jimmy Chang Gesendet: Dienstag, 09. April 2013 06:56 An: freeswitch-users at lists.freeswitch.org Betreff: [Freeswitch-users] get agent status Hi, I try to maintain the status(state) of the FS agents from these two commands. list_users callcenter_config agent list I'm confused with their results. says the agent is error/user_not_registered. says this agent is Available|Waiting. Does this normal? Should I change some other commands? Thanks. Jimmy _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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/20130409/f0665b2e/attachment.html From chang33.tw at gmail.com Tue Apr 9 12:21:39 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Tue, 09 Apr 2013 16:21:39 +0800 Subject: [Freeswitch-users] get agent status In-Reply-To: References: <51639F44.7060804@gmail.com> <5163B7AA.2030601@gmail.com> <5163C5BC.4080606@gmail.com> Message-ID: <5163CF93.6020701@gmail.com> Ok, I'll try it. Thanks. ? 2013/4/9 ?? 03:52, Gerald Weber ??: > > Then i guess you have to call > > sofia_contact > > in your app for each of you agents > > which either returns the sip contact > (sofia/internal/sip:2001 at 192.168.20.150:53372) > > or error/user_not_registered > > e.g.: > > freeswitch at fstest> sofia_contact [call_timeout=10]user/2001 > > sofia/internal/sip:2001 at 192.168.20.150:53372 > > freeswitch at fstest> > > another solution ist o patch mod_callcenter.c to support this in the > agent list command > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von > *Jimmy Chang > *Gesendet:* Dienstag, 09. April 2013 09:40 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] get agent status > > Hi, > > > callcenter_config agent list lists the ?agents" defined in mod_callcenter. > They can be Available/Waiting even when their contact (the user/xxx > thing) is not registered. > > > In this situation, if I use callcenter_config command, I don't really > know how many agents registered. > If one agent doesn't register, he or she should be unavailable in our > monitor screen. > > Any suggestions? > > Jimmy > > > ? 2013/4/9 ?? 02:48, Gerald Weber ??: > > Either use > > callcenter_config agent get state xxx for a specific agent > > or > > callcenter_config agent list for all agents > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag > von *Jimmy Chang > *Gesendet:* Dienstag, 09. April 2013 08:40 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] get agent status > > Hi, > > That's what I want to know. > Thanks. > > If I want to get lastest state of agent, any suggestions? > Thanks in advance. > > Jimmy > > ? 2013/4/9 ?? 01:57, Gerald Weber ??: > > Hi, > > those 2 commands have nothing to do with each other. > > list_users goes through the directory listing each configured > user that can be used to register a sip device. > > error/user_not_registered means that no sip device used the > username to register with. > > callcenter_config agent list lists the "agents" defined in > mod_callcenter. > > They can be Available/Waiting even when their contact (the > user/xxx thing) is not registered. > > mod_callcenter agent != user > > Hope this helps...or maybe i got your question wrong ? > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im > Auftrag von *Jimmy Chang > *Gesendet:* Dienstag, 09. April 2013 06:56 > *An:* freeswitch-users at lists.freeswitch.org > > *Betreff:* [Freeswitch-users] get agent status > > Hi, > > I try to maintain the status(state) of the FS agents from > these two commands. > list_users > callcenter_config agent list > > I'm confused with their results. > says the agent is error/user_not_registered. > says this agent is > Available|Waiting. > > Does this normal? > Should I change some other commands? > > Thanks. > Jimmy > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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/20130409/40db3453/attachment-0001.html From chrisbware at yahoo.it Tue Apr 9 13:38:12 2013 From: chrisbware at yahoo.it (Chris B. Ware) Date: Tue, 9 Apr 2013 10:38:12 +0100 (BST) Subject: [Freeswitch-users] voicemail issue with happy end Message-ID: <1365500292.17076.YahooMailNeo@web171804.mail.ir2.yahoo.com> Hi, I like to share with FS community a strange issue, now fixed, since it seems many people has problems with sendmail and FS. I've a couple of servers with Debian 6.0.7 and FreeSWITCH Version 1.5.1b. A working configuration for voicemail and fax2mail (tested on same distribution)? didn't work:?same old sendmail segfault error. My previous Freeswitch version was last stable (1.2.8) and the issue was always there. Mail failed even calling system sendmail from FS_cli. Rising log on sendmail I've seen that mail passed by FS to sendmail, generated a parsing error on '@' char. So I changed my lua test script as follow: freeswitch.email("testmail at yahoo.it", ? ? ? ? ? ? ? ? ?"info", ? ? ? ? ? ? ? ? ?"subject:Prova\n\n", ? ? ? ? ? ? ? ? ?"Prova.") Note the missing '@mydomain.it' on from argument of the function. I'm not so expert to explain why, but now everything works as expected! Hope this info is useful for someone. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/156ff30d/attachment.html From mehroz.ashraf85 at gmail.com Tue Apr 9 13:57:25 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Tue, 9 Apr 2013 02:57:25 -0700 (PDT) Subject: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode In-Reply-To: References: <2DBB65A0-2551-4FE8-88E0-CB4624325733@edge-net.net> Message-ID: <1365501445584-7589541.post@n2.nabble.com> Any luck with this issue? I am having the same configurations, FS acting as normal mode i.e bypass and proxy on default(commented) Clients are Linphone, and SAS does not match...... Over jitsi ... SAS are identical ONLY IF IF linphone does not support "zrtp-hash" attribute in SIP/SDP, what is the next possible solution? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/problems-with-freeswitch-zrtp-in-proxy-media-mode-tp7586936p7589541.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ashish at nms.co.in Tue Apr 9 14:10:44 2013 From: ashish at nms.co.in (Ashish gautam) Date: Tue, 9 Apr 2013 15:40:44 +0530 Subject: [Freeswitch-users] Default maximum number of simultaneous outgoing PSTN calls In-Reply-To: References: <3E944195-31F2-4239-A9FB-D9F9D5758557@freeswitch.org> Message-ID: Thanks Steven. The provider is saying the SETUP ACK requests are being blocked on my side and probably this is the reason its sending SETUP signals to the Telecom switch again and again. What could be the issue? -- Regards. Ashish On Tue, Apr 9, 2013 at 12:30 PM, Ashish gautam wrote: > I am currently on FS version 1.3. Should I switch back to 1.2, will it > help? > > Also the Dahdi version is 2.4 and the libpri version is 1.4.12. Are all > the versions I am using fine? > > > On Tue, Apr 9, 2013 at 4:15 AM, Steven Ayre wrote: > >> Compare CDRs with the provider to see whether they see the calls. If they >> do the problem isn't at your end. They could be generating that cause for a >> number of reasons, or it could be coming from even further upstream. >> >> Steve >> >> >> >> On 8 Apr 2013, at 12:57, Ashish gautam wrote: >> >> Steve, >> >> I have a PRI line and I have also checked with the provider that all the >> channels are open. >> >> Thanks. >> >> -- Ashish >> >> On Mon, Apr 8, 2013 at 5:16 PM, Steven Ayre wrote: >> >>> Also what kind of line do you have? PRI/BRI sets a limits on the number >>> of channels. Even though PRI gives 32 the provider might be limiting it to >>> 10. >>> >>> >>> On 8 April 2013 02:42, Brian West wrote: >>> >>>> How many calls are you trying to start at once? You could just be >>>> pissing off the switch on the other side by bring up too many calls quickly. >>>> On Apr 6, 2013, at 2:18 AM, Ashish gautam wrote: >>>> >>>> > Hi, >>>> > >>>> > I am making simultaneous outgoing calls to PSTN network through >>>> freeTDM and >>>> > PRI card via Event socket. FS only generates at max. four calls and >>>> for the >>>> > rest it shows NORMAL_CIRCUIT_CONGESTION. >>>> > >>>> > What could be the possible reason for this? Is there any default >>>> maximum >>>> > limit on the number of simultaneous outgoing calls through >>>> > freeTDM/libpri/DAHDI stack.? >>>> > >>>> > Kindly throw some light on this. >>>> > >>>> > Regards. >>>> >>>> -- >>>> Brian West >>>> brian at freeswitch.org >>>> FreeSWITCH Solutions, LLC >>>> PO BOX PO BOX 2531 >>>> Brookfield, WI 53008-2531 >>>> >>>> Twitter: @FreeSWITCH_Wire >>>> http://freeswitchcookbook.com >>>> http://freeswitchbook.com >>>> >>>> T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST >>>> iNUM: +883 5100 1286 0410 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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/20130409/4cafb02f/attachment-0001.html From martyn at magiccow.co.uk Tue Apr 9 14:31:37 2013 From: martyn at magiccow.co.uk (Martyn Davies) Date: Tue, 9 Apr 2013 11:31:37 +0100 Subject: [Freeswitch-users] T316 timed out, resending RESTART request In-Reply-To: References: Message-ID: That is puzzling. There is such a thing as a fractional E1/PRI, I'm wondering if there might be a configuration issue on the provider side, i.e. they think there are fewer than 30 channels available. I note that the freetdm config explicitly sets the bearer channel numbers (http://wiki.freeswitch.org/wiki/FreeTDM). It's probably worth checking with the provider that they agree with the settings, e.g. [span wanpipe PRI_1] trunk_type => E1 b-channel => 7:1-15 d-channel => 7:16 b-channel => 7:17-31 I'm guessing that if the 'd-channel' setting was wrong, then nothing would be working, but check the b-channel ranges. I'm assuming that the RESTART is issued by the card, and the RESTART ACK is expected from the network. If the RESTART ACK never comes, then it's fair to ask the telco why this is. The signalling won't work until that handshake is complete. Regards, Martyn On 8 April 2013 13:23, Ashish gautam wrote: > Thanks Martyn for your response. > > Yes some calls go out ( nearly 4 simultaneously ) and for the rest, it shows > congestion. I have checked the signalling protocol its fine. Out of the 30 > channels 4 get restarted normally (probably this is the reason why only four > calls can go out) and rest are not, they remain in the suspended state. > > --Ashish > > > On Mon, Apr 8, 2013 at 5:30 PM, Martyn Davies wrote: >> >> The RESTART is sent out for PRI (and ISDN) when the signalling is >> first established, or the link to the switch needs to be restarted. >> If the switch does not respond to RESTART (and the timer T316 expires >> - IIRC this is a long timeout), then no channels will be available for >> calls. >> >> The first thing to check I think is that your PRI board is definitely >> set to the right variant of signalling protocol, so that it matches >> that at the switch/provider. There are many variations of protocol >> (NI-1, EuroISDN etc), and if there's a mismatch it could be that one >> end or the other is using RESTART to try to recover the link after >> some kind of failure. >> >> Tell me, do any calls go out at all? Or is it that some calls go out, >> then you get congestion? >> >> Regards, >> Martyn >> >> >> On 8 April 2013 06:09, Ashish gautam wrote: >> > Hi, >> > >> > I am getting this warning: "2013-03-15 12:25:15.816238 [WARNING] >> > ftmod_libpri.c:1954 [s1c29][1:29] -- T316 timed out, resending RESTART >> > request" when I start my FS box. >> > >> > After these warnings with three restart attempts I get this error: >> > >> > 2013-03-15 12:25:45.816239 [ERR] ftmod_libpri.c:1950 [s1c31][1:31] -- >> > T316 >> > timed out, channel reached restart attempt limit '3' and is suspended >> > >> > I am also facing NORMAL_CIRCUIT_CONGESTION error when generation >> > outgoing >> > calls through FreeTDM. Is this a bug? >> > >> > Kindly help. >> > >> > -- >> > Ashish >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 mehroz.ashraf85 at gmail.com Tue Apr 9 14:45:43 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Tue, 9 Apr 2013 03:45:43 -0700 (PDT) Subject: [Freeswitch-users] zrtp passthrough In-Reply-To: References: Message-ID: <1365504343201-7589544.post@n2.nabble.com> Any luck with this issue? I am having the same configurations, FS acting as normal mode i.e bypass and proxy on default(commented) Clients are Linphone, and SAS does not match...... Over jitsi ... SAS are identical ONLY IF inbound-proxy-media is TRUE IF linphone does not support "zrtp-hash" attribute in SIP/SDP, what is the next possible solution? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/zrtp-passthrough-tp7586267p7589544.html Sent from the freeswitch-users mailing list archive at Nabble.com. From nandy1925 at gmail.com Tue Apr 9 16:58:21 2013 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Tue, 9 Apr 2013 20:58:21 +0800 Subject: [Freeswitch-users] Connecting Internal IP phones to PSTN In-Reply-To: References: Message-ID: You need to buy a 2-port FXO SIP gateway. You can install an FXO card e.g. Digium. But a gateway is recommended. Configure the gateway and the FS sip_profiles, of course. Create a dialplan in the default context to route the call to the FXO gateway when you dial outside line e.g. 9+local_number. ================================================ www.magicbox.ph - *the better magic* VoIP phone for Filipinos *Lapulapu City, Phils Phone: +63-32-3401807, Mobile: +63-920-6373450 *USA# :* (646)547-1226 *Worldwide:* [**462 + 17772930540 + #*] via any 200+ Access Numbers (37 Countries) On Tue, Apr 9, 2013 at 1:34 AM, Ebrahim Bararian wrote: > Hi, > I have some soft IP phones in an internal local network. > I want to connect these IP phones to PSTN via 2 PSTN lines. > > How many ways are there to solve this problem? > Which hardwares should I buy? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/8e1b1526/attachment.html From ebrahim.bararian at gmail.com Tue Apr 9 18:00:00 2013 From: ebrahim.bararian at gmail.com (Ebrahim Bararian) Date: Tue, 9 Apr 2013 18:30:00 +0430 Subject: [Freeswitch-users] Connecting Internal IP phones to PSTN In-Reply-To: References: Message-ID: You need to buy a 2-port FXO SIP gateway. You can install an FXO card e.g. Digium. But a gateway is recommended. *Why is a gateway recommended?* * * Configure the gateway and the FS sip_profiles, of course. Create a dialplan in the default context to route the call to the FXO gateway when you dial outside line e.g. 9+local_number. * and how can I set the freeswitch to route the PSTN calls to the IP phone?* On Tue, Apr 9, 2013 at 5:28 PM, Nandy Dagondon wrote: > You need to buy a 2-port FXO SIP gateway. You can install an FXO card e.g. > Digium. But a gateway is recommended. Configure the gateway and the FS > sip_profiles, of course. Create a dialplan in the default context to route > the call to the FXO gateway when you dial outside line e.g. 9+local_number. > > ================================================ > www.magicbox.ph - *the better magic* VoIP phone for Filipinos > *Lapulapu City, Phils > Phone: +63-32-3401807, > Mobile: +63-920-6373450 > *USA# :* (646)547-1226 > *Worldwide:* [**462 + 17772930540 + #*] via any 200+ Access Numbers (37 > Countries) > > > > On Tue, Apr 9, 2013 at 1:34 AM, Ebrahim Bararian < > ebrahim.bararian at gmail.com> wrote: > >> Hi, >> I have some soft IP phones in an internal local network. >> I want to connect these IP phones to PSTN via 2 PSTN lines. >> >> How many ways are there to solve this problem? >> Which hardwares should I buy? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20130409/aebf1227/attachment.html From krice at freeswitch.org Tue Apr 9 18:46:28 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 09 Apr 2013 09:46:28 -0500 Subject: [Freeswitch-users] zrtp passthrough In-Reply-To: <1365504343201-7589544.post@n2.nabble.com> Message-ID: You can just put FS in proxy mode... On 4/9/13 5:45 AM, "mehroz" wrote: > Any luck with this issue? > > I am having the same configurations, FS acting as normal mode i.e bypass and > proxy on default(commented) > > Clients are Linphone, and SAS does not match...... > > Over jitsi ... SAS are identical ONLY IF inbound-proxy-media is TRUE > > IF linphone does not support "zrtp-hash" attribute in SIP/SDP, what is the > next possible solution? > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/zrtp-passthrough-tp7586267p75895 > 44.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From steveayre at gmail.com Tue Apr 9 19:10:17 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 9 Apr 2013 16:10:17 +0100 Subject: [Freeswitch-users] Connecting Internal IP phones to PSTN In-Reply-To: References: Message-ID: > > *Why is a gateway recommended?* Probably because you can use it from multiple FreeSWITCH servers, or a remote location. Of course that comes with its own problems too (an extra point to secure). *and how can I set the freeswitch to route the PSTN calls to the IP phone?* That'll depend on the hardware. If it's a card then probably via the freetdm endpoint. If via a gateway then you'd probably send the call to the gateway via SIP (sofia endpoint). For more information look at these pages: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge http://wiki.freeswitch.org/wiki/FreeTDM http://wiki.freeswitch.org/wiki/Sofia http://wiki.freeswitch.org/wiki/Getting_Started_Guide Also, FXO would be for a normal analog PSTN connection (like you'd normally have at home). If it's a BRI/PRI connection (pretty common in businesses) you'd need a different (but similar) card/gateway for that. -Steve On 9 April 2013 15:00, Ebrahim Bararian wrote: > You need to buy a 2-port FXO SIP gateway. You can install an FXO card e.g. > Digium. But a gateway is recommended. > *Why is a gateway recommended?* > * > * > Configure the gateway and the FS sip_profiles, of course. Create a > dialplan in the default context to route the call to the FXO gateway when > you dial outside line e.g. 9+local_number. > * and how can I set the freeswitch to route the PSTN calls to the IP > phone?* > > > > > On Tue, Apr 9, 2013 at 5:28 PM, Nandy Dagondon wrote: > >> You need to buy a 2-port FXO SIP gateway. You can install an FXO card >> e.g. Digium. But a gateway is recommended. Configure the gateway and the FS >> sip_profiles, of course. Create a dialplan in the default context to route >> the call to the FXO gateway when you dial outside line e.g. 9+local_number. >> >> ================================================ >> www.magicbox.ph - *the better magic* VoIP phone for Filipinos >> *Lapulapu City, Phils >> Phone: +63-32-3401807, >> Mobile: +63-920-6373450 >> *USA# :* (646)547-1226 >> *Worldwide:* [**462 + 17772930540 + #*] via any 200+ Access Numbers (37 >> Countries) >> >> >> >> On Tue, Apr 9, 2013 at 1:34 AM, Ebrahim Bararian < >> ebrahim.bararian at gmail.com> wrote: >> >>> Hi, >>> I have some soft IP phones in an internal local network. >>> I want to connect these IP phones to PSTN via 2 PSTN lines. >>> >>> How many ways are there to solve this problem? >>> Which hardwares should I buy? >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20130409/dacbee73/attachment.html From krice at freeswitch.org Tue Apr 9 19:11:50 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 09 Apr 2013 10:11:50 -0500 Subject: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode In-Reply-To: <1365501445584-7589541.post@n2.nabble.com> Message-ID: Also, keep in mind that FS in a mode other then proxy or bypass media, when you have a ZRTP call is kind of pointless, that leads to a vector for a man in the middle attack which is also the reason the SASs don't match is you have 2 different ZRTP legs with FS in the middle decrypting and re-encrypting the conversation... There is a way to do Trusted MITM I believe there is a setting for that On 4/9/13 4:57 AM, "mehroz" wrote: > Any luck with this issue? > > I am having the same configurations, FS acting as normal mode i.e bypass and > proxy on default(commented) > > Clients are Linphone, and SAS does not match...... > > Over jitsi ... SAS are identical ONLY IF > > IF linphone does not support "zrtp-hash" attribute in SIP/SDP, what is the > next possible solution? > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/problems-with-freeswitch-zrtp-in > -proxy-media-mode-tp7586936p7589541.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From marwan.idriss at gmail.com Tue Apr 9 17:22:37 2013 From: marwan.idriss at gmail.com (Marwan Idriss) Date: Tue, 9 Apr 2013 16:22:37 +0300 Subject: [Freeswitch-users] Java ESL example for outbound connection Message-ID: Greeting, I am trying to use Java API to control outbound socket, But I am not finding any example especially about this idea : listen for an incoming connection on a socket, accept the incoming connection from FreeSWITCH, fork a new copy of your process if you want to listen for more connections, and then pass the file number of the socket to new($fd). so if you can provide me with can example java example for using new($fd) is enough thanks for your effort -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/4b4b26e5/attachment.html From steveayre at gmail.com Tue Apr 9 20:00:21 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 9 Apr 2013 17:00:21 +0100 Subject: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode In-Reply-To: References: <1365501445584-7589541.post@n2.nabble.com> Message-ID: > > There is a way to do Trusted MITM I believe there is a setting for that On 9 April 2013 16:11, Ken Rice wrote: > Also, keep in mind that FS in a mode other then proxy or bypass media, when > you have a ZRTP call is kind of pointless, that leads to a vector for a man > in the middle attack which is also the reason the SASs don't match is you > have 2 different ZRTP legs with FS in the middle decrypting and > re-encrypting the conversation... > > There is a way to do Trusted MITM I believe there is a setting for that > > > On 4/9/13 4:57 AM, "mehroz" wrote: > > > Any luck with this issue? > > > > I am having the same configurations, FS acting as normal mode i.e bypass > and > > proxy on default(commented) > > > > Clients are Linphone, and SAS does not match...... > > > > Over jitsi ... SAS are identical ONLY IF > > > > IF linphone does not support "zrtp-hash" attribute in SIP/SDP, what is > the > > next possible solution? > > > > > > > > -- > > View this message in context: > > > http://freeswitch-users.2379917.n2.nabble.com/problems-with-freeswitch-zrtp-in > > -proxy-media-mode-tp7586936p7589541.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 > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #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/20130409/c6096d59/attachment-0001.html From Hector.Geraldino at ipsoft.com Tue Apr 9 20:24:29 2013 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Tue, 9 Apr 2013 16:24:29 +0000 Subject: [Freeswitch-users] Java ESL example for outbound connection In-Reply-To: References: Message-ID: Why are you taking such a complicated approach? Wouldn't it be simpler (and better) to just leverage on 3rd party libraries (such as netty) to handle everything related to sockets and forking processes (or triggering new threads), and focus on what your application actually has to do? I would recommend you to take a look at the Java ESL Client library [http://wiki.freeswitch.org/wiki/Java_ESL_Client], a java nio (netty) library that can help you to accomplish what you're trying to do. When working in outbound mode, FreeSWITCH will establish a new socket connection for each incoming call, and you don't really have to worry about handling the sockets as every new connection is handled in its own thread. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Marwan Idriss Sent: Tuesday, April 09, 2013 9:23 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Java ESL example for outbound connection Greeting, I am trying to use Java API to control outbound socket, But I am not finding any example especially about this idea : listen for an incoming connection on a socket, accept the incoming connection from FreeSWITCH, fork a new copy of your process if you want to listen for more connections, and then pass the file number of the socket to new($fd). so if you can provide me with can example java example for using new($fd) is enough thanks for your effort -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/f8f88ee0/attachment.html From sdevoy at bizfocused.com Tue Apr 9 20:28:15 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 9 Apr 2013 12:28:15 -0400 Subject: [Freeswitch-users] Voicemail Logfile questions. Message-ID: <003c01ce353f$3c9fa3f0$b5deebd0$@bizfocused.com> HI, I had a user say she got disconnected from voicemail and wanted to know why. I saw in the log where she called in, listened, pressed 1, listened again and then: 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:247 Handle play-file:[voicemail/vm-listen_saved.wav] (en:en) 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:1332 Codec Activated L16 at 16000hz 1 channels 20ms 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] 2013-04-09 11:14:38.114065 [DEBUG] sofia.c:5599 Channel sofia/external/@ entering state [terminated][487] 2013-04-09 11:14:38.114065 [DEBUG] switch_channel.c:2994 (sofia/external/@) Callstate Change EARLY -> HANGUP 2013-04-09 11:14:38.114065 [NOTICE] sofia.c:6383 Hangup sofia/external/@ [CS_EXECUTE] [ORIGINATOR_CANCEL] So if FS says she hung up and she says FS hung up, is it safe to say her internet connection failed momentarily? Also, why "Callstate Change EARLY -> HANGUP". Shouldn't it be "Callstate Change ACTIVE -> HANGUP"? Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/2c61964f/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/2c61964f/attachment.gif From vishal.kakkar at gmail.com Tue Apr 9 20:34:35 2013 From: vishal.kakkar at gmail.com (Vishal Kakkar) Date: Tue, 9 Apr 2013 22:04:35 +0530 Subject: [Freeswitch-users] Freetdm Channel utilization in order(ascending/descending) Message-ID: [span wanpipe wp1] trunk_type => e1 group=dialGrp1 b-channel => 2:1-15 b-channel => 2:17-31 d-channel => 2:16 [span wanpipe wp2] trunk_type => e1 group=dialGrp1 b-channel => 1:1-15 b-channel => 1:17-31 d-channel => 1:16 freetdm/dialGrp1/*a*/xxxxx initiates call from channel 1 of span 2 i.e. (2:1) freetdm/dialGrp1/*A/*xxxxx initiates call from channel 31 of span 2 i.e. (2:31).. but i was expecting it to be 31st channel of Span1. Also 2nd question in above config all calls are going through wp1(span2) only.. even if i initiate more than 30 calls simultaneously.. no call goes to wp2(span 1) Can anyone please help how can i make ascending/descending channel utilization using one single group across both above spans.. Thanks a lot.. FS Rocks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/045df626/attachment.html From sibxol at btconnect.com Tue Apr 9 20:44:26 2013 From: sibxol at btconnect.com (sibu) Date: Tue, 9 Apr 2013 17:44:26 +0100 Subject: [Freeswitch-users] freeswitch on ipsec Message-ID: <201304091744.26773.sibxol@btconnect.com> Dear Freeswitch-Users/Developers I am new to this list and freeswitch. I would like to know if anyone has tried freeswitch with an ipsec VPN such as openswan or strongswan and what were/should-be the settings (eg transport- mode, tunnel-mode etc), results and requirements vis a vis cpu-power, network-speed etc etc all hints and suggestions welcomed thanks in advance sibu xolo From mike at jerris.com Tue Apr 9 21:06:28 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 9 Apr 2013 13:06:28 -0400 Subject: [Freeswitch-users] Voicemail Logfile questions. In-Reply-To: <003c01ce353f$3c9fa3f0$b5deebd0$@bizfocused.com> References: <003c01ce353f$3c9fa3f0$b5deebd0$@bizfocused.com> Message-ID: this is exactly 32 seconds in to the call, right? On Apr 9, 2013, at 12:28 PM, Sean Devoy wrote: > HI, > > I had a user say she got disconnected from voicemail and wanted to know why. > > I saw in the log where she called in, listened, pressed 1, listened again and then: > > 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:247 Handle play-file:[voicemail/vm-listen_saved.wav] (en:en) > 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:1332 Codec Activated L16 at 16000hz 1 channels 20ms > 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] > 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] > 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] > 2013-04-09 11:14:38.114065 [DEBUG] sofia.c:5599 Channel sofia/external/@ entering state [terminated][487] > 2013-04-09 11:14:38.114065 [DEBUG] switch_channel.c:2994 (sofia/external/@) Callstate Change EARLY -> HANGUP > 2013-04-09 11:14:38.114065 [NOTICE] sofia.c:6383 Hangup sofia/external/@ [CS_EXECUTE] [ORIGINATOR_CANCEL] > > So if FS says she hung up and she says FS hung up, is it safe to say her internet connection failed momentarily? > > Also, why ?Callstate Change EARLY -> HANGUP?. Shouldn?t it be ?Callstate Change ACTIVE -> HANGUP?? > > Thanks, > Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/b0b0c1cf/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Tue Apr 9 21:07:19 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 9 Apr 2013 18:07:19 +0100 Subject: [Freeswitch-users] Metaswitch perimeta SBC - any thoughts? Message-ID: Hello all, Does anyone here have any first hand experience using the Metaswitch perimeta SBC with FreeSWITCH? http://www.metaswitch.com/products/perimeta-session-border-controller I know this isn't strictly on-topic, but I find the people on this list are usually best placed to give useful feedback on related products. At the moment we're weighing up the pros/cons of using our own in-house SBC, F5 LTM, Stingray TM, OpenSER, or buying a purpose made SBC such as the Perimeta. Any thoughts? Cal (Mods: if you aren't happy with this being discussed on the list, just let me know). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/9d9ea819/attachment.html From eburke at edge-net.net Tue Apr 9 21:17:48 2013 From: eburke at edge-net.net (Eli Burke) Date: Tue, 9 Apr 2013 13:17:48 -0400 Subject: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode In-Reply-To: References: Message-ID: We were never able to get Trusted MITM working, but ZRTP can work in proxy-media mode as long as your client sends the zrtp-hash attribute in its SDP. When Freeswitch sees the zrtp-hash attribute it enables logic to pass the zrtp packets through unmolested (using the same SSRC for each call leg). As Mehroz noted in his original question, Linphone does not currently support zrtp-hash so this can't work. However, Mehroz, if you look in the Feb 2013 archives of the linphone-developers mailing list you can find a patch that adds zrtp-hash support to Linphone. It was recently rejected because it causes side-effects when video is enabled, but you can test it out and see if it fixes the your problem. -Eli On Apr 9, 2013, at 12:01 PM, freeswitch-users-request at lists.freeswitch.org wrote: > From: Steven Ayre > Subject: Re: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode > Date: April 9, 2013 12:00:21 PM EDT > To: FreeSWITCH Users Help > Reply-To: FreeSWITCH Users Help > > > There is a way to do Trusted MITM I believe there is a setting for that > > > > > On 9 April 2013 16:11, Ken Rice wrote: > Also, keep in mind that FS in a mode other then proxy or bypass media, when > you have a ZRTP call is kind of pointless, that leads to a vector for a man > in the middle attack which is also the reason the SASs don't match is you > have 2 different ZRTP legs with FS in the middle decrypting and > re-encrypting the conversation... > > There is a way to do Trusted MITM I believe there is a setting for that > > > On 4/9/13 4:57 AM, "mehroz" wrote: > > > Any luck with this issue? > > > > I am having the same configurations, FS acting as normal mode i.e bypass and > > proxy on default(commented) > > > > Clients are Linphone, and SAS does not match...... > > > > Over jitsi ... SAS are identical ONLY IF > > > > IF linphone does not support "zrtp-hash" attribute in SIP/SDP, what is the > > next possible solution? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/3302dea6/attachment.html From mike at jerris.com Tue Apr 9 21:20:58 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 9 Apr 2013 13:20:58 -0400 Subject: [Freeswitch-users] Metaswitch perimeta SBC - any thoughts? In-Reply-To: References: Message-ID: <44A43FCD-DBAC-4961-9C64-251046D4E0E3@jerris.com> Can you describe a bit more what exactly you want to get out of an SBC for your setup? This topic is totally fine for the list Mike On Apr 9, 2013, at 1:07 PM, Cal Leeming [Simplicity Media Ltd] wrote: > Hello all, > > Does anyone here have any first hand experience using the Metaswitch perimeta SBC with FreeSWITCH? > http://www.metaswitch.com/products/perimeta-session-border-controller > > I know this isn't strictly on-topic, but I find the people on this list are usually best placed to give useful feedback on related products. > > At the moment we're weighing up the pros/cons of using our own in-house SBC, F5 LTM, Stingray TM, OpenSER, or buying a purpose made SBC such as the Perimeta. > > Any thoughts? > > Cal > > (Mods: if you aren't happy with this being discussed on the list, just let me know). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/30ebe713/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue Apr 9 21:41:28 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 9 Apr 2013 18:41:28 +0100 Subject: [Freeswitch-users] Metaswitch perimeta SBC - any thoughts? In-Reply-To: <44A43FCD-DBAC-4961-9C64-251046D4E0E3@jerris.com> References: <44A43FCD-DBAC-4961-9C64-251046D4E0E3@jerris.com> Message-ID: We attempted to use FreeSWITCH as an SBC but came up against problems scaling. Instead my idea was to place a dumb proxy in front of lots of FreeSWITCH instances, with two profiles exposed.. one for UC registrations, and another for the backend FS instances. When a UC sends a register, it forwards the request to a backend, if it was successful then a persist entry is created. If a user tried to call another user in the same domain (or another domain that we hold), it would send the request through to the SBC, the SBC would then check the persistance table for an entry for that user, and redirect the query to the appropriate server where the user was currently registered. If no entry was found, a random server would be picked to handle voicemail and redirects etc. Any INVITEs would also have their SIP/SDP re-written to allow RTP proxying as well for full topology hiding. This allows a single domain to be spread across an unlimited amount of servers. In theory the bottleneck would be the speed of which the SBC can proxy packets.. initial tests show that a python prototype can forward approx 350mbit/sec on a single thread (using eventlet) before it starts to choke, and we were able to saturate a gbit link with zero packet loss and jitter using 8 threads (that's with full SIP/SDP parsing too). We've tested using the F5 and Stingray too, but their throughput isn't as good. However, my concern is that no one (from what I can tell) is using this approach to scale voice platforms, and this makes me question whether or not it is the correct thing to do. Cal On Tue, Apr 9, 2013 at 6:20 PM, Michael Jerris wrote: > Can you describe a bit more what exactly you want to get out of an SBC for > your setup? This topic is totally fine for the list > > Mike > > > On Apr 9, 2013, at 1:07 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > Hello all, > > Does anyone here have any first hand experience using the Metaswitch > perimeta SBC with FreeSWITCH? > http://www.metaswitch.com/products/perimeta-session-border-controller > > I know this isn't strictly on-topic, but I find the people on this list > are usually best placed to give useful feedback on related products. > > At the moment we're weighing up the pros/cons of using our own in-house > SBC, F5 LTM, Stingray TM, OpenSER, or buying a purpose made SBC such as the > Perimeta. > > Any thoughts? > > Cal > > (Mods: if you aren't happy with this being discussed on the list, just let > me know). > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/117f4c23/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue Apr 9 22:42:40 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 9 Apr 2013 19:42:40 +0100 Subject: [Freeswitch-users] freeswitch on ipsec In-Reply-To: <201304091744.26773.sibxol@btconnect.com> References: <201304091744.26773.sibxol@btconnect.com> Message-ID: >From personal experience, I would strongly recommend running the VPN within the network layer rather than directly on the server. You could use a pre-built appliance for this (such as Halon VSR), or build your own router using iptables, or even use a Cisco etc. This means you don't have to maintain a tunnel for each individual machines, and keeps a nice clean separation of layers which makes debugging networking problems easier. If you use this approach, then running FreeSWITCH over any tunnel should *just work*.. if you use it locally, then strange things might happen.. This is just based on my own personal experience, others may disagree.. YMMV :) Hope this helps Cal On Tue, Apr 9, 2013 at 5:44 PM, sibu wrote: > Dear Freeswitch-Users/Developers > > I am new to this list and freeswitch. > > I would like to know if anyone has tried freeswitch with an ipsec VPN such > as > openswan or strongswan and what were/should-be the settings (eg transport- > mode, tunnel-mode etc), results and requirements vis a vis cpu-power, > network-speed etc etc > > all hints and suggestions welcomed > thanks in advance > sibu xolo > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/d4af3d3d/attachment-0001.html From sdevoy at bizfocused.com Tue Apr 9 22:54:57 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 9 Apr 2013 14:54:57 -0400 Subject: [Freeswitch-users] Lost connection during voicemail session Message-ID: <007701ce3553$bae80a90$30b81fb0$@bizfocused.com> HI, I had a user say she got disconnected from voicemail and wanted to know why. I saw in the log where she called in, listened, pressed 1, listened again and then: 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:247 Handle play-file:[voicemail/vm-listen_saved.wav] (en:en) 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:1332 Codec Activated L16 at 16000hz 1 channels 20ms 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] 2013-04-09 11:14:38.114065 [DEBUG] sofia.c:5599 Channel sofia/external/@ entering state [terminated][487] 2013-04-09 11:14:38.114065 [DEBUG] switch_channel.c:2994 (sofia/external/@) Callstate Change EARLY -> HANGUP 2013-04-09 11:14:38.114065 [NOTICE] sofia.c:6383 Hangup sofia/external/@ [CS_EXECUTE] [ORIGINATOR_CANCEL] So if FS says she hung up and she says FS hung up, is it safe to say her internet connection failed momentarily? Also, why "Callstate Change EARLY -> HANGUP". Shouldn't it be "Callstate Change ACTIVE -> HANGUP"? Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/c9d58f28/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/c9d58f28/attachment.gif From anthony.minessale at gmail.com Tue Apr 9 23:01:32 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 9 Apr 2013 14:01:32 -0500 Subject: [Freeswitch-users] Lost connection during voicemail session In-Reply-To: <007701ce3553$bae80a90$30b81fb0$@bizfocused.com> References: <007701ce3553$bae80a90$30b81fb0$@bizfocused.com> Message-ID: Look at more of the log and see if the channel was ever answered. If you are routing the call to voicemail by way of dial-plan just add the answer app before you reach voicemail. On Tue, Apr 9, 2013 at 1:54 PM, Sean Devoy wrote: > HI,**** > > ** ** > > I had a user say she got disconnected from voicemail and wanted to know > why. **** > > ** ** > > I saw in the log where she called in, listened, pressed 1, listened again > and then:**** > > ** ** > > 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:247 Handle > play-file:[voicemail/vm-listen_saved.wav] (en:en)**** > > 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:1332 Codec > Activated L16 at 16000hz 1 channels 20ms**** > > 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/@ [BREAK]**** > > 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/@ [BREAK]**** > > 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/@ [BREAK]**** > > 2013-04-09 11:14:38.114065 [DEBUG] sofia.c:5599 Channel > sofia/external/@ entering state [terminated][487]**** > > 2013-04-09 11:14:38.114065 [DEBUG] switch_channel.c:2994 > (sofia/external/@) Callstate Change EARLY -> HANGUP**** > > 2013-04-09 11:14:38.114065 [NOTICE] sofia.c:6383 Hangup > sofia/external/@ [CS_EXECUTE] [ORIGINATOR_CANCEL]**** > > ** ** > > So if FS says she hung up and she says FS hung up, is it safe to say her > internet connection failed momentarily?**** > > ** ** > > Also, why ?Callstate Change EARLY -> HANGUP?. Shouldn?t it be ?Callstate > Change ACTIVE -> HANGUP??**** > > ** ** > > Thanks,**** > > Sean**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130409/e4c9957b/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/e4c9957b/attachment.gif From schoch+freeswitch.org at xwin32.com Tue Apr 9 23:27:29 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 9 Apr 2013 12:27:29 -0700 Subject: [Freeswitch-users] loopback Message-ID: In the default dialplan, it bridges a call to voicemail using these actions: I have found the the following also seems to work: My question is why does the default dialplan use a bridge to loopback instead of just calling the voicemail app directly? What is the difference? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/b34cde78/attachment-0001.html From sdevoy at bizfocused.com Tue Apr 9 23:40:47 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 9 Apr 2013 15:40:47 -0400 Subject: [Freeswitch-users] Lost connection during voicemail session In-Reply-To: References: <007701ce3553$bae80a90$30b81fb0$@bizfocused.com> Message-ID: <01ca01ce355a$2286c730$67945590$@bizfocused.com> Of course you are right the first time AM! Should I also be answering before bridging calls? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Tuesday, April 09, 2013 3:02 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Lost connection during voicemail session Look at more of the log and see if the channel was ever answered. If you are routing the call to voicemail by way of dial-plan just add the answer app before you reach voicemail. On Tue, Apr 9, 2013 at 1:54 PM, Sean Devoy wrote: HI, I had a user say she got disconnected from voicemail and wanted to know why. I saw in the log where she called in, listened, pressed 1, listened again and then: 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:247 Handle play-file:[voicemail/vm-listen_saved.wav] (en:en) 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:1332 Codec Activated L16 at 16000hz 1 channels 20ms 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] 2013-04-09 11:14:38.114065 [DEBUG] sofia.c:5599 Channel sofia/external/@ entering state [terminated][487] 2013-04-09 11:14:38.114065 [DEBUG] switch_channel.c:2994 (sofia/external/@) Callstate Change EARLY -> HANGUP 2013-04-09 11:14:38.114065 [NOTICE] sofia.c:6383 Hangup sofia/external/@ [CS_EXECUTE] [ORIGINATOR_CANCEL] So if FS says she hung up and she says FS hung up, is it safe to say her internet connection failed momentarily? Also, why "Callstate Change EARLY -> HANGUP". Shouldn't it be "Callstate Change ACTIVE -> HANGUP"? Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130409/ec78d259/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/ec78d259/attachment.gif From anthony.minessale at gmail.com Tue Apr 9 23:45:58 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 9 Apr 2013 14:45:58 -0500 Subject: [Freeswitch-users] Lost connection during voicemail session In-Reply-To: <01ca01ce355a$2286c730$67945590$@bizfocused.com> References: <007701ce3553$bae80a90$30b81fb0$@bizfocused.com> <01ca01ce355a$2286c730$67945590$@bizfocused.com> Message-ID: You don't have to answer before bridging calls because the answer will be inherited. Its a special thing to do with billing etc. But when talking to apps its advised to answer before running the app. Only a small number of apps answer for you. On Tue, Apr 9, 2013 at 2:40 PM, Sean Devoy wrote: > Of course you are right the first time AM!**** > > ** ** > > Should I also be answering before bridging calls?**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Tuesday, April 09, 2013 3:02 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Lost connection during voicemail session > **** > > ** ** > > Look at more of the log and see if the channel was ever answered. If you > are routing the call to voicemail by way of dial-plan just add the answer > app before you reach voicemail.**** > > ** ** > > ** ** > > On Tue, Apr 9, 2013 at 1:54 PM, Sean Devoy wrote:* > *** > > **** > > HI,**** > > **** > > I had a user say she got disconnected from voicemail and wanted to know > why. **** > > **** > > I saw in the log where she called in, listened, pressed 1, listened again > and then:**** > > **** > > 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:247 Handle > play-file:[voicemail/vm-listen_saved.wav] (en:en)**** > > 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:1332 Codec > Activated L16 at 16000hz 1 channels 20ms**** > > 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/@ [BREAK]**** > > 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/@ [BREAK]**** > > 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/@ [BREAK]**** > > 2013-04-09 11:14:38.114065 [DEBUG] sofia.c:5599 Channel > sofia/external/@ entering state [terminated][487]**** > > 2013-04-09 11:14:38.114065 [DEBUG] switch_channel.c:2994 > (sofia/external/@) Callstate Change EARLY -> HANGUP**** > > 2013-04-09 11:14:38.114065 [NOTICE] sofia.c:6383 Hangup > sofia/external/@ [CS_EXECUTE] [ORIGINATOR_CANCEL]**** > > **** > > So if FS says she hung up and she says FS hung up, is it safe to say her > internet connection failed momentarily?**** > > **** > > Also, why ?Callstate Change EARLY -> HANGUP?. Shouldn?t it be ?Callstate > Change ACTIVE -> HANGUP??**** > > **** > > Thanks,**** > > Sean**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130409/afa9ee8b/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/afa9ee8b/attachment-0001.gif From vipkilla at gmail.com Wed Apr 10 00:03:24 2013 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 9 Apr 2013 16:03:24 -0400 Subject: [Freeswitch-users] loopback In-Reply-To: References: Message-ID: It is just there as an example of one way to use loopback to call other applications. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/61631107/attachment.html From nandy1925 at gmail.com Wed Apr 10 00:24:34 2013 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 10 Apr 2013 04:24:34 +0800 Subject: [Freeswitch-users] CDR In-Reply-To: References: Message-ID: the ${destination_number} field constitute the dialed (DTMF) digits. ================================================ www.magicbox.ph - *the better magic* VoIP phone for Filipinos *Lapulapu City, Phils Phone: +63-32-3401807, Mobile: +63-920-6373450 *USA# :* (646)547-1226 *Worldwide:* [**462 + 17772930540 + #*] via any 200+ Access Numbers (37 Countries) On Mon, Apr 8, 2013 at 4:23 PM, Khue Nguyen Minh wrote: > I added this line into cdr_csv.conf.xml > > but, in output file I don't receive dtmf. This is output file > "khuenm", "984713985", "301", "" > "khuenm", "984713985", "302", "" > > > > 2013/4/8 Avi Marcus > >> Try logging: digits_dialed >> >> -Avi Marcus >> BestFone >> >> >> On Mon, Apr 8, 2013 at 6:47 AM, Khue Nguyen Minh wrote: >> >>> I was export cdr file success. Thank you very much. >>> >>> Now, I want write all dtmf into cdr file, how I can do that? Please help >>> me. >>> >>> Brs, >>> Khue >>> >>> >>> 2013/4/5 Ken Rice >>> >>>> You don?t do that in the dialplan, mod_cdr_csv does that for you >>>> >>>> >>>> >>>> On 4/5/13 2:24 AM, "Khue Nguyen Minh" wrote: >>>> >>>> Hi all, >>>> >>>> I want create cdr (filetype: csv) file in freeswitch. But, I don't >>>> known how I can do it from xml dialplan. Please guide me. >>>> >>>> Thanks & Best regards, >>>> Khue. >>>> >>>> ------------------------------ >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> -- >>>> Ken >>>> *http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> *irc.freenode.net #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/20130410/bf14207e/attachment.html From nandy1925 at gmail.com Wed Apr 10 00:32:13 2013 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 10 Apr 2013 04:32:13 +0800 Subject: [Freeswitch-users] pstn call control from computer In-Reply-To: References: Message-ID: Hi Zhnu, Welcome to FS. I have tried HT503. It worked but I have issues with humming noise. I wouldn't recommend this. I haven't tried Obihai. Re controlling incoming calls from a computer. FS has a powerful module mod_event_socket. http://wiki.freeswitch.org/wiki/Event_Socket . You can create your script to do what you want. FS is really a telephony Swiss knife and I think there are other modules available. Just go over the Wiki. Cheers, /Nandy ================================================ www.magicbox.ph - *the better magic* VoIP phone for Filipinos *Lapulapu City, Phils Phone: +63-32-3401807, Mobile: +63-920-6373450 *USA# :* (646)547-1226 *Worldwide:* [**462 + 17772930540 + #*] via any 200+ Access Numbers (37 Countries) On Sat, Apr 6, 2013 at 1:50 AM, Zhnupy Gonzalez wrote: > Hi > I'm a total newbie looking a way to controll PSTN calls from my computer, > by control I mean: see caller id on screen, answering with softh phone, > pressing a button to play a song, that kind of stuff. > Is it possible with hardware like grandstream Ht503 or obihai OBi110 or do > I need more (and more expensive) hardware? > > regards > zhnu > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/e42da6a4/attachment.html From gadle.arpit at gmail.com Wed Apr 10 00:05:19 2013 From: gadle.arpit at gmail.com (Arpit Gadle) Date: Wed, 10 Apr 2013 01:35:19 +0530 Subject: [Freeswitch-users] FreeSwitch 1.3.7 Final - mod_freetdm Compilation Failure Message-ID: Hi All, I had downloaded *freeswitch-1.3.17-final.tar.gz *from http://git.freeswitch.org/git/freeswitch/. ./bootstrap.sh and ./configure steps were completed successfully but make command failed with following error message *CC ftmod_sangoma_ss7_la-ftmod_sangoma_ss7_support.lo* *In file included from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:55,* * from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:35:* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:77: error: ?SCT_MAX_NET_ADDRS? undeclared here (not in a function)* *In file included from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:55,* * from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:35:* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:105: error: ?MW_MAX_NUM_OF_PEER? undeclared here (not in a function)* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:109: error: ?MW_MAX_NUM_OF_INTF? undeclared here (not in a function)* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:112: error: ?MW_MAX_NUM_OF_CLUSTER? undeclared here (not in a function)* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:129: error: expected declaration specifiers or ?...? before ?SbMgmt?* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:130: error: expected declaration specifiers or ?...? before ?MwMgmt?* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:131: error: expected declaration specifiers or ?...? before ?NwMgmt?* *In file included from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:35:* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:804: error: expected declaration specifiers or ?...? before ?MwMgmt?* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:805: error: expected declaration specifiers or ?...? before ?NwMgmt?* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:806: error: expected declaration specifiers or ?...? before ?HiMngmt?* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:807: error: expected declaration specifiers or ?...? before ?SbMgmt?* *cc1: warnings being treated as errors* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c: In function ?sngss7_save_iam?:* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:3340: error: format ?%d? expects type ?int?, but argument 10 has type ?size_t?* *make[6]: *** [ftmod_sangoma_ss7_la-ftmod_sangoma_ss7_support.lo] Error 1* *make[5]: *** [../libfreetdm.la] Error 2* *make[4]: *** [all] Error 1* *make[3]: *** [mod_freetdm-all] Error 1* *make[2]: *** [all-recursive] Error 1* *make[1]: *** [all-recursive] Error 1* *make: *** [all] Error 2* * * Any help is appreciated. Thanks & Regards, Arpit Gadle -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/12af1284/attachment-0001.html From msc at freeswitch.org Wed Apr 10 01:31:25 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 9 Apr 2013 14:31:25 -0700 Subject: [Freeswitch-users] CDR In-Reply-To: References: Message-ID: Well, sort of. The destination_number is essentially the number that the caller dialed. The dialed_digits chan var contains a list of all the DTMFs that caller dialed. This assumes, of course, that the system was listening for DTMFs. If the caller's DTMFs are inband then you'll need to run the start_dtmf application as soon as the call is answered, otherwise FreeSWITCH won't listen for in-band DTMFs. -MC On Tue, Apr 9, 2013 at 1:24 PM, Nandy Dagondon wrote: > the ${destination_number} field constitute the dialed (DTMF) digits. > > ================================================ > www.magicbox.ph - *the better magic* VoIP phone for Filipinos > *Lapulapu City, Phils > Phone: +63-32-3401807, > Mobile: +63-920-6373450 > *USA# :* (646)547-1226 > *Worldwide:* [**462 + 17772930540 + #*] via any 200+ Access Numbers (37 > Countries) > > > > On Mon, Apr 8, 2013 at 4:23 PM, Khue Nguyen Minh wrote: > >> I added this line into cdr_csv.conf.xml >> >> but, in output file I don't receive dtmf. This is output file >> "khuenm", "984713985", "301", "" >> "khuenm", "984713985", "302", "" >> >> >> >> 2013/4/8 Avi Marcus >> >>> Try logging: digits_dialed >>> >>> -Avi Marcus >>> BestFone >>> >>> >>> On Mon, Apr 8, 2013 at 6:47 AM, Khue Nguyen Minh wrote: >>> >>>> I was export cdr file success. Thank you very much. >>>> >>>> Now, I want write all dtmf into cdr file, how I can do that? Please >>>> help me. >>>> >>>> Brs, >>>> Khue >>>> >>>> >>>> 2013/4/5 Ken Rice >>>> >>>>> You don?t do that in the dialplan, mod_cdr_csv does that for you >>>>> >>>>> >>>>> >>>>> On 4/5/13 2:24 AM, "Khue Nguyen Minh" wrote: >>>>> >>>>> Hi all, >>>>> >>>>> I want create cdr (filetype: csv) file in freeswitch. But, I don't >>>>> known how I can do it from xml dialplan. Please guide me. >>>>> >>>>> Thanks & Best regards, >>>>> Khue. >>>>> >>>>> ------------------------------ >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> -- >>>>> Ken >>>>> *http://www.FreeSWITCH.org >>>>> http://www.ClueCon.com >>>>> http://www.OSTAG.org >>>>> *irc.freenode.net #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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/8bd066a1/attachment.html From steveayre at gmail.com Wed Apr 10 02:30:30 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 9 Apr 2013 23:30:30 +0100 Subject: [Freeswitch-users] Lost connection during voicemail session In-Reply-To: References: <007701ce3553$bae80a90$30b81fb0$@bizfocused.com> <01ca01ce355a$2286c730$67945590$@bizfocused.com> Message-ID: <08C7907B-0CFE-4B3F-B302-95B6DDC2F538@gmail.com> They'll hear the voicemail in early media - as long as their phone doesn't suppress it and generate its own ring back anyway. The phone probably won't send audio though, and that probably includes dtmf for interacting with the switch. Probably better to answer. The log shows 487, which generally is the reply to CANCEL, ie they hung up first Steve On 9 Apr 2013, at 20:45, Anthony Minessale wrote: > You don't have to answer before bridging calls because the answer will be inherited. Its a special thing to do with billing etc. > But when talking to apps its advised to answer before running the app. Only a small number of apps answer for you. > > > > On Tue, Apr 9, 2013 at 2:40 PM, Sean Devoy wrote: >> Of course you are right the first time AM! >> >> >> >> Should I also be answering before bridging calls? >> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale >> Sent: Tuesday, April 09, 2013 3:02 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Lost connection during voicemail session >> >> >> >> Look at more of the log and see if the channel was ever answered. If you are routing the call to voicemail by way of dial-plan just add the answer app before you reach voicemail. >> >> >> >> >> >> On Tue, Apr 9, 2013 at 1:54 PM, Sean Devoy wrote: >> >> >> >> HI, >> >> >> >> I had a user say she got disconnected from voicemail and wanted to know why. >> >> >> >> I saw in the log where she called in, listened, pressed 1, listened again and then: >> >> >> >> 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:247 Handle play-file:[voicemail/vm-listen_saved.wav] (en:en) >> >> 2013-04-09 11:14:37.134093 [DEBUG] switch_ivr_play_say.c:1332 Codec Activated L16 at 16000hz 1 channels 20ms >> >> 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] >> >> 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] >> >> 2013-04-09 11:14:38.094065 [DEBUG] switch_core_session.c:975 Send signal sofia/external/@ [BREAK] >> >> 2013-04-09 11:14:38.114065 [DEBUG] sofia.c:5599 Channel sofia/external/@ entering state [terminated][487] >> >> 2013-04-09 11:14:38.114065 [DEBUG] switch_channel.c:2994 (sofia/external/@) Callstate Change EARLY -> HANGUP >> >> 2013-04-09 11:14:38.114065 [NOTICE] sofia.c:6383 Hangup sofia/external/@ [CS_EXECUTE] [ORIGINATOR_CANCEL] >> >> >> >> So if FS says she hung up and she says FS hung up, is it safe to say her internet connection failed momentarily? >> >> >> >> Also, why ?Callstate Change EARLY -> HANGUP?. Shouldn?t it be ?Callstate Change ACTIVE -> HANGUP?? >> >> >> >> Thanks, >> >> Sean >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at 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 > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at 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/20130409/99f707ee/attachment-0001.html From jeff at jefflenk.com Wed Apr 10 02:45:26 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Tue, 9 Apr 2013 15:45:26 -0700 (PDT) Subject: [Freeswitch-users] loopback In-Reply-To: References: Message-ID: <1365547526840-7589575.post@n2.nabble.com> It also allows an attended transfer to voicemail to work. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/loopback-tp7589565p7589575.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ira at connectmevoice.com Wed Apr 10 03:48:00 2013 From: ira at connectmevoice.com (Ira Tessler) Date: Tue, 9 Apr 2013 19:48:00 -0400 Subject: [Freeswitch-users] Presence Sanity Check In-Reply-To: References: Message-ID: I second the hear hear!!! Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com On Mon, Apr 8, 2013 at 5:50 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hear hear! > > On Mon, Apr 8, 2013 at 10:17 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Its actually somewhat ridiculous and one of my personal favorites in >> terms of RFC smoke and mirrors in SIP to try and cover up a flaw with more >> specs. They basically say: >> >> >> If the total packet including the sip headers and the payload exceeds the >> MTU and you are using the UDP transport, you MUST try sending the packet >> over TCP instead. If that times out or fails then you SHOULD send it over >> UDP anyway. >> >> >> THEREFORE by virtue of this decree: >> >> You MUST implement your sip stack to accept UDP packets of up to 65536 >> bytes. >> >> AND >> >> You MUST implement both TCP and UDP transports. >> >> >> So in short, you are not supposed to send anything over udp that exceeds >> the mtu yet you are required to implement it so its possible. >> >> Many stacks, including Asterisk for many of the first half of FS >> existence, did not implement TCP so with this rule being enforced, the >> packets would sit there for 2-5 min then give up and change to UDP. How's >> that for PDD. >> >> Anyway, we choose to ignore this rule intentionally and just stick with >> the negotiated protocol. If you find yourself in this situation the >> solution is to use TCP. >> >> >> P.S. >> >> If they did not use 2k of XML to transmit about 12 bytes worth of useful >> info regarding the state of the presence, we would not have this problem to >> begin with ;) >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mon, Apr 8, 2013 at 4:02 PM, Ira Tessler wrote: >> >>> Thank you for your information. It did help! >>> >>> Ira >>> >>> Ira Tessler >>> Lead Software Engineer >>> ConnectMe >>> (732) 490-9007 x2 >>> ira at connectmevoice.com >>> >>> >>> On Sun, Apr 7, 2013 at 12:49 PM, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>>> In regards to the UDP fragmentation, this is an extremely good question. >>>> >>>> Only yesterday I started to build a simple forwarding SBC in Python >>>> using UDP sockets, however I came up against the same theoretical problem >>>> of packets being larger than 1500 bytes. >>>> >>>> I've had a read through various documentation; >>>> http://www.rfc-ref.org/RFC-TEXTS/3261/chapter18.html >>>> http://www.ietf.org/rfc/rfc3428.txt >>>> >>>> https://lists.cs.columbia.edu/pipermail/sip-implementors/2006-August/013857.html >>>> >>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-February/068372.html >>>> >>>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05912.html >>>> >>>> Anthony has stated the following; >>>> *"the only reliable answer is use TCP."* >>>> >>>> There was also the option of enabling compact headers; >>>> >>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-December/078633.html >>>> >>>> From what I can tell, there is no way to guarantee this problem won't >>>> happen unless you use TCP. You could reduce the packet size by compacting >>>> headers or removing codecs, but this would be on the assumption that every >>>> hop is running at 1500 MTU. >>>> >>>> Hope this helps! >>>> >>>> Cal >>>> >>>> On Sat, Apr 6, 2013 at 2:28 PM, Ira Tessler wrote: >>>> >>>>> I just need a little guidance with the way presence works. Forgive me >>>>> if I am asking novice questions. >>>>> >>>>> Background (simple version) >>>>> We run Freeswitch in a hosted/cloud environment in a data center. We >>>>> have IP phones in our office on our LAN. >>>>> >>>>> That way I am understanding how Presence works, I am just learning >>>>> this, is that when a BLF button is programmed on a phone, that phone will >>>>> send a "Subscribe" message to Freeswitch. The subscriptions are stored in >>>>> the sip_subscriptions table (i think) in the sofia database for the sip >>>>> profile. When calls come in for that subscription, Freeswitch will send out >>>>> a NOTIFY message to the phone that subscribed in order to change the state >>>>> of the BLF Light. >>>>> >>>>> He is my questions/issue/confusion. >>>>> All our phones use UDP which has a maximum packet size of 1500 bytes. >>>>> When doing a sofia global siptrace on, I notice that most of the NOTIFY >>>>> messages are greater then 1500 bytes. That will cause packet fragmentation. >>>>> So if the NOTIFY message is fragmented, will it get to the phone correctly? >>>>> (all the time, some of the time, never??) >>>>> >>>>> If the the answer is other then ("all the time"), how do I fix this? >>>>> The only solution I can come up with is having my phones use TCP instead of >>>>> UDP. Is that the correctly solution? Did anyone else out there run into >>>>> this issue and if so, what is the "best practice" solution (if there is >>>>> one)? >>>>> >>>>> Thank you in advance! >>>>> >>>>> Ira Tessler >>>>> Lead Software Engineer >>>>> ConnectMe >>>>> (732) 490-9007 x2 >>>>> ira at connectmevoice.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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/5e4cadee/attachment-0001.html From schoch+freeswitch.org at xwin32.com Wed Apr 10 04:23:04 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 9 Apr 2013 17:23:04 -0700 Subject: [Freeswitch-users] loopback In-Reply-To: <1365547526840-7589575.post@n2.nabble.com> References: <1365547526840-7589575.post@n2.nabble.com> Message-ID: On Tue, Apr 9, 2013 at 3:45 PM, Jeff Lenk wrote: > It also allows an attended transfer to voicemail to work. > I get it. If you use application="voicemail", and do an attended transfer, you can login and set it up to play a message, but when you complete the transfer, the first party will go back to the place as if he had dialed voicemail directly. Furthermore, if you get a call, and then conference your voicemail, you can both be listening to your messages, but as soon as you hang up, the called party will get switched to "Please enter your password." Bridging to loopback/app=voicemail solves these problems. Thank you. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130409/7131b66d/attachment.html From lconroy at insensate.co.uk Wed Apr 10 05:09:17 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Wed, 10 Apr 2013 02:09:17 +0100 Subject: [Freeswitch-users] Presence Sanity Check In-Reply-To: References: Message-ID: <4FC40472-8AE5-44D4-A3F8-1553B84F8F8B@insensate.co.uk> Hi Folks, ignition on> You have to be there to get your way. When all this was being nailed down, no-one spoke up. Trust me, I was listening. I was there during this phase -- the SIP crowd who came up with the standards *were* open to well argued ideas, so if you all had been there, you could well have deflected the dumb schemes as they came up. [OK, you'd have argued with a bunch of folk from MoFoCos, who *really* wanted to get SIP working as they know that circuit switched infrastructures just were not going to cut it, and had their own priorities; frankly, they were the main ones speaking and there wasn't anyone else to argue against them]. SIP was always designed as a "single packet per message" scheme. IIRC, there were always two SIP camps -- those who required secure signalling (and thus TCP/SRTP and so on), and those who wanted a really lightweight scheme with low bandwidth overheads [remember, this all came up in the '90s, when bandwidth was NOT cheap]. Drop back to TCP was seen as a rare event, as the SIP main body was tight (very tight, if you used the short forms), and you simply wouldn't have sub-parts that big with *sane* SIP messages so that MTU was going to hurt. If you WERE going to send something that was over the MTU, you'd start with or switch over to TCP -- the assumption being that this latter would only happen in rare situations (even when carrying all kinds of piggy-backed ISUP data). As for messaging -- remember that the IETF was stuck in the grinder between the big IM providers, who were trying to find a common protocol between their networks. This looked like being some IMPP-style mung. In parallel with that, the SIP folks were initially content with MESSAGE; after all, who in their right mind would send an HTML page full of mung, when it was just for display on a phone -- thus we had an SMS style scheme that worked fine, and all was cool. IIRC, MESSAGE (and sending in-signalling channel messages) was knocked up overnight during an IETF meeting, just to point out to the IMPP folk and the "data architects" that it didn't need to be hard if you have a sensible signalling scheme. Then the XMPP stuff arose from the ashes of the IM network interop work (which had crashed and burned fairly spectacularly in the IETF). The XMPP folk had already drunk far too much of the XML cool aid. By this time, SIP had already produced a Presence model that was rapidly getting complex, and choosing to switch over to XML as the one true representation made the SIP sub-parts expand really quickly. At this point, I personally lost interest. Even the names were getting beyond me: I mean, a "presentity" -- had the 400 pound Gorilla who specified this lot not been taught English? In particular, the MoFoCos wanted to be able to provide "rich presence" and something at least as good as IM, which had become fashionable even they had heard of it. As they'd just rearranged their networks to use SIP at the core, that meant that they needed some serious work done. Recall that a lot of the parameters and extra mung in the headers was introduced just for commercial providers (and both 3GPP and 3GPP2 were using SIP to carry everything including the kitchen sink inside the signalling packets -- i.e., SIP, which gave the main body a heck of a bloat, before we start on the sub-parts). If SIP was to be able to carry these kind of bloated HTML-like monstrosities both as headers AND to cover the kind of "rich presence" and IM stuff we all have grown to love, the clean -- single packet -- model started to creak, and the transition from UDP to TCP was no longer going to be a rare event for loons who just couldn't control themselves and hadn't heard of sigcomp. As for the 1500 byte limit (i.e. the typical MTU) for UDP messages, that's a reflection of the original model of single packet messages. Going for a scheme that sends out > MTU messages (i.e. sends fragments out over the net), was and is a hideous kludge, and falls over with a lot of the cheap home routers people seem to use. So ... if you're going to do complex headers and messaging or the 15 million different variants on presence (as they are now, i.e. XML to the hilt), then start out with TCP -- don't even bother trying to use god's own protocol in a session. If you're doing voice and -basic- presence, UDP is fine (and don't forget, sigcomp exists to keep the bandwidth down, particularly between servers). I second the hear hear!!! > > Ira Tessler > Lead Software Engineer > ConnectMe > (732) 490-9007 x2 > ira at connectmevoice.com > > > On Mon, Apr 8, 2013 at 5:50 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hear hear! >> >> On Mon, Apr 8, 2013 at 10:17 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Its actually somewhat ridiculous and one of my personal favorites in >>> terms of RFC smoke and mirrors in SIP to try and cover up a flaw with more >>> specs. They basically say: >>> >>> >>> If the total packet including the sip headers and the payload exceeds the >>> MTU and you are using the UDP transport, you MUST try sending the packet >>> over TCP instead. If that times out or fails then you SHOULD send it over >>> UDP anyway. >>> >>> >>> THEREFORE by virtue of this decree: >>> >>> You MUST implement your sip stack to accept UDP packets of up to 65536 >>> bytes. >>> >>> AND >>> >>> You MUST implement both TCP and UDP transports. >>> >>> >>> So in short, you are not supposed to send anything over udp that exceeds >>> the mtu yet you are required to implement it so its possible. >>> >>> Many stacks, including Asterisk for many of the first half of FS >>> existence, did not implement TCP so with this rule being enforced, the >>> packets would sit there for 2-5 min then give up and change to UDP. How's >>> that for PDD. >>> >>> Anyway, we choose to ignore this rule intentionally and just stick with >>> the negotiated protocol. If you find yourself in this situation the >>> solution is to use TCP. >>> >>> >>> P.S. >>> >>> If they did not use 2k of XML to transmit about 12 bytes worth of useful >>> info regarding the state of the presence, we would not have this problem to >>> begin with ;) >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Mon, Apr 8, 2013 at 4:02 PM, Ira Tessler wrote: >>> >>>> Thank you for your information. It did help! >>>> >>>> Ira >>>> >>>> Ira Tessler >>>> Lead Software Engineer >>>> ConnectMe >>>> (732) 490-9007 x2 >>>> ira at connectmevoice.com >>>> >>>> >>>> On Sun, Apr 7, 2013 at 12:49 PM, Cal Leeming [Simplicity Media Ltd] < >>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>> >>>>> In regards to the UDP fragmentation, this is an extremely good question. >>>>> >>>>> Only yesterday I started to build a simple forwarding SBC in Python >>>>> using UDP sockets, however I came up against the same theoretical problem >>>>> of packets being larger than 1500 bytes. >>>>> >>>>> I've had a read through various documentation; >>>>> http://www.rfc-ref.org/RFC-TEXTS/3261/chapter18.html >>>>> http://www.ietf.org/rfc/rfc3428.txt >>>>> >>>>> https://lists.cs.columbia.edu/pipermail/sip-implementors/2006-August/013857.html >>>>> >>>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-February/068372.html >>>>> >>>>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05912.html >>>>> >>>>> Anthony has stated the following; >>>>> *"the only reliable answer is use TCP."* >>>>> >>>>> There was also the option of enabling compact headers; >>>>> >>>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-December/078633.html >>>>> >>>>> From what I can tell, there is no way to guarantee this problem won't >>>>> happen unless you use TCP. You could reduce the packet size by compacting >>>>> headers or removing codecs, but this would be on the assumption that every >>>>> hop is running at 1500 MTU. >>>>> >>>>> Hope this helps! >>>>> >>>>> Cal >>>>> >>>>> On Sat, Apr 6, 2013 at 2:28 PM, Ira Tessler wrote: >>>>> >>>>>> I just need a little guidance with the way presence works. Forgive me >>>>>> if I am asking novice questions. >>>>>> >>>>>> Background (simple version) >>>>>> We run Freeswitch in a hosted/cloud environment in a data center. We >>>>>> have IP phones in our office on our LAN. >>>>>> >>>>>> That way I am understanding how Presence works, I am just learning >>>>>> this, is that when a BLF button is programmed on a phone, that phone will >>>>>> send a "Subscribe" message to Freeswitch. The subscriptions are stored in >>>>>> the sip_subscriptions table (i think) in the sofia database for the sip >>>>>> profile. When calls come in for that subscription, Freeswitch will send out >>>>>> a NOTIFY message to the phone that subscribed in order to change the state >>>>>> of the BLF Light. >>>>>> >>>>>> He is my questions/issue/confusion. >>>>>> All our phones use UDP which has a maximum packet size of 1500 bytes. >>>>>> When doing a sofia global siptrace on, I notice that most of the NOTIFY >>>>>> messages are greater then 1500 bytes. That will cause packet fragmentation. >>>>>> So if the NOTIFY message is fragmented, will it get to the phone correctly? >>>>>> (all the time, some of the time, never??) >>>>>> >>>>>> If the the answer is other then ("all the time"), how do I fix this? >>>>>> The only solution I can come up with is having my phones use TCP instead of >>>>>> UDP. Is that the correctly solution? Did anyone else out there run into >>>>>> this issue and if so, what is the "best practice" solution (if there is >>>>>> one)? >>>>>> >>>>>> Thank you in advance! >>>>>> >>>>>> Ira Tessler >>>>>> Lead Software Engineer >>>>>> ConnectMe >>>>>> (732) 490-9007 x2 >>>>>> ira at connectmevoice.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 >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Wed Apr 10 05:24:50 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 9 Apr 2013 20:24:50 -0500 Subject: [Freeswitch-users] Presence Sanity Check In-Reply-To: <4FC40472-8AE5-44D4-A3F8-1553B84F8F8B@insensate.co.uk> References: <4FC40472-8AE5-44D4-A3F8-1553B84F8F8B@insensate.co.uk> Message-ID: Indeed! But we have a 2nd chance with WebRTC ... oh wait..... They're doing what???? The secure-only guys you mentioned have come back from exile to seek their revenge! http://www.freeswitch.org/node/437 On Tue, Apr 9, 2013 at 8:09 PM, Lawrence Conroy wrote: > Hi Folks, > > ignition on> > > You have to be there to get your way. When all this was being nailed down, > no-one spoke up. Trust me, I was listening. > I was there during this phase -- the SIP crowd who came up with the > standards *were* open to well argued ideas, so if you all had been there, > you could well have deflected the dumb schemes as they came up. [OK, you'd > have argued with a bunch of folk from MoFoCos, who *really* wanted to get > SIP working as they know that circuit switched infrastructures just were > not going to cut it, and had their own priorities; frankly, they were the > main ones speaking and there wasn't anyone else to argue against them]. > > SIP was always designed as a "single packet per message" scheme. > IIRC, there were always two SIP camps -- those who required secure > signalling (and thus TCP/SRTP and so on), and those who wanted a really > lightweight scheme with low bandwidth overheads [remember, this all came up > in the '90s, when bandwidth was NOT cheap]. > Drop back to TCP was seen as a rare event, as the SIP main body was tight > (very tight, if you used the short forms), and you simply wouldn't have > sub-parts that big with *sane* SIP messages so that MTU was going to hurt. > If you WERE going to send something that was over the MTU, you'd start with > or switch over to TCP -- the assumption being that this latter would only > happen in rare situations (even when carrying all kinds of piggy-backed > ISUP data). > > As for messaging -- remember that the IETF was stuck in the grinder > between the big IM providers, who were trying to find a common protocol > between their networks. > This looked like being some IMPP-style mung. > In parallel with that, the SIP folks were initially content with MESSAGE; > after all, who in their right mind would send an HTML page full of mung, > when it was just for display on a phone -- thus we had an SMS style scheme > that worked fine, and all was cool. IIRC, MESSAGE (and sending > in-signalling channel messages) was knocked up overnight during an IETF > meeting, just to point out to the IMPP folk and the "data architects" that > it didn't need to be hard if you have a sensible signalling scheme. > > Then the XMPP stuff arose from the ashes of the IM network interop work > (which had crashed and burned fairly spectacularly in the IETF). > The XMPP folk had already drunk far too much of the XML cool aid. > > By this time, SIP had already produced a Presence model that was rapidly > getting complex, and choosing to switch over to XML as the one true > representation made the SIP sub-parts expand really quickly. At this point, > I personally lost interest. Even the names were getting beyond me: I mean, > a "presentity" -- had the 400 pound Gorilla who specified this lot not been > taught English? > > In particular, the MoFoCos wanted to be able to provide "rich presence" > and something at least as good as IM, which had become fashionable even > they had heard of it. As they'd just rearranged their networks to use SIP > at the core, that meant that they needed some serious work done. Recall > that a lot of the parameters and extra mung in the headers was introduced > just for commercial providers (and both 3GPP and 3GPP2 were using SIP to > carry everything including the kitchen sink inside the signalling packets > -- i.e., SIP, which gave the main body a heck of a bloat, before we start > on the sub-parts). > If SIP was to be able to carry these kind of bloated HTML-like > monstrosities both as headers AND to cover the kind of "rich presence" and > IM stuff we all have grown to love, the clean -- single packet -- model > started to creak, and the transition from UDP to TCP was no longer going to > be a rare event for loons who just couldn't control themselves and hadn't > heard of sigcomp. > > As for the 1500 byte limit (i.e. the typical MTU) for UDP messages, that's > a reflection of the original model of single packet messages. Going for a > scheme that sends out > MTU messages (i.e. sends fragments out over the > net), was and is a hideous kludge, and falls over with a lot of the cheap > home routers people seem to use. > > So ... if you're going to do complex headers and messaging or the 15 > million different variants on presence (as they are now, i.e. XML to the > hilt), then start out with TCP -- don't even bother trying to use god's own > protocol in a session. > If you're doing voice and -basic- presence, UDP is fine (and don't forget, > sigcomp exists to keep the bandwidth down, particularly between servers). > > all the best, > Lawrence > > On 10 Apr 2013, at 00:48, Ira Tessler wrote: > > I second the hear hear!!! > > > > Ira Tessler > > Lead Software Engineer > > ConnectMe > > (732) 490-9007 x2 > > ira at connectmevoice.com > > > > > > On Mon, Apr 8, 2013 at 5:50 PM, Cal Leeming [Simplicity Media Ltd] < > > cal.leeming at simplicitymedialtd.co.uk> wrote: > > > >> Hear hear! > >> > >> On Mon, Apr 8, 2013 at 10:17 PM, Anthony Minessale < > >> anthony.minessale at gmail.com> wrote: > >> > >>> Its actually somewhat ridiculous and one of my personal favorites in > >>> terms of RFC smoke and mirrors in SIP to try and cover up a flaw with > more > >>> specs. They basically say: > >>> > >>> > >>> If the total packet including the sip headers and the payload exceeds > the > >>> MTU and you are using the UDP transport, you MUST try sending the > packet > >>> over TCP instead. If that times out or fails then you SHOULD send it > over > >>> UDP anyway. > >>> > >>> > >>> THEREFORE by virtue of this decree: > >>> > >>> You MUST implement your sip stack to accept UDP packets of up to 65536 > >>> bytes. > >>> > >>> AND > >>> > >>> You MUST implement both TCP and UDP transports. > >>> > >>> > >>> So in short, you are not supposed to send anything over udp that > exceeds > >>> the mtu yet you are required to implement it so its possible. > >>> > >>> Many stacks, including Asterisk for many of the first half of FS > >>> existence, did not implement TCP so with this rule being enforced, the > >>> packets would sit there for 2-5 min then give up and change to UDP. > How's > >>> that for PDD. > >>> > >>> Anyway, we choose to ignore this rule intentionally and just stick with > >>> the negotiated protocol. If you find yourself in this situation the > >>> solution is to use TCP. > >>> > >>> > >>> P.S. > >>> > >>> If they did not use 2k of XML to transmit about 12 bytes worth of > useful > >>> info regarding the state of the presence, we would not have this > problem to > >>> begin with ;) > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> On Mon, Apr 8, 2013 at 4:02 PM, Ira Tessler >wrote: > >>> > >>>> Thank you for your information. It did help! > >>>> > >>>> Ira > >>>> > >>>> Ira Tessler > >>>> Lead Software Engineer > >>>> ConnectMe > >>>> (732) 490-9007 x2 > >>>> ira at connectmevoice.com > >>>> > >>>> > >>>> On Sun, Apr 7, 2013 at 12:49 PM, Cal Leeming [Simplicity Media Ltd] < > >>>> cal.leeming at simplicitymedialtd.co.uk> wrote: > >>>> > >>>>> In regards to the UDP fragmentation, this is an extremely good > question. > >>>>> > >>>>> Only yesterday I started to build a simple forwarding SBC in Python > >>>>> using UDP sockets, however I came up against the same theoretical > problem > >>>>> of packets being larger than 1500 bytes. > >>>>> > >>>>> I've had a read through various documentation; > >>>>> http://www.rfc-ref.org/RFC-TEXTS/3261/chapter18.html > >>>>> http://www.ietf.org/rfc/rfc3428.txt > >>>>> > >>>>> > https://lists.cs.columbia.edu/pipermail/sip-implementors/2006-August/013857.html > >>>>> > >>>>> > http://lists.freeswitch.org/pipermail/freeswitch-users/2011-February/068372.html > >>>>> > >>>>> > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05912.html > >>>>> > >>>>> Anthony has stated the following; > >>>>> *"the only reliable answer is use TCP."* > >>>>> > >>>>> There was also the option of enabling compact headers; > >>>>> > >>>>> > http://lists.freeswitch.org/pipermail/freeswitch-users/2011-December/078633.html > >>>>> > >>>>> From what I can tell, there is no way to guarantee this problem won't > >>>>> happen unless you use TCP. You could reduce the packet size by > compacting > >>>>> headers or removing codecs, but this would be on the assumption that > every > >>>>> hop is running at 1500 MTU. > >>>>> > >>>>> Hope this helps! > >>>>> > >>>>> Cal > >>>>> > >>>>> On Sat, Apr 6, 2013 at 2:28 PM, Ira Tessler >wrote: > >>>>> > >>>>>> I just need a little guidance with the way presence works. Forgive > me > >>>>>> if I am asking novice questions. > >>>>>> > >>>>>> Background (simple version) > >>>>>> We run Freeswitch in a hosted/cloud environment in a data center. We > >>>>>> have IP phones in our office on our LAN. > >>>>>> > >>>>>> That way I am understanding how Presence works, I am just learning > >>>>>> this, is that when a BLF button is programmed on a phone, that > phone will > >>>>>> send a "Subscribe" message to Freeswitch. The subscriptions are > stored in > >>>>>> the sip_subscriptions table (i think) in the sofia database for the > sip > >>>>>> profile. When calls come in for that subscription, Freeswitch will > send out > >>>>>> a NOTIFY message to the phone that subscribed in order to change > the state > >>>>>> of the BLF Light. > >>>>>> > >>>>>> He is my questions/issue/confusion. > >>>>>> All our phones use UDP which has a maximum packet size of 1500 > bytes. > >>>>>> When doing a sofia global siptrace on, I notice that most of the > NOTIFY > >>>>>> messages are greater then 1500 bytes. That will cause packet > fragmentation. > >>>>>> So if the NOTIFY message is fragmented, will it get to the phone > correctly? > >>>>>> (all the time, some of the time, never??) > >>>>>> > >>>>>> If the the answer is other then ("all the time"), how do I fix this? > >>>>>> The only solution I can come up with is having my phones use TCP > instead of > >>>>>> UDP. Is that the correctly solution? Did anyone else out there run > into > >>>>>> this issue and if so, what is the "best practice" solution (if > there is > >>>>>> one)? > >>>>>> > >>>>>> Thank you in advance! > >>>>>> > >>>>>> Ira Tessler > >>>>>> Lead Software Engineer > >>>>>> ConnectMe > >>>>>> (732) 490-9007 x2 > >>>>>> ira at connectmevoice.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 > >>>> > >>>> > >>> > >>> > >>> -- > >>> Anthony Minessale II > >>> > >>> FreeSWITCH http://www.freeswitch.org/ > >>> ClueCon http://www.cluecon.com/ > >>> Twitter: http://twitter.com/FreeSWITCH_wire > >>> > >>> AIM: anthm > >>> MSN:anthony_minessale at 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 > >> > >> > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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/20130409/3207c7f5/attachment-0001.html From nneul at mst.edu Wed Apr 10 05:54:16 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 09 Apr 2013 20:54:16 -0500 Subject: [Freeswitch-users] annoying jira automatic actions week or two after tickets closed Message-ID: <5164C648.7000305@mst.edu> Seeing messages about tickets being "closed" automatically, but they are just taking it from a resolved state back to unresolved, making it show back up in my jira dashboard. Can y'all look into the rules you have defined for those automatic actions? ---------------- Due to a long period of inactivity (13 or more days), this issue is due to be automatically close within 24 hours. If this issue is not actually resolved, please reopen it and add appropriate comments to help developers fix the issue. Thanks, Jira Admin ----------------- -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From mnrao2001 at gmail.com Wed Apr 10 08:05:02 2013 From: mnrao2001 at gmail.com (Nageshwara Rao Moova) Date: Wed, 10 Apr 2013 09:35:02 +0530 Subject: [Freeswitch-users] Freetdm Channel utilization in order(ascending/descending) In-Reply-To: References: Message-ID: i see you have listed span 1channels in span2 & vice-versa change it appropriately On Apr 9, 2013 4:41 PM, "Vishal Kakkar" wrote: > [span wanpipe wp1] > trunk_type => e1 > group=dialGrp1 > b-channel => 2:1-15 > b-channel => 2:17-31 > d-channel => 2:16 > > [span wanpipe wp2] > trunk_type => e1 > group=dialGrp1 > b-channel => 1:1-15 > b-channel => 1:17-31 > d-channel => 1:16 > > > freetdm/dialGrp1/*a*/xxxxx initiates call from channel 1 of span 2 i.e. > (2:1) > > freetdm/dialGrp1/*A/*xxxxx initiates call from channel 31 of span 2 i.e. > (2:31).. but i was expecting it to be 31st channel of Span1. > > Also 2nd question in above config all calls are going through wp1(span2) > only.. even if i initiate more than 30 calls simultaneously.. no call goes > to wp2(span 1) > > Can anyone please help how can i make ascending/descending channel > utilization using one single group across both above spans.. > > Thanks a lot.. > FS Rocks. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/43365902/attachment.html From ashish at nms.co.in Wed Apr 10 10:09:06 2013 From: ashish at nms.co.in (Ashish gautam) Date: Wed, 10 Apr 2013 11:39:06 +0530 Subject: [Freeswitch-users] Default SIP users not registering Message-ID: Hi, I am trying to register Default SIP users i.e. 1019,1004 etc from a softphone but they are not registering. Sofia status also not showing any user registrations. FreeSWITCH is listening on port 5060 and every configuration has been done according to the docs. What could be the issue? Any help appreciated. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/9be5c278/attachment.html From miha at softnet.si Wed Apr 10 10:22:19 2013 From: miha at softnet.si (Miha) Date: Wed, 10 Apr 2013 08:22:19 +0200 Subject: [Freeswitch-users] Default SIP users not registering In-Reply-To: References: Message-ID: <5165051B.5050906@softnet.si> Hi, was does sip trace say. You must add user to default directory and that do reloadxml. You can past you log to see what is going on.. miha Dne 4/10/2013 8:09 AM, pis(e Ashish gautam: > Hi, > > I am trying to register Default SIP users i.e. 1019,1004 etc from a > softphone but they are not registering. Sofia status also not showing > any user registrations. FreeSWITCH is listening on port 5060 and every > configuration has been done according to the docs. What could be the > issue? > > Any help appreciated. > > Thanks in advance. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/ad1df924/attachment.html From ashish at nms.co.in Wed Apr 10 10:28:51 2013 From: ashish at nms.co.in (Ashish gautam) Date: Wed, 10 Apr 2013 11:58:51 +0530 Subject: [Freeswitch-users] Default SIP users not registering In-Reply-To: <5165051B.5050906@softnet.si> References: <5165051B.5050906@softnet.si> Message-ID: Thanks Miha, sip trace shows nothing. On Wed, Apr 10, 2013 at 11:52 AM, Miha wrote: > > Hi, > > was does sip trace say. You must add user to default directory and that do > reloadxml. > > You can past you log to see what is going on.. > > miha > > Dne 4/10/2013 8:09 AM, pi?e Ashish gautam: > > Hi, > > I am trying to register Default SIP users i.e. 1019,1004 etc from a > softphone but they are not registering. Sofia status also not showing any > user registrations. FreeSWITCH is listening on port 5060 and every > configuration has been done according to the docs. What could be the issue? > > Any help appreciated. > > Thanks in advance. > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/92645d5e/attachment.html From miha at softnet.si Wed Apr 10 10:54:26 2013 From: miha at softnet.si (Miha) Date: Wed, 10 Apr 2013 08:54:26 +0200 Subject: [Freeswitch-users] Default SIP users not registering In-Reply-To: References: <5165051B.5050906@softnet.si> Message-ID: <51650CA2.8090106@softnet.si> did you try to use whireshark? miha Dne 4/10/2013 8:28 AM, pis(e Ashish gautam: > Thanks Miha, > > sip trace shows nothing. > > > On Wed, Apr 10, 2013 at 11:52 AM, Miha > wrote: > > > Hi, > > was does sip trace say. You must add user to default directory and > that do reloadxml. > > You can past you log to see what is going on.. > > miha > > Dne 4/10/2013 8:09 AM, pis(e Ashish gautam: >> Hi, >> >> I am trying to register Default SIP users i.e. 1019,1004 etc from >> a softphone but they are not registering. Sofia status also not >> showing any user registrations. FreeSWITCH is listening on port >> 5060 and every configuration has been done according to the docs. >> What could be the issue? >> >> Any help appreciated. >> >> Thanks in advance. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20130410/b78dba03/attachment-0001.html From ml88888 at hotmail.com Wed Apr 10 10:51:07 2013 From: ml88888 at hotmail.com (FSX) Date: Tue, 9 Apr 2013 23:51:07 -0700 (PDT) Subject: [Freeswitch-users] Build errors in 2010.express solution In-Reply-To: <1365424893457-7589513.post@n2.nabble.com> References: <1365285612276-7589476.post@n2.nabble.com> <1365424893457-7589513.post@n2.nabble.com> Message-ID: <1365576667168-7589585.post@n2.nabble.com> What exactly do you mean "fresh clone"? Last time I build this solution without any problems was two weeks ago. Yesterday I did "Clean Solution" and build.Got a lot of fatal errors. I applied "Rebuild Solution", got the same result. Today I've pulled latest git "c9a49b0", I've removed solution files and pulled them from git repository back, I've run "git status" to make sure that I have latest and correct sources. I've repeated "Clean Solution" again and build it twice to get this bottom line: ========== Build: 21 succeeded, 72 failed, 53 up-to-date, 18 skipped ========== The only thing I do with solution properties configuration is I turn off project "mod_managed", Debug_CLR configuration, Win32 platform, because I always getting errors building it. The rest is default. And, as I said, I build it successfully for years. But now (actually a week ago or so) something becomes suddenly wrong and I'm getting lot of fatal errors... The first fatal error is this: LINK : fatal error LNK1181: cannot open input file 'C:\Dev\FS\Win32\Debug\FreeSwitchCore.lib' It's because when VS tried to make FreeSwitchCoreLib, it failed: ------ Build started: Project: FreeSwitchCoreLib, Configuration: Debug Win32 ------ Generating switch_version.inc Checking if we're building a newer git version switch_rtp.c ..\..\src\switch_rtp.c(2270): error C2220: warning treated as error - no 'object' file generated ..\..\src\switch_rtp.c(2270): warning C4013: 'SSL_CTX_set_tlsext_use_srtp' undefined; assuming extern returning int ------ Build started: Project: iksemel, Configuration: Debug Win32 ------ ... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Build-errors-in-2010-express-solution-tp7589476p7589585.html Sent from the freeswitch-users mailing list archive at Nabble.com. From marwan.idriss at gmail.com Wed Apr 10 10:56:26 2013 From: marwan.idriss at gmail.com (Marwan Idriss) Date: Wed, 10 Apr 2013 09:56:26 +0300 Subject: [Freeswitch-users] Use java nio (netty) library to get fd (FileDescriptor) Message-ID: Greeting, How I can use java nio (netty) library mentioned in http://wiki.freeswitch.org/wiki/Java_ESL_Client to get fd (FileDescriptor), that is need to make new connection to FreeSwitch Regards Marwan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/db6a158a/attachment.html From marwan.idriss at gmail.com Wed Apr 10 11:00:56 2013 From: marwan.idriss at gmail.com (Marwan Idriss) Date: Wed, 10 Apr 2013 10:00:56 +0300 Subject: [Freeswitch-users] Use Java to get fd that can be used in ESLconnection con = new ESLconnection(fd); Message-ID: Greeting, In esl.jar I can use ESLconnection con = new ESLconnection(fd); to make connection for outbound freeswith socket; but How I can fd ? , this is the Question Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/f09e6282/attachment.html From marwan.idriss at gmail.com Wed Apr 10 11:07:36 2013 From: marwan.idriss at gmail.com (Marwan Idriss) Date: Wed, 10 Apr 2013 10:07:36 +0300 Subject: [Freeswitch-users] http://wiki.freeswitch.org/wiki/Java_ESL_Client, how I can make a connection before starting to handle Outbound connection ? Message-ID: Greeting, bellow is an example to handle outbound connection and make hung up: but it never show How to make the connection in first place is their an example in main of how this function work ? /* * Copyright 2010 david varnes. * * Licensed under the Apache License, version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at: * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. */ package org.freeswitch.esl.client.outbound.example; import org.freeswitch.esl.client.outbound.AbstractOutboundClientHandler; import org.freeswitch.esl.client.transport.SendMsg; import org.freeswitch.esl.client.transport.event.EslEvent; import org.freeswitch.esl.client.transport.message.EslMessage; import org.freeswitch.esl.client.transport.message.EslHeaders.Name; import org.jboss.netty.channel.Channel; import org.jboss.netty.channel.ChannelHandlerContext; /** * Simple example of a handler for outbound connection from FreeSWITCH server. * This class will log some of the FreeSWTICH call channel variables and * then hangup the call. * * @author david varnes */ public class SimpleHangupOutboundHandler extends AbstractOutboundClientHandler { @Override protected void handleConnectResponse( ChannelHandlerContext ctx, EslEvent event ) { log.info( "Received connect response [{}]", event ); if ( event.getEventName().equalsIgnoreCase( "CHANNEL_DATA" ) ) { // this is the response to the initial connect log.info( "======================= incoming channel data =============================" ); log.info( "Event-Date-Local: [{}]", event.getEventDateLocal() ); log.info( "Unique-ID: [{}]", event.getEventHeaders().get( "Unique-ID" ) ); log.info( "Channel-ANI: [{}]", event.getEventHeaders().get( "Channel-ANI" ) ); log.info( "Answer-State: [{}]", event.getEventHeaders().get( "Answer-State" ) ); log.info( "Caller-Destination-Number: [{}]", event.getEventHeaders().get( "Caller-Destination-Number" ) ); log.info( "======================= = = = = = = = = = = = =============================" ); // now hangup the call hangupCall( ctx.getChannel() ); } else { throw new IllegalStateException( "Unexpected event after connect: [" + event.getEventName() + ']' ); } } @Override protected void handleEslEvent( ChannelHandlerContext ctx, EslEvent event ) { log.info( "Received event [{}]", event ); } private void hangupCall( Channel channel ) { SendMsg hangupMsg = new SendMsg(); hangupMsg.addCallCommand( "execute" ); hangupMsg.addExecuteAppName( "hangup" ); EslMessage response = sendSyncMultiLineCommand( channel, hangupMsg.getMsgLines() ); if ( response.getHeaderValue( Name.REPLY_TEXT ).startsWith( "+OK" ) ) { log.info( "Call hangup successful" ); } else { log.error( "Call hangup failed: [{}}", response.getHeaderValue( Name.REPLY_TEXT ) ); } } } -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/ffa5d4b7/attachment.html From marwan.idriss at gmail.com Wed Apr 10 11:19:22 2013 From: marwan.idriss at gmail.com (Marwan Idriss) Date: Wed, 10 Apr 2013 10:19:22 +0300 Subject: [Freeswitch-users] public ESLconnection(int socket) of esl.jar Message-ID: In esl.jar library , their is a public function to make make outbound connection with freeswitch : public ESLconnection(int socket) { this(eslJNI.new_ESLconnection__SWIG_2(socket), true); } but how I can (int socket) ? Regards Marwan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/86d1a24c/attachment.html From marwan.idriss at gmail.com Wed Apr 10 11:23:41 2013 From: marwan.idriss at gmail.com (Marwan Idriss) Date: Wed, 10 Apr 2013 10:23:41 +0300 Subject: [Freeswitch-users] Javadoc for esl.jar Message-ID: Greeting, Is their any Javadoc for esl.jar ? Regards Marwan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/4a5f7614/attachment.html From ashish at nms.co.in Wed Apr 10 13:07:35 2013 From: ashish at nms.co.in (Ashish gautam) Date: Wed, 10 Apr 2013 14:37:35 +0530 Subject: [Freeswitch-users] Default SIP users not registering In-Reply-To: <51650CA2.8090106@softnet.si> References: <5165051B.5050906@softnet.si> <51650CA2.8090106@softnet.si> Message-ID: Miha, This is the output of the tcpdump -nq -s 0 -A -vvv -i eth0 port 5060 for the related packet : 13:29:37.050278 IP (tos 0x0, ttl 128, id 9943, offset 0, flags [none], proto UDP (17), length 640) 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 612 E...&....... ... ........l..SUBSCRIBE sip:1100 at 10.1.30.229 SIP/2.0 Via: SIP/2.0/UDP 10.1.30.210:53652 ;rport;branch=z9hG4bKPj91db66336d564de5b3b3e50c03ac32b3 Max-Forwards: 70 From: "Ashish" ;tag=3058239a404c4a4dbcd438916aa8033b To: Contact: Call-ID: b3d45216211242afa13ae86734ea4408 CSeq: 30541 SUBSCRIBE Event: message-summary Expires: 3600 Accept: application/simple-message-summary Allow-Events: conference, message-summary, dialog, presence, presence.winfo, xcap-diff, dialog.winfo, refer User-Agent: Blink 0.2.10 (Windows) Content-Length: 0 On Wed, Apr 10, 2013 at 12:24 PM, Miha wrote: > did you try to use whireshark? > > miha > > Dne 4/10/2013 8:28 AM, pi?e Ashish gautam: > > Thanks Miha, > > sip trace shows nothing. > > > On Wed, Apr 10, 2013 at 11:52 AM, Miha wrote: > >> >> Hi, >> >> was does sip trace say. You must add user to default directory and that >> do reloadxml. >> >> You can past you log to see what is going on.. >> >> miha >> >> Dne 4/10/2013 8:09 AM, pi?e Ashish gautam: >> >> Hi, >> >> I am trying to register Default SIP users i.e. 1019,1004 etc from a >> softphone but they are not registering. Sofia status also not showing any >> user registrations. FreeSWITCH is listening on port 5060 and every >> configuration has been done according to the docs. What could be the issue? >> >> Any help appreciated. >> >> Thanks in advance. >> >> >> _________________________________________________________________________ >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/533632a4/attachment-0001.html From miha at softnet.si Wed Apr 10 13:44:23 2013 From: miha at softnet.si (Miha) Date: Wed, 10 Apr 2013 11:44:23 +0200 Subject: [Freeswitch-users] Default SIP users not registering In-Reply-To: References: <5165051B.5050906@softnet.si> <51650CA2.8090106@softnet.si> Message-ID: <51653477.5050407@softnet.si> Hi, what about registration packet and etc? this is only subscribe for presence. miha Dne 4/10/2013 11:07 AM, pis(e Ashish gautam: > Miha, > > This is the output of the tcpdump -nq -s 0 -A -vvv -i eth0 port 5060 > for the related packet : > > 13:29:37.050278 IP (tos 0x0, ttl 128, id 9943, offset 0, flags [none], > proto UDP (17), length 640) > 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 612 > E...&....... > ... > ........l..SUBSCRIBE sip:1100 at 10.1.30.229 > SIP/2.0 > Via: SIP/2.0/UDP > 10.1.30.210:53652;rport;branch=z9hG4bKPj91db66336d564de5b3b3e50c03ac32b3 > Max-Forwards: 70 > From: "Ashish" >;tag=3058239a404c4a4dbcd438916aa8033b > To: > > Contact: > > Call-ID: b3d45216211242afa13ae86734ea4408 > CSeq: 30541 SUBSCRIBE > Event: message-summary > Expires: 3600 > Accept: application/simple-message-summary > Allow-Events: conference, message-summary, dialog, presence, > presence.winfo, xcap-diff, dialog.winfo, refer > User-Agent: Blink 0.2.10 (Windows) > Content-Length: 0 > > > > > On Wed, Apr 10, 2013 at 12:24 PM, Miha > wrote: > > did you try to use whireshark? > > miha > > Dne 4/10/2013 8:28 AM, pis(e Ashish gautam: >> Thanks Miha, >> >> sip trace shows nothing. >> >> >> On Wed, Apr 10, 2013 at 11:52 AM, Miha > > wrote: >> >> >> Hi, >> >> was does sip trace say. You must add user to default >> directory and that do reloadxml. >> >> You can past you log to see what is going on.. >> >> miha >> >> Dne 4/10/2013 8:09 AM, pis(e Ashish gautam: >>> Hi, >>> >>> I am trying to register Default SIP users i.e. 1019,1004 etc >>> from a softphone but they are not registering. Sofia status >>> also not showing any user registrations. FreeSWITCH is >>> listening on port 5060 and every configuration has been done >>> according to the docs. What could be the issue? >>> >>> Any help appreciated. >>> >>> Thanks in advance. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20130410/41974c6f/attachment.html From 4orbit at gmail.com Wed Apr 10 13:44:52 2013 From: 4orbit at gmail.com (Sergey Zhuravlov) Date: Wed, 10 Apr 2013 13:44:52 +0400 Subject: [Freeswitch-users] [ANN] CID Lookup and Fat Free CRM Message-ID: Hey Guys, We are excited to announce that Fat Free CRM is now available to be used with FreeSWITCH's mod_cidkookup. Solution based on standart mod_cidlookup and also may be used with Asterisk. FFCRM -- lightweight CRM used Ruby on Rails. http://www.fatfreecrm.com/ An open source CRM designed to be highly customizable; elegant in simplicity. Yet doesn't have any integration with PBX. Perfect system, especially for Rails' funs. Source code https://github.com/4orbit/ffcrmcidlookup and https://github.com/4orbit/fat_free_crm/tree/cidlookup See short article in Russian http://ru.it.ntcom.lv/blog/cid-lookup-ffcrm The future plans of closer integration with the FreeSWITCH: - wake up the customer's card for an incoming call - "smart" routing call to manager Supposed to use websocket! Supporters are welcomed -- WBR, Sergey GTALK/JABBER:4orbit at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/1d1fb5d2/attachment.html From ashish at nms.co.in Wed Apr 10 14:00:16 2013 From: ashish at nms.co.in (Ashish gautam) Date: Wed, 10 Apr 2013 15:30:16 +0530 Subject: [Freeswitch-users] Default SIP users not registering In-Reply-To: <51653477.5050407@softnet.si> References: <5165051B.5050906@softnet.si> <51650CA2.8090106@softnet.si> <51653477.5050407@softnet.si> Message-ID: Hi Miha, This is the registration packet: 15:29:25.509003 IP (tos 0x0, ttl 128, id 31452, offset 0, flags [none], proto UDP (17), length 525) 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 497 z.....lK ... .........8%REGISTER sip:10.1.30.229 SIP/2.0 Via: SIP/2.0/UDP 10.1.30.210:53652 ;rport;branch=z9hG4bKPj9ccb691852ca419ea305c4cdd79aba44 Max-Forwards: 70 From: "Ashish" ;tag=5970e44547c642a2b6e335b5bbee73ea To: "Ashish" Contact: ;+sip.instance="" Call-ID: a88024e2bdd04101a57f713d57af681e CSeq: 1 REGISTER Expires: 3600 User-Agent: Blink 0.2.10 (Windows) Content-Length: 0 On Wed, Apr 10, 2013 at 3:14 PM, Miha wrote: > Hi, > > what about registration packet and etc? > > this is only subscribe for presence. > > miha > > Dne 4/10/2013 11:07 AM, pi?e Ashish gautam: > > Miha, > > This is the output of the tcpdump -nq -s 0 -A -vvv -i eth0 port 5060 > for the related packet : > > 13:29:37.050278 IP (tos 0x0, ttl 128, id 9943, offset 0, flags [none], > proto UDP (17), length 640) > 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 612 > E...&....... > ... > ........l..SUBSCRIBE sip:1100 at 10.1.30.229 SIP/2.0 > Via: SIP/2.0/UDP 10.1.30.210:53652 > ;rport;branch=z9hG4bKPj91db66336d564de5b3b3e50c03ac32b3 > Max-Forwards: 70 > From: "Ashish" ;tag=3058239a404c4a4dbcd438916aa8033b > To: > Contact: > Call-ID: b3d45216211242afa13ae86734ea4408 > CSeq: 30541 SUBSCRIBE > Event: message-summary > Expires: 3600 > Accept: application/simple-message-summary > Allow-Events: conference, message-summary, dialog, presence, > presence.winfo, xcap-diff, dialog.winfo, refer > User-Agent: Blink 0.2.10 (Windows) > Content-Length: 0 > > > > > On Wed, Apr 10, 2013 at 12:24 PM, Miha wrote: > >> did you try to use whireshark? >> >> miha >> >> Dne 4/10/2013 8:28 AM, pi?e Ashish gautam: >> >> Thanks Miha, >> >> sip trace shows nothing. >> >> >> On Wed, Apr 10, 2013 at 11:52 AM, Miha wrote: >> >>> >>> Hi, >>> >>> was does sip trace say. You must add user to default directory and that >>> do reloadxml. >>> >>> You can past you log to see what is going on.. >>> >>> miha >>> >>> Dne 4/10/2013 8:09 AM, pi?e Ashish gautam: >>> >>> Hi, >>> >>> I am trying to register Default SIP users i.e. 1019,1004 etc from a >>> softphone but they are not registering. Sofia status also not showing any >>> user registrations. FreeSWITCH is listening on port 5060 and every >>> configuration has been done according to the docs. What could be the issue? >>> >>> Any help appreciated. >>> >>> Thanks in advance. >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/2210ed81/attachment-0001.html From miha at softnet.si Wed Apr 10 14:21:16 2013 From: miha at softnet.si (Miha) Date: Wed, 10 Apr 2013 12:21:16 +0200 Subject: [Freeswitch-users] Default SIP users not registering In-Reply-To: References: <5165051B.5050906@softnet.si> <51650CA2.8090106@softnet.si> <51653477.5050407@softnet.si> Message-ID: <51653D1C.1040604@softnet.si> OK, just must post whole sip trace. This is just registration packet from blink/uac, I can not see what FS replied on this. miha Dne 4/10/2013 12:00 PM, pis(e Ashish gautam: > Hi Miha, > > This is the registration packet: > > 15:29:25.509003 IP (tos 0x0, ttl 128, id 31452, offset 0, flags > [none], proto UDP (17), length 525) > 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 497 > z.....lK > ... > .........8%REGISTER sip:10.1.30.229 SIP/2.0 > Via: SIP/2.0/UDP > 10.1.30.210:53652;rport;branch=z9hG4bKPj9ccb691852ca419ea305c4cdd79aba44 > Max-Forwards: 70 > From: "Ashish" >;tag=5970e44547c642a2b6e335b5bbee73ea > To: "Ashish" > > Contact: >;+sip.instance="" > Call-ID: a88024e2bdd04101a57f713d57af681e > CSeq: 1 REGISTER > Expires: 3600 > User-Agent: Blink 0.2.10 (Windows) > Content-Length: 0 > > > On Wed, Apr 10, 2013 at 3:14 PM, Miha > wrote: > > Hi, > > what about registration packet and etc? > > this is only subscribe for presence. > > miha > > Dne 4/10/2013 11:07 AM, pis(e Ashish gautam: >> Miha, >> >> This is the output of the tcpdump -nq -s 0 -A -vvv -i eth0 port >> 5060 for the related packet : >> >> 13:29:37.050278 IP (tos 0x0, ttl 128, id 9943, offset 0, flags >> [none], proto UDP (17), length 640) >> 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 612 >> E...&....... >> ... >> ........l..SUBSCRIBE sip:1100 at 10.1.30.229 >> SIP/2.0 >> Via: SIP/2.0/UDP >> 10.1.30.210:53652;rport;branch=z9hG4bKPj91db66336d564de5b3b3e50c03ac32b3 >> Max-Forwards: 70 >> From: "Ashish" > >;tag=3058239a404c4a4dbcd438916aa8033b >> To: > >> Contact: > > >> Call-ID: b3d45216211242afa13ae86734ea4408 >> CSeq: 30541 SUBSCRIBE >> Event: message-summary >> Expires: 3600 >> Accept: application/simple-message-summary >> Allow-Events: conference, message-summary, dialog, presence, >> presence.winfo, xcap-diff, dialog.winfo, refer >> User-Agent: Blink 0.2.10 (Windows) >> Content-Length: 0 >> >> >> >> >> On Wed, Apr 10, 2013 at 12:24 PM, Miha > > wrote: >> >> did you try to use whireshark? >> >> miha >> >> Dne 4/10/2013 8:28 AM, pis(e Ashish gautam: >>> Thanks Miha, >>> >>> sip trace shows nothing. >>> >>> >>> On Wed, Apr 10, 2013 at 11:52 AM, Miha >> > wrote: >>> >>> >>> Hi, >>> >>> was does sip trace say. You must add user to default >>> directory and that do reloadxml. >>> >>> You can past you log to see what is going on.. >>> >>> miha >>> >>> Dne 4/10/2013 8:09 AM, pis(e Ashish gautam: >>>> Hi, >>>> >>>> I am trying to register Default SIP users i.e. >>>> 1019,1004 etc from a softphone but they are not >>>> registering. Sofia status also not showing any user >>>> registrations. FreeSWITCH is listening on port 5060 and >>>> every configuration has been done according to the >>>> docs. What could be the issue? >>>> >>>> Any help appreciated. >>>> >>>> Thanks in advance. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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/20130410/099b31cc/attachment-0001.html From ashish at nms.co.in Wed Apr 10 14:28:30 2013 From: ashish at nms.co.in (Ashish gautam) Date: Wed, 10 Apr 2013 15:58:30 +0530 Subject: [Freeswitch-users] Default SIP users not registering In-Reply-To: <51653D1C.1040604@softnet.si> References: <5165051B.5050906@softnet.si> <51650CA2.8090106@softnet.si> <51653477.5050407@softnet.si> <51653D1C.1040604@softnet.si> Message-ID: FS is not replying anything there are just subscription and registration packets. Thanks. On Wed, Apr 10, 2013 at 3:51 PM, Miha wrote: > OK, just must post whole sip trace. This is just registration packet > from blink/uac, I can not see what FS replied on this. > > miha > > Dne 4/10/2013 12:00 PM, pi?e Ashish gautam: > > Hi Miha, > > This is the registration packet: > > 15:29:25.509003 IP (tos 0x0, ttl 128, id 31452, offset 0, flags [none], > proto UDP (17), length 525) > 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 497 > z.....lK > ... > .........8%REGISTER sip:10.1.30.229 SIP/2.0 > Via: SIP/2.0/UDP 10.1.30.210:53652 > ;rport;branch=z9hG4bKPj9ccb691852ca419ea305c4cdd79aba44 > Max-Forwards: 70 > From: "Ashish" ;tag=5970e44547c642a2b6e335b5bbee73ea > To: "Ashish" > Contact: >;+sip.instance="" > Call-ID: a88024e2bdd04101a57f713d57af681e > CSeq: 1 REGISTER > Expires: 3600 > User-Agent: Blink 0.2.10 (Windows) > Content-Length: 0 > > > On Wed, Apr 10, 2013 at 3:14 PM, Miha wrote: > >> Hi, >> >> what about registration packet and etc? >> >> this is only subscribe for presence. >> >> miha >> >> Dne 4/10/2013 11:07 AM, pi?e Ashish gautam: >> >> Miha, >> >> This is the output of the tcpdump -nq -s 0 -A -vvv -i eth0 port 5060 >> for the related packet : >> >> 13:29:37.050278 IP (tos 0x0, ttl 128, id 9943, offset 0, flags [none], >> proto UDP (17), length 640) >> 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 612 >> E...&....... >> ... >> ........l..SUBSCRIBE sip:1100 at 10.1.30.229 SIP/2.0 >> Via: SIP/2.0/UDP 10.1.30.210:53652 >> ;rport;branch=z9hG4bKPj91db66336d564de5b3b3e50c03ac32b3 >> Max-Forwards: 70 >> From: "Ashish" > >;tag=3058239a404c4a4dbcd438916aa8033b >> To: >> Contact: >> Call-ID: b3d45216211242afa13ae86734ea4408 >> CSeq: 30541 SUBSCRIBE >> Event: message-summary >> Expires: 3600 >> Accept: application/simple-message-summary >> Allow-Events: conference, message-summary, dialog, presence, >> presence.winfo, xcap-diff, dialog.winfo, refer >> User-Agent: Blink 0.2.10 (Windows) >> Content-Length: 0 >> >> >> >> >> On Wed, Apr 10, 2013 at 12:24 PM, Miha wrote: >> >>> did you try to use whireshark? >>> >>> miha >>> >>> Dne 4/10/2013 8:28 AM, pi?e Ashish gautam: >>> >>> Thanks Miha, >>> >>> sip trace shows nothing. >>> >>> >>> On Wed, Apr 10, 2013 at 11:52 AM, Miha wrote: >>> >>>> >>>> Hi, >>>> >>>> was does sip trace say. You must add user to default directory and that >>>> do reloadxml. >>>> >>>> You can past you log to see what is going on.. >>>> >>>> miha >>>> >>>> Dne 4/10/2013 8:09 AM, pi?e Ashish gautam: >>>> >>>> Hi, >>>> >>>> I am trying to register Default SIP users i.e. 1019,1004 etc from a >>>> softphone but they are not registering. Sofia status also not showing any >>>> user registrations. FreeSWITCH is listening on port 5060 and every >>>> configuration has been done according to the docs. What could be the issue? >>>> >>>> Any help appreciated. >>>> >>>> Thanks in advance. >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/e7fcee02/attachment-0001.html From intralanman at freeswitch.org Wed Apr 10 14:34:10 2013 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 10 Apr 2013 06:34:10 -0400 Subject: [Freeswitch-users] annoying jira automatic actions week or two after tickets closed In-Reply-To: <5164C648.7000305@mst.edu> References: <5164C648.7000305@mst.edu> Message-ID: This is the first I've heard of this. The "Auto Admin" is usually a big help. That said, email me directly with examples and ill take a look. -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/30d94ca6/attachment.html From steveayre at gmail.com Thu Apr 11 12:35:57 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 11 Apr 2013 09:35:57 +0100 Subject: [Freeswitch-users] Default SIP users not registering In-Reply-To: References: <5165051B.5050906@softnet.si> <51650CA2.8090106@softnet.si> <51653477.5050407@softnet.si> <51653D1C.1040604@softnet.si> Message-ID: Are you sure that FS is listening on that port on that specific ip? Any firewall that might be blocking the packets? Run: sofia global siptrace on That'll include the sip trace in the FS debug logs, confirming whether FS is actually receiving the packets. Steve On 10 Apr 2013, at 11:28, Ashish gautam wrote: > FS is not replying anything there are just subscription and registration packets. > > Thanks. > > On Wed, Apr 10, 2013 at 3:51 PM, Miha wrote: >> OK, just must post whole sip trace. This is just registration packet from blink/uac, I can not see what FS replied on this. >> >> miha >> >> Dne 4/10/2013 12:00 PM, pi?e Ashish gautam: >>> Hi Miha, >>> >>> This is the registration packet: >>> >>> 15:29:25.509003 IP (tos 0x0, ttl 128, id 31452, offset 0, flags [none], proto UDP (17), length 525) >>> 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 497 >>> z.....lK >>> ... >>> .........8%REGISTER sip:10.1.30.229 SIP/2.0 >>> Via: SIP/2.0/UDP 10.1.30.210:53652;rport;branch=z9hG4bKPj9ccb691852ca419ea305c4cdd79aba44 >>> Max-Forwards: 70 >>> From: "Ashish" ;tag=5970e44547c642a2b6e335b5bbee73ea >>> To: "Ashish" >>> Contact: ;+sip.instance="" >>> Call-ID: a88024e2bdd04101a57f713d57af681e >>> CSeq: 1 REGISTER >>> Expires: 3600 >>> User-Agent: Blink 0.2.10 (Windows) >>> Content-Length: 0 >>> >>> >>> On Wed, Apr 10, 2013 at 3:14 PM, Miha wrote: >>>> Hi, >>>> >>>> what about registration packet and etc? >>>> >>>> this is only subscribe for presence. >>>> >>>> miha >>>> >>>> Dne 4/10/2013 11:07 AM, pi?e Ashish gautam: >>>>> Miha, >>>>> >>>>> This is the output of the tcpdump -nq -s 0 -A -vvv -i eth0 port 5060 for the related packet : >>>>> >>>>> 13:29:37.050278 IP (tos 0x0, ttl 128, id 9943, offset 0, flags [none], proto UDP (17), length 640) >>>>> 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 612 >>>>> E...&....... >>>>> ... >>>>> ........l..SUBSCRIBE sip:1100 at 10.1.30.229 SIP/2.0 >>>>> Via: SIP/2.0/UDP 10.1.30.210:53652;rport;branch=z9hG4bKPj91db66336d564de5b3b3e50c03ac32b3 >>>>> Max-Forwards: 70 >>>>> From: "Ashish" ;tag=3058239a404c4a4dbcd438916aa8033b >>>>> To: >>>>> Contact: >>>>> Call-ID: b3d45216211242afa13ae86734ea4408 >>>>> CSeq: 30541 SUBSCRIBE >>>>> Event: message-summary >>>>> Expires: 3600 >>>>> Accept: application/simple-message-summary >>>>> Allow-Events: conference, message-summary, dialog, presence, presence.winfo, xcap-diff, dialog.winfo, refer >>>>> User-Agent: Blink 0.2.10 (Windows) >>>>> Content-Length: 0 >>>>> >>>>> >>>>> >>>>> >>>>> On Wed, Apr 10, 2013 at 12:24 PM, Miha wrote: >>>>>> did you try to use whireshark? >>>>>> >>>>>> miha >>>>>> >>>>>> Dne 4/10/2013 8:28 AM, pi?e Ashish gautam: >>>>>>> Thanks Miha, >>>>>>> >>>>>>> sip trace shows nothing. >>>>>>> >>>>>>> >>>>>>> On Wed, Apr 10, 2013 at 11:52 AM, Miha wrote: >>>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> was does sip trace say. You must add user to default directory and that do reloadxml. >>>>>>>> >>>>>>>> You can past you log to see what is going on.. >>>>>>>> >>>>>>>> miha >>>>>>>> >>>>>>>> Dne 4/10/2013 8:09 AM, pi?e Ashish gautam: >>>>>>>>> Hi, >>>>>>>>> >>>>>>>>> I am trying to register Default SIP users i.e. 1019,1004 etc from a softphone but they are not registering. Sofia status also not showing any user registrations. FreeSWITCH is listening on port 5060 and every configuration has been done according to the docs. What could be the issue? >>>>>>>>> >>>>>>>>> Any help appreciated. >>>>>>>>> >>>>>>>>> Thanks in advance. >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at 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/20130411/9a84a84b/attachment-0001.html From ashish at nms.co.in Wed Apr 10 14:46:11 2013 From: ashish at nms.co.in (Ashish gautam) Date: Wed, 10 Apr 2013 16:16:11 +0530 Subject: [Freeswitch-users] Default SIP users not registering In-Reply-To: References: <5165051B.5050906@softnet.si> <51650CA2.8090106@softnet.si> <51653477.5050407@softnet.si> <51653D1C.1040604@softnet.si> Message-ID: Hi Steve, The domain is not defined according to the logs, I am getting this warning: 2013-04-10 16:13:18.614508 [WARNING] sofia_reg.c:2506 Can't find user [ 300 at 10.1.30.229] You must define a domain called '10.1.30.229' in your directory and add a user with the id="300" attribute and you must configure your device to use the proper domain in it's authentication credentials. I have defined domain in vars.xml as: On Thu, Apr 11, 2013 at 2:05 PM, Steven Ayre wrote: > Are you sure that FS is listening on that port on that specific ip? Any > firewall that might be blocking the packets? > > Run: > sofia global siptrace on > > That'll include the sip trace in the FS debug logs, confirming whether FS > is actually receiving the packets. > > Steve > > > > On 10 Apr 2013, at 11:28, Ashish gautam wrote: > > FS is not replying anything there are just subscription and registration > packets. > > Thanks. > > On Wed, Apr 10, 2013 at 3:51 PM, Miha wrote: > >> OK, just must post whole sip trace. This is just registration packet >> from blink/uac, I can not see what FS replied on this. >> >> miha >> >> Dne 4/10/2013 12:00 PM, pi?e Ashish gautam: >> >> Hi Miha, >> >> This is the registration packet: >> >> 15:29:25.509003 IP (tos 0x0, ttl 128, id 31452, offset 0, flags [none], >> proto UDP (17), length 525) >> 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 497 >> z.....lK >> ... >> .........8%REGISTER sip:10.1.30.229 SIP/2.0 >> Via: SIP/2.0/UDP 10.1.30.210:53652 >> ;rport;branch=z9hG4bKPj9ccb691852ca419ea305c4cdd79aba44 >> Max-Forwards: 70 >> From: "Ashish" > >;tag=5970e44547c642a2b6e335b5bbee73ea >> To: "Ashish" >> Contact: > >;+sip.instance="" >> Call-ID: a88024e2bdd04101a57f713d57af681e >> CSeq: 1 REGISTER >> Expires: 3600 >> User-Agent: Blink 0.2.10 (Windows) >> Content-Length: 0 >> >> >> On Wed, Apr 10, 2013 at 3:14 PM, Miha wrote: >> >>> Hi, >>> >>> what about registration packet and etc? >>> >>> this is only subscribe for presence. >>> >>> miha >>> >>> Dne 4/10/2013 11:07 AM, pi?e Ashish gautam: >>> >>> Miha, >>> >>> This is the output of the tcpdump -nq -s 0 -A -vvv -i eth0 port 5060 >>> for the related packet : >>> >>> 13:29:37.050278 IP (tos 0x0, ttl 128, id 9943, offset 0, flags [none], >>> proto UDP (17), length 640) >>> 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 612 >>> E...&....... >>> ... >>> ........l..SUBSCRIBE sip:1100 at 10.1.30.229 SIP/2.0 >>> Via: SIP/2.0/UDP 10.1.30.210:53652 >>> ;rport;branch=z9hG4bKPj91db66336d564de5b3b3e50c03ac32b3 >>> Max-Forwards: 70 >>> From: "Ashish" >> >;tag=3058239a404c4a4dbcd438916aa8033b >>> To: >>> Contact: >>> Call-ID: b3d45216211242afa13ae86734ea4408 >>> CSeq: 30541 SUBSCRIBE >>> Event: message-summary >>> Expires: 3600 >>> Accept: application/simple-message-summary >>> Allow-Events: conference, message-summary, dialog, presence, >>> presence.winfo, xcap-diff, dialog.winfo, refer >>> User-Agent: Blink 0.2.10 (Windows) >>> Content-Length: 0 >>> >>> >>> >>> >>> On Wed, Apr 10, 2013 at 12:24 PM, Miha wrote: >>> >>>> did you try to use whireshark? >>>> >>>> miha >>>> >>>> Dne 4/10/2013 8:28 AM, pi?e Ashish gautam: >>>> >>>> Thanks Miha, >>>> >>>> sip trace shows nothing. >>>> >>>> >>>> On Wed, Apr 10, 2013 at 11:52 AM, Miha wrote: >>>> >>>>> >>>>> Hi, >>>>> >>>>> was does sip trace say. You must add user to default directory and >>>>> that do reloadxml. >>>>> >>>>> You can past you log to see what is going on.. >>>>> >>>>> miha >>>>> >>>>> Dne 4/10/2013 8:09 AM, pi?e Ashish gautam: >>>>> >>>>> Hi, >>>>> >>>>> I am trying to register Default SIP users i.e. 1019,1004 etc from a >>>>> softphone but they are not registering. Sofia status also not showing any >>>>> user registrations. FreeSWITCH is listening on port 5060 and every >>>>> configuration has been done according to the docs. What could be the issue? >>>>> >>>>> Any help appreciated. >>>>> >>>>> Thanks in advance. >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20130410/e14d3e4c/attachment-0001.html From mehroz.ashraf85 at gmail.com Wed Apr 10 15:15:29 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Wed, 10 Apr 2013 04:15:29 -0700 (PDT) Subject: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode In-Reply-To: References: Message-ID: <1365592528949-7589602.post@n2.nabble.com> Thanks Everyone for your feedback! 1) ZRTP SAS exchange works flawlessly using with jitsi when proxy is turned ON (with and without FS as MITM) 2) Works with bypass media BUT only on the same network at client side (Very Strange, why is that so?) Regarding linphone patch, yes it was tested today (thanks Eli) but Video call is troublesome. Can you give me any idea about the reason of this issue, we might fight with it! and the final words........ What could be done to make everything working? FS development or changing SIP stack to PJSIP/CSIPSIMPLE? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/problems-with-freeswitch-zrtp-in-proxy-media-mode-tp7586936p7589602.html Sent from the freeswitch-users mailing list archive at Nabble.com. From cal.leeming at simplicitymedialtd.co.uk Wed Apr 10 15:18:11 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 10 Apr 2013 12:18:11 +0100 Subject: [Freeswitch-users] Presence Sanity Check In-Reply-To: <4FC40472-8AE5-44D4-A3F8-1553B84F8F8B@insensate.co.uk> References: <4FC40472-8AE5-44D4-A3F8-1553B84F8F8B@insensate.co.uk> Message-ID: Thank you for taking the time to write this up, it was an interesting read! Cal On Wed, Apr 10, 2013 at 2:09 AM, Lawrence Conroy wrote: > Hi Folks, > > ignition on> > > You have to be there to get your way. When all this was being nailed down, > no-one spoke up. Trust me, I was listening. > I was there during this phase -- the SIP crowd who came up with the > standards *were* open to well argued ideas, so if you all had been there, > you could well have deflected the dumb schemes as they came up. [OK, you'd > have argued with a bunch of folk from MoFoCos, who *really* wanted to get > SIP working as they know that circuit switched infrastructures just were > not going to cut it, and had their own priorities; frankly, they were the > main ones speaking and there wasn't anyone else to argue against them]. > > SIP was always designed as a "single packet per message" scheme. > IIRC, there were always two SIP camps -- those who required secure > signalling (and thus TCP/SRTP and so on), and those who wanted a really > lightweight scheme with low bandwidth overheads [remember, this all came up > in the '90s, when bandwidth was NOT cheap]. > Drop back to TCP was seen as a rare event, as the SIP main body was tight > (very tight, if you used the short forms), and you simply wouldn't have > sub-parts that big with *sane* SIP messages so that MTU was going to hurt. > If you WERE going to send something that was over the MTU, you'd start with > or switch over to TCP -- the assumption being that this latter would only > happen in rare situations (even when carrying all kinds of piggy-backed > ISUP data). > > As for messaging -- remember that the IETF was stuck in the grinder > between the big IM providers, who were trying to find a common protocol > between their networks. > This looked like being some IMPP-style mung. > In parallel with that, the SIP folks were initially content with MESSAGE; > after all, who in their right mind would send an HTML page full of mung, > when it was just for display on a phone -- thus we had an SMS style scheme > that worked fine, and all was cool. IIRC, MESSAGE (and sending > in-signalling channel messages) was knocked up overnight during an IETF > meeting, just to point out to the IMPP folk and the "data architects" that > it didn't need to be hard if you have a sensible signalling scheme. > > Then the XMPP stuff arose from the ashes of the IM network interop work > (which had crashed and burned fairly spectacularly in the IETF). > The XMPP folk had already drunk far too much of the XML cool aid. > > By this time, SIP had already produced a Presence model that was rapidly > getting complex, and choosing to switch over to XML as the one true > representation made the SIP sub-parts expand really quickly. At this point, > I personally lost interest. Even the names were getting beyond me: I mean, > a "presentity" -- had the 400 pound Gorilla who specified this lot not been > taught English? > > In particular, the MoFoCos wanted to be able to provide "rich presence" > and something at least as good as IM, which had become fashionable even > they had heard of it. As they'd just rearranged their networks to use SIP > at the core, that meant that they needed some serious work done. Recall > that a lot of the parameters and extra mung in the headers was introduced > just for commercial providers (and both 3GPP and 3GPP2 were using SIP to > carry everything including the kitchen sink inside the signalling packets > -- i.e., SIP, which gave the main body a heck of a bloat, before we start > on the sub-parts). > If SIP was to be able to carry these kind of bloated HTML-like > monstrosities both as headers AND to cover the kind of "rich presence" and > IM stuff we all have grown to love, the clean -- single packet -- model > started to creak, and the transition from UDP to TCP was no longer going to > be a rare event for loons who just couldn't control themselves and hadn't > heard of sigcomp. > > As for the 1500 byte limit (i.e. the typical MTU) for UDP messages, that's > a reflection of the original model of single packet messages. Going for a > scheme that sends out > MTU messages (i.e. sends fragments out over the > net), was and is a hideous kludge, and falls over with a lot of the cheap > home routers people seem to use. > > So ... if you're going to do complex headers and messaging or the 15 > million different variants on presence (as they are now, i.e. XML to the > hilt), then start out with TCP -- don't even bother trying to use god's own > protocol in a session. > If you're doing voice and -basic- presence, UDP is fine (and don't forget, > sigcomp exists to keep the bandwidth down, particularly between servers). > > all the best, > Lawrence > > On 10 Apr 2013, at 00:48, Ira Tessler wrote: > > I second the hear hear!!! > > > > Ira Tessler > > Lead Software Engineer > > ConnectMe > > (732) 490-9007 x2 > > ira at connectmevoice.com > > > > > > On Mon, Apr 8, 2013 at 5:50 PM, Cal Leeming [Simplicity Media Ltd] < > > cal.leeming at simplicitymedialtd.co.uk> wrote: > > > >> Hear hear! > >> > >> On Mon, Apr 8, 2013 at 10:17 PM, Anthony Minessale < > >> anthony.minessale at gmail.com> wrote: > >> > >>> Its actually somewhat ridiculous and one of my personal favorites in > >>> terms of RFC smoke and mirrors in SIP to try and cover up a flaw with > more > >>> specs. They basically say: > >>> > >>> > >>> If the total packet including the sip headers and the payload exceeds > the > >>> MTU and you are using the UDP transport, you MUST try sending the > packet > >>> over TCP instead. If that times out or fails then you SHOULD send it > over > >>> UDP anyway. > >>> > >>> > >>> THEREFORE by virtue of this decree: > >>> > >>> You MUST implement your sip stack to accept UDP packets of up to 65536 > >>> bytes. > >>> > >>> AND > >>> > >>> You MUST implement both TCP and UDP transports. > >>> > >>> > >>> So in short, you are not supposed to send anything over udp that > exceeds > >>> the mtu yet you are required to implement it so its possible. > >>> > >>> Many stacks, including Asterisk for many of the first half of FS > >>> existence, did not implement TCP so with this rule being enforced, the > >>> packets would sit there for 2-5 min then give up and change to UDP. > How's > >>> that for PDD. > >>> > >>> Anyway, we choose to ignore this rule intentionally and just stick with > >>> the negotiated protocol. If you find yourself in this situation the > >>> solution is to use TCP. > >>> > >>> > >>> P.S. > >>> > >>> If they did not use 2k of XML to transmit about 12 bytes worth of > useful > >>> info regarding the state of the presence, we would not have this > problem to > >>> begin with ;) > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> On Mon, Apr 8, 2013 at 4:02 PM, Ira Tessler >wrote: > >>> > >>>> Thank you for your information. It did help! > >>>> > >>>> Ira > >>>> > >>>> Ira Tessler > >>>> Lead Software Engineer > >>>> ConnectMe > >>>> (732) 490-9007 x2 > >>>> ira at connectmevoice.com > >>>> > >>>> > >>>> On Sun, Apr 7, 2013 at 12:49 PM, Cal Leeming [Simplicity Media Ltd] < > >>>> cal.leeming at simplicitymedialtd.co.uk> wrote: > >>>> > >>>>> In regards to the UDP fragmentation, this is an extremely good > question. > >>>>> > >>>>> Only yesterday I started to build a simple forwarding SBC in Python > >>>>> using UDP sockets, however I came up against the same theoretical > problem > >>>>> of packets being larger than 1500 bytes. > >>>>> > >>>>> I've had a read through various documentation; > >>>>> http://www.rfc-ref.org/RFC-TEXTS/3261/chapter18.html > >>>>> http://www.ietf.org/rfc/rfc3428.txt > >>>>> > >>>>> > https://lists.cs.columbia.edu/pipermail/sip-implementors/2006-August/013857.html > >>>>> > >>>>> > http://lists.freeswitch.org/pipermail/freeswitch-users/2011-February/068372.html > >>>>> > >>>>> > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05912.html > >>>>> > >>>>> Anthony has stated the following; > >>>>> *"the only reliable answer is use TCP."* > >>>>> > >>>>> There was also the option of enabling compact headers; > >>>>> > >>>>> > http://lists.freeswitch.org/pipermail/freeswitch-users/2011-December/078633.html > >>>>> > >>>>> From what I can tell, there is no way to guarantee this problem won't > >>>>> happen unless you use TCP. You could reduce the packet size by > compacting > >>>>> headers or removing codecs, but this would be on the assumption that > every > >>>>> hop is running at 1500 MTU. > >>>>> > >>>>> Hope this helps! > >>>>> > >>>>> Cal > >>>>> > >>>>> On Sat, Apr 6, 2013 at 2:28 PM, Ira Tessler >wrote: > >>>>> > >>>>>> I just need a little guidance with the way presence works. Forgive > me > >>>>>> if I am asking novice questions. > >>>>>> > >>>>>> Background (simple version) > >>>>>> We run Freeswitch in a hosted/cloud environment in a data center. We > >>>>>> have IP phones in our office on our LAN. > >>>>>> > >>>>>> That way I am understanding how Presence works, I am just learning > >>>>>> this, is that when a BLF button is programmed on a phone, that > phone will > >>>>>> send a "Subscribe" message to Freeswitch. The subscriptions are > stored in > >>>>>> the sip_subscriptions table (i think) in the sofia database for the > sip > >>>>>> profile. When calls come in for that subscription, Freeswitch will > send out > >>>>>> a NOTIFY message to the phone that subscribed in order to change > the state > >>>>>> of the BLF Light. > >>>>>> > >>>>>> He is my questions/issue/confusion. > >>>>>> All our phones use UDP which has a maximum packet size of 1500 > bytes. > >>>>>> When doing a sofia global siptrace on, I notice that most of the > NOTIFY > >>>>>> messages are greater then 1500 bytes. That will cause packet > fragmentation. > >>>>>> So if the NOTIFY message is fragmented, will it get to the phone > correctly? > >>>>>> (all the time, some of the time, never??) > >>>>>> > >>>>>> If the the answer is other then ("all the time"), how do I fix this? > >>>>>> The only solution I can come up with is having my phones use TCP > instead of > >>>>>> UDP. Is that the correctly solution? Did anyone else out there run > into > >>>>>> this issue and if so, what is the "best practice" solution (if > there is > >>>>>> one)? > >>>>>> > >>>>>> Thank you in advance! > >>>>>> > >>>>>> Ira Tessler > >>>>>> Lead Software Engineer > >>>>>> ConnectMe > >>>>>> (732) 490-9007 x2 > >>>>>> ira at connectmevoice.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 > >>>> > >>>> > >>> > >>> > >>> -- > >>> Anthony Minessale II > >>> > >>> FreeSWITCH http://www.freeswitch.org/ > >>> ClueCon http://www.cluecon.com/ > >>> Twitter: http://twitter.com/FreeSWITCH_wire > >>> > >>> AIM: anthm > >>> MSN:anthony_minessale at 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 > >> > >> > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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/20130410/6731a044/attachment-0001.html From oej at edvina.net Wed Apr 10 15:45:03 2013 From: oej at edvina.net (Olle E. Johansson) Date: Wed, 10 Apr 2013 13:45:03 +0200 Subject: [Freeswitch-users] Presence Sanity Check In-Reply-To: References: <4FC40472-8AE5-44D4-A3F8-1553B84F8F8B@insensate.co.uk> Message-ID: <9367B367-D8BC-46EB-A7DA-F6D32792F1CC@edvina.net> +100 - especially about the SIMPLE part that I also believe got totally out of hands. /O 10 apr 2013 kl. 13:18 skrev "Cal Leeming [Simplicity Media Ltd]" : > Thank you for taking the time to write this up, it was an interesting read! > > Cal > > On Wed, Apr 10, 2013 at 2:09 AM, Lawrence Conroy wrote: > Hi Folks, > > ignition on> > > You have to be there to get your way. When all this was being nailed down, no-one spoke up. Trust me, I was listening. > I was there during this phase -- the SIP crowd who came up with the standards *were* open to well argued ideas, so if you all had been there, you could well have deflected the dumb schemes as they came up. [OK, you'd have argued with a bunch of folk from MoFoCos, who *really* wanted to get SIP working as they know that circuit switched infrastructures just were not going to cut it, and had their own priorities; frankly, they were the main ones speaking and there wasn't anyone else to argue against them]. > > SIP was always designed as a "single packet per message" scheme. > IIRC, there were always two SIP camps -- those who required secure signalling (and thus TCP/SRTP and so on), and those who wanted a really lightweight scheme with low bandwidth overheads [remember, this all came up in the '90s, when bandwidth was NOT cheap]. > Drop back to TCP was seen as a rare event, as the SIP main body was tight (very tight, if you used the short forms), and you simply wouldn't have sub-parts that big with *sane* SIP messages so that MTU was going to hurt. If you WERE going to send something that was over the MTU, you'd start with or switch over to TCP -- the assumption being that this latter would only happen in rare situations (even when carrying all kinds of piggy-backed ISUP data). > > As for messaging -- remember that the IETF was stuck in the grinder between the big IM providers, who were trying to find a common protocol between their networks. > This looked like being some IMPP-style mung. > In parallel with that, the SIP folks were initially content with MESSAGE; after all, who in their right mind would send an HTML page full of mung, when it was just for display on a phone -- thus we had an SMS style scheme that worked fine, and all was cool. IIRC, MESSAGE (and sending in-signalling channel messages) was knocked up overnight during an IETF meeting, just to point out to the IMPP folk and the "data architects" that it didn't need to be hard if you have a sensible signalling scheme. > > Then the XMPP stuff arose from the ashes of the IM network interop work (which had crashed and burned fairly spectacularly in the IETF). > The XMPP folk had already drunk far too much of the XML cool aid. > > By this time, SIP had already produced a Presence model that was rapidly getting complex, and choosing to switch over to XML as the one true representation made the SIP sub-parts expand really quickly. At this point, I personally lost interest. Even the names were getting beyond me: I mean, a "presentity" -- had the 400 pound Gorilla who specified this lot not been taught English? > > In particular, the MoFoCos wanted to be able to provide "rich presence" and something at least as good as IM, which had become fashionable even they had heard of it. As they'd just rearranged their networks to use SIP at the core, that meant that they needed some serious work done. Recall that a lot of the parameters and extra mung in the headers was introduced just for commercial providers (and both 3GPP and 3GPP2 were using SIP to carry everything including the kitchen sink inside the signalling packets -- i.e., SIP, which gave the main body a heck of a bloat, before we start on the sub-parts). > If SIP was to be able to carry these kind of bloated HTML-like monstrosities both as headers AND to cover the kind of "rich presence" and IM stuff we all have grown to love, the clean -- single packet -- model started to creak, and the transition from UDP to TCP was no longer going to be a rare event for loons who just couldn't control themselves and hadn't heard of sigcomp. > > As for the 1500 byte limit (i.e. the typical MTU) for UDP messages, that's a reflection of the original model of single packet messages. Going for a scheme that sends out > MTU messages (i.e. sends fragments out over the net), was and is a hideous kludge, and falls over with a lot of the cheap home routers people seem to use. > > So ... if you're going to do complex headers and messaging or the 15 million different variants on presence (as they are now, i.e. XML to the hilt), then start out with TCP -- don't even bother trying to use god's own protocol in a session. > If you're doing voice and -basic- presence, UDP is fine (and don't forget, sigcomp exists to keep the bandwidth down, particularly between servers). > > all the best, > Lawrence > > On 10 Apr 2013, at 00:48, Ira Tessler wrote: > > I second the hear hear!!! > > > > Ira Tessler > > Lead Software Engineer > > ConnectMe > > (732) 490-9007 x2 > > ira at connectmevoice.com > > > > > > On Mon, Apr 8, 2013 at 5:50 PM, Cal Leeming [Simplicity Media Ltd] < > > cal.leeming at simplicitymedialtd.co.uk> wrote: > > > >> Hear hear! > >> > >> On Mon, Apr 8, 2013 at 10:17 PM, Anthony Minessale < > >> anthony.minessale at gmail.com> wrote: > >> > >>> Its actually somewhat ridiculous and one of my personal favorites in > >>> terms of RFC smoke and mirrors in SIP to try and cover up a flaw with more > >>> specs. They basically say: > >>> > >>> > >>> If the total packet including the sip headers and the payload exceeds the > >>> MTU and you are using the UDP transport, you MUST try sending the packet > >>> over TCP instead. If that times out or fails then you SHOULD send it over > >>> UDP anyway. > >>> > >>> > >>> THEREFORE by virtue of this decree: > >>> > >>> You MUST implement your sip stack to accept UDP packets of up to 65536 > >>> bytes. > >>> > >>> AND > >>> > >>> You MUST implement both TCP and UDP transports. > >>> > >>> > >>> So in short, you are not supposed to send anything over udp that exceeds > >>> the mtu yet you are required to implement it so its possible. > >>> > >>> Many stacks, including Asterisk for many of the first half of FS > >>> existence, did not implement TCP so with this rule being enforced, the > >>> packets would sit there for 2-5 min then give up and change to UDP. How's > >>> that for PDD. > >>> > >>> Anyway, we choose to ignore this rule intentionally and just stick with > >>> the negotiated protocol. If you find yourself in this situation the > >>> solution is to use TCP. > >>> > >>> > >>> P.S. > >>> > >>> If they did not use 2k of XML to transmit about 12 bytes worth of useful > >>> info regarding the state of the presence, we would not have this problem to > >>> begin with ;) > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> On Mon, Apr 8, 2013 at 4:02 PM, Ira Tessler wrote: > >>> > >>>> Thank you for your information. It did help! > >>>> > >>>> Ira > >>>> > >>>> Ira Tessler > >>>> Lead Software Engineer > >>>> ConnectMe > >>>> (732) 490-9007 x2 > >>>> ira at connectmevoice.com > >>>> > >>>> > >>>> On Sun, Apr 7, 2013 at 12:49 PM, Cal Leeming [Simplicity Media Ltd] < > >>>> cal.leeming at simplicitymedialtd.co.uk> wrote: > >>>> > >>>>> In regards to the UDP fragmentation, this is an extremely good question. > >>>>> > >>>>> Only yesterday I started to build a simple forwarding SBC in Python > >>>>> using UDP sockets, however I came up against the same theoretical problem > >>>>> of packets being larger than 1500 bytes. > >>>>> > >>>>> I've had a read through various documentation; > >>>>> http://www.rfc-ref.org/RFC-TEXTS/3261/chapter18.html > >>>>> http://www.ietf.org/rfc/rfc3428.txt > >>>>> > >>>>> https://lists.cs.columbia.edu/pipermail/sip-implementors/2006-August/013857.html > >>>>> > >>>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-February/068372.html > >>>>> > >>>>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05912.html > >>>>> > >>>>> Anthony has stated the following; > >>>>> *"the only reliable answer is use TCP."* > >>>>> > >>>>> There was also the option of enabling compact headers; > >>>>> > >>>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-December/078633.html > >>>>> > >>>>> From what I can tell, there is no way to guarantee this problem won't > >>>>> happen unless you use TCP. You could reduce the packet size by compacting > >>>>> headers or removing codecs, but this would be on the assumption that every > >>>>> hop is running at 1500 MTU. > >>>>> > >>>>> Hope this helps! > >>>>> > >>>>> Cal > >>>>> > >>>>> On Sat, Apr 6, 2013 at 2:28 PM, Ira Tessler wrote: > >>>>> > >>>>>> I just need a little guidance with the way presence works. Forgive me > >>>>>> if I am asking novice questions. > >>>>>> > >>>>>> Background (simple version) > >>>>>> We run Freeswitch in a hosted/cloud environment in a data center. We > >>>>>> have IP phones in our office on our LAN. > >>>>>> > >>>>>> That way I am understanding how Presence works, I am just learning > >>>>>> this, is that when a BLF button is programmed on a phone, that phone will > >>>>>> send a "Subscribe" message to Freeswitch. The subscriptions are stored in > >>>>>> the sip_subscriptions table (i think) in the sofia database for the sip > >>>>>> profile. When calls come in for that subscription, Freeswitch will send out > >>>>>> a NOTIFY message to the phone that subscribed in order to change the state > >>>>>> of the BLF Light. > >>>>>> > >>>>>> He is my questions/issue/confusion. > >>>>>> All our phones use UDP which has a maximum packet size of 1500 bytes. > >>>>>> When doing a sofia global siptrace on, I notice that most of the NOTIFY > >>>>>> messages are greater then 1500 bytes. That will cause packet fragmentation. > >>>>>> So if the NOTIFY message is fragmented, will it get to the phone correctly? > >>>>>> (all the time, some of the time, never??) > >>>>>> > >>>>>> If the the answer is other then ("all the time"), how do I fix this? > >>>>>> The only solution I can come up with is having my phones use TCP instead of > >>>>>> UDP. Is that the correctly solution? Did anyone else out there run into > >>>>>> this issue and if so, what is the "best practice" solution (if there is > >>>>>> one)? > >>>>>> > >>>>>> Thank you in advance! > >>>>>> > >>>>>> Ira Tessler > >>>>>> Lead Software Engineer > >>>>>> ConnectMe > >>>>>> (732) 490-9007 x2 > >>>>>> ira at connectmevoice.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 > >>>> > >>>> > >>> > >>> > >>> -- > >>> Anthony Minessale II > >>> > >>> FreeSWITCH http://www.freeswitch.org/ > >>> ClueCon http://www.cluecon.com/ > >>> Twitter: http://twitter.com/FreeSWITCH_wire > >>> > >>> AIM: anthm > >>> MSN:anthony_minessale at 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 > >> > >> > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/384784cb/attachment-0001.html From miha at softnet.si Wed Apr 10 15:54:46 2013 From: miha at softnet.si (Miha) Date: Wed, 10 Apr 2013 13:54:46 +0200 Subject: [Freeswitch-users] Default SIP users not registering In-Reply-To: References: <5165051B.5050906@softnet.si> <51650CA2.8090106@softnet.si> <51653477.5050407@softnet.si> <51653D1C.1040604@softnet.si> Message-ID: <51655306.4080904@softnet.si> Did you start FS?:) firewaill, acl,... Dne 4/11/2013 10:35 AM, pis(e Steven Ayre: > Are you sure that FS is listening on that port on that specific ip? > Any firewall that might be blocking the packets? > > Run: > sofia global siptrace on > > That'll include the sip trace in the FS debug logs, confirming whether > FS is actually receiving the packets. > > Steve > > > > On 10 Apr 2013, at 11:28, Ashish gautam > wrote: > >> FS is not replying anything there are just subscription and >> registration packets. >> >> Thanks. >> >> On Wed, Apr 10, 2013 at 3:51 PM, Miha > > wrote: >> >> OK, just must post whole sip trace. This is just registration >> packet from blink/uac, I can not see what FS replied on this. >> >> miha >> >> Dne 4/10/2013 12:00 PM, pis(e Ashish gautam: >>> Hi Miha, >>> >>> This is the registration packet: >>> >>> 15:29:25.509003 IP (tos 0x0, ttl 128, id 31452, offset 0, flags >>> [none], proto UDP (17), length 525) >>> 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, >>> length 497 >>> z.....lK >>> ... >>> .........8%REGISTER sip:10.1.30.229 SIP/2.0 >>> Via: SIP/2.0/UDP >>> 10.1.30.210:53652;rport;branch=z9hG4bKPj9ccb691852ca419ea305c4cdd79aba44 >>> Max-Forwards: 70 >>> From: "Ashish" >> >;tag=5970e44547c642a2b6e335b5bbee73ea >>> To: "Ashish" > >>> Contact: >> >;+sip.instance="" >>> Call-ID: a88024e2bdd04101a57f713d57af681e >>> CSeq: 1 REGISTER >>> Expires: 3600 >>> User-Agent: Blink 0.2.10 (Windows) >>> Content-Length: 0 >>> >>> >>> On Wed, Apr 10, 2013 at 3:14 PM, Miha >> > wrote: >>> >>> Hi, >>> >>> what about registration packet and etc? >>> >>> this is only subscribe for presence. >>> >>> miha >>> >>> Dne 4/10/2013 11:07 AM, pis(e Ashish gautam: >>>> Miha, >>>> >>>> This is the output of the tcpdump -nq -s 0 -A -vvv -i eth0 >>>> port 5060 for the related packet : >>>> >>>> 13:29:37.050278 IP (tos 0x0, ttl 128, id 9943, offset 0, >>>> flags [none], proto UDP (17), length 640) >>>> 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, >>>> length 612 >>>> E...&....... >>>> ... >>>> ........l..SUBSCRIBE sip:1100 at 10.1.30.229 >>>> SIP/2.0 >>>> Via: SIP/2.0/UDP >>>> 10.1.30.210:53652;rport;branch=z9hG4bKPj91db66336d564de5b3b3e50c03ac32b3 >>>> Max-Forwards: 70 >>>> From: "Ashish" >>> >;tag=3058239a404c4a4dbcd438916aa8033b >>>> To: > >>>> Contact: >>> > >>>> Call-ID: b3d45216211242afa13ae86734ea4408 >>>> CSeq: 30541 SUBSCRIBE >>>> Event: message-summary >>>> Expires: 3600 >>>> Accept: application/simple-message-summary >>>> Allow-Events: conference, message-summary, dialog, >>>> presence, presence.winfo, xcap-diff, dialog.winfo, refer >>>> User-Agent: Blink 0.2.10 (Windows) >>>> Content-Length: 0 >>>> >>>> >>>> >>>> >>>> On Wed, Apr 10, 2013 at 12:24 PM, Miha >>> > wrote: >>>> >>>> did you try to use whireshark? >>>> >>>> miha >>>> >>>> Dne 4/10/2013 8:28 AM, pis(e Ashish gautam: >>>>> Thanks Miha, >>>>> >>>>> sip trace shows nothing. >>>>> >>>>> >>>>> On Wed, Apr 10, 2013 at 11:52 AM, Miha >>>>> > wrote: >>>>> >>>>> >>>>> Hi, >>>>> >>>>> was does sip trace say. You must add user to >>>>> default directory and that do reloadxml. >>>>> >>>>> You can past you log to see what is going on.. >>>>> >>>>> miha >>>>> >>>>> Dne 4/10/2013 8:09 AM, pis(e Ashish gautam: >>>>>> Hi, >>>>>> >>>>>> I am trying to register Default SIP users i.e. >>>>>> 1019,1004 etc from a softphone but they are not >>>>>> registering. Sofia status also not showing any >>>>>> user registrations. FreeSWITCH is listening on >>>>>> port 5060 and every configuration has been done >>>>>> according to the docs. What could be the issue? >>>>>> >>>>>> Any help appreciated. >>>>>> >>>>>> Thanks in advance. >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel >>>>> Communication Server >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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/20130410/c54037df/attachment-0001.html From gvvsubhashkumar at gmail.com Wed Apr 10 11:43:23 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Wed, 10 Apr 2013 13:13:23 +0530 Subject: [Freeswitch-users] Sound Not Ok(Choppy Sound) After 250 Ports Message-ID: Hi, I ran the load test, raising the port count slowly (every 5 to 10 minutes or so) to 100, 150, 200, 250, 275 and 300 ports. There were no audio problems until I reached 275 ports. After probably 10 minutes at 275 ports, audio problems started, and after that they were consistent. Of course I can?t guarantee that there were no audio problems at lower port counts - if the problem only occurred intermittently I might not have caught it or I might not have waited long enough. Once the audio problems started, I slowly lowered the number of ports back to 250 ports. The audio problems disappeared after about 2 1/2 minutes at 250 ports. I then increased the number of ports back to 275 ports. Thanks, Subhash -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/12b0f432/attachment.html From steveayre at gmail.com Wed Apr 10 16:11:28 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 10 Apr 2013 13:11:28 +0100 Subject: [Freeswitch-users] Default SIP users not registering In-Reply-To: References: <5165051B.5050906@softnet.si> <51650CA2.8090106@softnet.si> <51653477.5050407@softnet.si> <51653D1C.1040604@softnet.si> Message-ID: Those settings don't have any effect on their own. They set variables which are used elsewhere in the config files. At least that suggests FS is indeed seeing the packets. Though that also means you should get a response too (probably 403 Forbidden) -Steve On 10 April 2013 11:46, Ashish gautam wrote: > Hi Steve, > > The domain is not defined according to the logs, I am getting this warning: > > 2013-04-10 16:13:18.614508 [WARNING] sofia_reg.c:2506 Can't find user [ > 300 at 10.1.30.229] > You must define a domain called '10.1.30.229' in your directory and add a > user with the id="300" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > > I have defined domain in vars.xml as: > > > > > > > > On Thu, Apr 11, 2013 at 2:05 PM, Steven Ayre wrote: > >> Are you sure that FS is listening on that port on that specific ip? Any >> firewall that might be blocking the packets? >> >> Run: >> sofia global siptrace on >> >> That'll include the sip trace in the FS debug logs, confirming whether FS >> is actually receiving the packets. >> >> Steve >> >> >> >> On 10 Apr 2013, at 11:28, Ashish gautam wrote: >> >> FS is not replying anything there are just subscription and registration >> packets. >> >> Thanks. >> >> On Wed, Apr 10, 2013 at 3:51 PM, Miha wrote: >> >>> OK, just must post whole sip trace. This is just registration packet >>> from blink/uac, I can not see what FS replied on this. >>> >>> miha >>> >>> Dne 4/10/2013 12:00 PM, pi?e Ashish gautam: >>> >>> Hi Miha, >>> >>> This is the registration packet: >>> >>> 15:29:25.509003 IP (tos 0x0, ttl 128, id 31452, offset 0, flags >>> [none], proto UDP (17), length 525) >>> 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 497 >>> z.....lK >>> ... >>> .........8%REGISTER sip:10.1.30.229 SIP/2.0 >>> Via: SIP/2.0/UDP 10.1.30.210:53652 >>> ;rport;branch=z9hG4bKPj9ccb691852ca419ea305c4cdd79aba44 >>> Max-Forwards: 70 >>> From: "Ashish" >> >;tag=5970e44547c642a2b6e335b5bbee73ea >>> To: "Ashish" >>> Contact: >> >;+sip.instance="" >>> Call-ID: a88024e2bdd04101a57f713d57af681e >>> CSeq: 1 REGISTER >>> Expires: 3600 >>> User-Agent: Blink 0.2.10 (Windows) >>> Content-Length: 0 >>> >>> >>> On Wed, Apr 10, 2013 at 3:14 PM, Miha wrote: >>> >>>> Hi, >>>> >>>> what about registration packet and etc? >>>> >>>> this is only subscribe for presence. >>>> >>>> miha >>>> >>>> Dne 4/10/2013 11:07 AM, pi?e Ashish gautam: >>>> >>>> Miha, >>>> >>>> This is the output of the tcpdump -nq -s 0 -A -vvv -i eth0 port 5060 >>>> for the related packet : >>>> >>>> 13:29:37.050278 IP (tos 0x0, ttl 128, id 9943, offset 0, flags >>>> [none], proto UDP (17), length 640) >>>> 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 612 >>>> E...&....... >>>> ... >>>> ........l..SUBSCRIBE sip:1100 at 10.1.30.229 SIP/2.0 >>>> Via: SIP/2.0/UDP 10.1.30.210:53652 >>>> ;rport;branch=z9hG4bKPj91db66336d564de5b3b3e50c03ac32b3 >>>> Max-Forwards: 70 >>>> From: "Ashish" >>> >;tag=3058239a404c4a4dbcd438916aa8033b >>>> To: >>>> Contact: >>>> Call-ID: b3d45216211242afa13ae86734ea4408 >>>> CSeq: 30541 SUBSCRIBE >>>> Event: message-summary >>>> Expires: 3600 >>>> Accept: application/simple-message-summary >>>> Allow-Events: conference, message-summary, dialog, presence, >>>> presence.winfo, xcap-diff, dialog.winfo, refer >>>> User-Agent: Blink 0.2.10 (Windows) >>>> Content-Length: 0 >>>> >>>> >>>> >>>> >>>> On Wed, Apr 10, 2013 at 12:24 PM, Miha wrote: >>>> >>>>> did you try to use whireshark? >>>>> >>>>> miha >>>>> >>>>> Dne 4/10/2013 8:28 AM, pi?e Ashish gautam: >>>>> >>>>> Thanks Miha, >>>>> >>>>> sip trace shows nothing. >>>>> >>>>> >>>>> On Wed, Apr 10, 2013 at 11:52 AM, Miha wrote: >>>>> >>>>>> >>>>>> Hi, >>>>>> >>>>>> was does sip trace say. You must add user to default directory and >>>>>> that do reloadxml. >>>>>> >>>>>> You can past you log to see what is going on.. >>>>>> >>>>>> miha >>>>>> >>>>>> Dne 4/10/2013 8:09 AM, pi?e Ashish gautam: >>>>>> >>>>>> Hi, >>>>>> >>>>>> I am trying to register Default SIP users i.e. 1019,1004 etc from a >>>>>> softphone but they are not registering. Sofia status also not showing any >>>>>> user registrations. FreeSWITCH is listening on port 5060 and every >>>>>> configuration has been done according to the docs. What could be the issue? >>>>>> >>>>>> Any help appreciated. >>>>>> >>>>>> Thanks in advance. >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> 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 >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20130410/c2bad223/attachment-0001.html From miha at softnet.si Wed Apr 10 16:24:21 2013 From: miha at softnet.si (Miha) Date: Wed, 10 Apr 2013 14:24:21 +0200 Subject: [Freeswitch-users] Default SIP users not registering In-Reply-To: References: <5165051B.5050906@softnet.si> <51650CA2.8090106@softnet.si> <51653477.5050407@softnet.si> <51653D1C.1040604@softnet.si> Message-ID: <516559F5.1040208@softnet.si> Dne 4/10/2013 2:11 PM, pis(e Steven Ayre: > Those settings don't have any effect on their own. They set variables > which are used elsewhere in the config files. > > At least that suggests FS is indeed seeing the packets. Though that > also means you should get a response too (probably 403 Forbidden) > > -Steve > > > On 10 April 2013 11:46, Ashish gautam > wrote: > > Hi Steve, > > The domain is not defined according to the logs, I am getting this > warning: > > 2013-04-10 16:13:18.614508 [WARNING] sofia_reg.c:2506 Can't find > user [300 at 10.1.30.229 ] > You must define a domain called '10.1.30.229' in your directory > and add a user with the id="300" attribute > and you must configure your device to use the proper domain in > it's authentication credentials. > > I have defined domain in vars.xml as: > > > > > > > > On Thu, Apr 11, 2013 at 2:05 PM, Steven Ayre > wrote: > > Are you sure that FS is listening on that port on that > specific ip? Any firewall that might be blocking the packets? > > Run: > sofia global siptrace on > > That'll include the sip trace in the FS debug logs, confirming > whether FS is actually receiving the packets. > > Steve > > > > On 10 Apr 2013, at 11:28, Ashish gautam > wrote: > >> FS is not replying anything there are just subscription and >> registration packets. >> >> Thanks. >> >> On Wed, Apr 10, 2013 at 3:51 PM, Miha > > wrote: >> >> OK, just must post whole sip trace. This is just >> registration packet from blink/uac, I can not see what FS >> replied on this. >> >> miha >> >> Dne 4/10/2013 12:00 PM, pis(e Ashish gautam: >>> Hi Miha, >>> >>> This is the registration packet: >>> >>> 15:29:25.509003 IP (tos 0x0, ttl 128, id 31452, offset >>> 0, flags [none], proto UDP (17), length 525) >>> 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, >>> length 497 >>> z.....lK >>> ... >>> .........8%REGISTER sip:10.1.30.229 SIP/2.0 >>> Via: SIP/2.0/UDP >>> 10.1.30.210:53652;rport;branch=z9hG4bKPj9ccb691852ca419ea305c4cdd79aba44 >>> Max-Forwards: 70 >>> From: "Ashish" >> >;tag=5970e44547c642a2b6e335b5bbee73ea >>> To: "Ashish" >> > >>> Contact: >> >;+sip.instance="" >>> Call-ID: a88024e2bdd04101a57f713d57af681e >>> CSeq: 1 REGISTER >>> Expires: 3600 >>> User-Agent: Blink 0.2.10 (Windows) >>> Content-Length: 0 >>> >>> >>> On Wed, Apr 10, 2013 at 3:14 PM, Miha >> > wrote: >>> >>> Hi, >>> >>> what about registration packet and etc? >>> >>> this is only subscribe for presence. >>> >>> miha >>> >>> Dne 4/10/2013 11:07 AM, pis(e Ashish gautam: >>>> Miha, >>>> >>>> This is the output of the tcpdump -nq -s 0 -A -vvv >>>> -i eth0 port 5060 for the related packet : >>>> >>>> 13:29:37.050278 IP (tos 0x0, ttl 128, id 9943, >>>> offset 0, flags [none], proto UDP (17), length 640) >>>> 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] >>>> UDP, length 612 >>>> E...&....... >>>> ... >>>> ........l..SUBSCRIBE sip:1100 at 10.1.30.229 >>>> SIP/2.0 >>>> Via: SIP/2.0/UDP >>>> 10.1.30.210:53652;rport;branch=z9hG4bKPj91db66336d564de5b3b3e50c03ac32b3 >>>> Max-Forwards: 70 >>>> From: "Ashish" >>> >;tag=3058239a404c4a4dbcd438916aa8033b >>>> To: >>> > >>>> Contact: >>> > >>>> Call-ID: b3d45216211242afa13ae86734ea4408 >>>> CSeq: 30541 SUBSCRIBE >>>> Event: message-summary >>>> Expires: 3600 >>>> Accept: application/simple-message-summary >>>> Allow-Events: conference, message-summary, dialog, >>>> presence, presence.winfo, xcap-diff, dialog.winfo, >>>> refer >>>> User-Agent: Blink 0.2.10 (Windows) >>>> Content-Length: 0 >>>> >>>> >>>> >>>> >>>> On Wed, Apr 10, 2013 at 12:24 PM, Miha >>>> > wrote: >>>> >>>> did you try to use whireshark? >>>> >>>> miha >>>> >>>> Dne 4/10/2013 8:28 AM, pis(e Ashish gautam: >>>>> Thanks Miha, >>>>> >>>>> sip trace shows nothing. >>>>> >>>>> >>>>> On Wed, Apr 10, 2013 at 11:52 AM, Miha >>>>> > wrote: >>>>> >>>>> >>>>> Hi, >>>>> >>>>> was does sip trace say. You must add user >>>>> to default directory and that do reloadxml. >>>>> >>>>> You can past you log to see what is going on.. >>>>> >>>>> miha >>>>> >>>>> Dne 4/10/2013 8:09 AM, pis(e Ashish gautam: >>>>>> Hi, >>>>>> >>>>>> I am trying to register Default SIP users >>>>>> i.e. 1019,1004 etc from a softphone but >>>>>> they are not registering. Sofia status >>>>>> also not showing any user registrations. >>>>>> FreeSWITCH is listening on port 5060 and >>>>>> every configuration has been done >>>>>> according to the docs. What could be the >>>>>> issue? >>>>>> >>>>>> Any help appreciated. >>>>>> >>>>>> Thanks in advance. >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel >>>>> Communication Server >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel >>>> Communication Server >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>> Server >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 did you define user 300 in default directory with pass and etc? after doing this do reloadxml miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/b0c17423/attachment-0001.html From ashish at nms.co.in Wed Apr 10 16:28:18 2013 From: ashish at nms.co.in (Ashish gautam) Date: Wed, 10 Apr 2013 17:58:18 +0530 Subject: [Freeswitch-users] Default SIP users not registering In-Reply-To: <516559F5.1040208@softnet.si> References: <5165051B.5050906@softnet.si> <51650CA2.8090106@softnet.si> <51653477.5050407@softnet.si> <51653D1C.1040604@softnet.si> <516559F5.1040208@softnet.si> Message-ID: Miha, I did this. On Wed, Apr 10, 2013 at 5:54 PM, Miha wrote: > Dne 4/10/2013 2:11 PM, pi?e Steven Ayre: > > Those settings don't have any effect on their own. They set variables > which are used elsewhere in the config files. > > At least that suggests FS is indeed seeing the packets. Though that also > means you should get a response too (probably 403 Forbidden) > > -Steve > > > On 10 April 2013 11:46, Ashish gautam wrote: > >> Hi Steve, >> >> The domain is not defined according to the logs, I am getting this >> warning: >> >> 2013-04-10 16:13:18.614508 [WARNING] sofia_reg.c:2506 Can't find user [ >> 300 at 10.1.30.229] >> You must define a domain called '10.1.30.229' in your directory and add a >> user with the id="300" attribute >> and you must configure your device to use the proper domain in it's >> authentication credentials. >> >> I have defined domain in vars.xml as: >> >> >> >> >> >> >> >> On Thu, Apr 11, 2013 at 2:05 PM, Steven Ayre wrote: >> >>> Are you sure that FS is listening on that port on that specific ip? >>> Any firewall that might be blocking the packets? >>> >>> Run: >>> sofia global siptrace on >>> >>> That'll include the sip trace in the FS debug logs, confirming whether >>> FS is actually receiving the packets. >>> >>> Steve >>> >>> >>> >>> On 10 Apr 2013, at 11:28, Ashish gautam wrote: >>> >>> FS is not replying anything there are just subscription and >>> registration packets. >>> >>> Thanks. >>> >>> On Wed, Apr 10, 2013 at 3:51 PM, Miha wrote: >>> >>>> OK, just must post whole sip trace. This is just registration packet >>>> from blink/uac, I can not see what FS replied on this. >>>> >>>> miha >>>> >>>> Dne 4/10/2013 12:00 PM, pi?e Ashish gautam: >>>> >>>> Hi Miha, >>>> >>>> This is the registration packet: >>>> >>>> 15:29:25.509003 IP (tos 0x0, ttl 128, id 31452, offset 0, flags >>>> [none], proto UDP (17), length 525) >>>> 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 497 >>>> z.....lK >>>> ... >>>> .........8%REGISTER sip:10.1.30.229 SIP/2.0 >>>> Via: SIP/2.0/UDP 10.1.30.210:53652 >>>> ;rport;branch=z9hG4bKPj9ccb691852ca419ea305c4cdd79aba44 >>>> Max-Forwards: 70 >>>> From: "Ashish" >>> >;tag=5970e44547c642a2b6e335b5bbee73ea >>>> To: "Ashish" >>>> Contact: >>> >;+sip.instance="" >>>> Call-ID: a88024e2bdd04101a57f713d57af681e >>>> CSeq: 1 REGISTER >>>> Expires: 3600 >>>> User-Agent: Blink 0.2.10 (Windows) >>>> Content-Length: 0 >>>> >>>> >>>> On Wed, Apr 10, 2013 at 3:14 PM, Miha wrote: >>>> >>>>> Hi, >>>>> >>>>> what about registration packet and etc? >>>>> >>>>> this is only subscribe for presence. >>>>> >>>>> miha >>>>> >>>>> Dne 4/10/2013 11:07 AM, pi?e Ashish gautam: >>>>> >>>>> Miha, >>>>> >>>>> This is the output of the tcpdump -nq -s 0 -A -vvv -i eth0 port >>>>> 5060 for the related packet : >>>>> >>>>> 13:29:37.050278 IP (tos 0x0, ttl 128, id 9943, offset 0, flags >>>>> [none], proto UDP (17), length 640) >>>>> 10.1.30.210.53652 > 10.1.30.229.sip: [udp sum ok] UDP, length 612 >>>>> E...&....... >>>>> ... >>>>> ........l..SUBSCRIBE sip:1100 at 10.1.30.229 SIP/2.0 >>>>> Via: SIP/2.0/UDP 10.1.30.210:53652 >>>>> ;rport;branch=z9hG4bKPj91db66336d564de5b3b3e50c03ac32b3 >>>>> Max-Forwards: 70 >>>>> From: "Ashish" >>>> >;tag=3058239a404c4a4dbcd438916aa8033b >>>>> To: >>>>> Contact: >>>>> Call-ID: b3d45216211242afa13ae86734ea4408 >>>>> CSeq: 30541 SUBSCRIBE >>>>> Event: message-summary >>>>> Expires: 3600 >>>>> Accept: application/simple-message-summary >>>>> Allow-Events: conference, message-summary, dialog, presence, >>>>> presence.winfo, xcap-diff, dialog.winfo, refer >>>>> User-Agent: Blink 0.2.10 (Windows) >>>>> Content-Length: 0 >>>>> >>>>> >>>>> >>>>> >>>>> On Wed, Apr 10, 2013 at 12:24 PM, Miha wrote: >>>>> >>>>>> did you try to use whireshark? >>>>>> >>>>>> miha >>>>>> >>>>>> Dne 4/10/2013 8:28 AM, pi?e Ashish gautam: >>>>>> >>>>>> Thanks Miha, >>>>>> >>>>>> sip trace shows nothing. >>>>>> >>>>>> >>>>>> On Wed, Apr 10, 2013 at 11:52 AM, Miha wrote: >>>>>> >>>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> was does sip trace say. You must add user to default directory and >>>>>>> that do reloadxml. >>>>>>> >>>>>>> You can past you log to see what is going on.. >>>>>>> >>>>>>> miha >>>>>>> >>>>>>> Dne 4/10/2013 8:09 AM, pi?e Ashish gautam: >>>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> I am trying to register Default SIP users i.e. 1019,1004 etc from >>>>>>> a softphone but they are not registering. Sofia status also not showing any >>>>>>> user registrations. FreeSWITCH is listening on port 5060 and every >>>>>>> configuration has been done according to the docs. What could be the issue? >>>>>>> >>>>>>> Any help appreciated. >>>>>>> >>>>>>> Thanks in advance. >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> 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 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at 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 >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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.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 > > did you define user 300 in default directory with pass and etc? > > after doing this do reloadxml > > miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/d4641aaf/attachment-0001.html From mike at jerris.com Wed Apr 10 16:30:36 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 Apr 2013 08:30:36 -0400 Subject: [Freeswitch-users] Javadoc for esl.jar In-Reply-To: References: Message-ID: 5 emails in less than 30 minutes, many of them being the exact same question is a little much. Please slow it down a bit and wait for answers before re-sending the same questions over and over again. On Apr 10, 2013, at 3:23 AM, Marwan Idriss wrote: > Greeting, > > Is their any Javadoc for esl.jar ? > > Regards > > Marwan From jeff at jefflenk.com Wed Apr 10 17:15:11 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 10 Apr 2013 06:15:11 -0700 (PDT) Subject: [Freeswitch-users] Build errors in 2010.express solution In-Reply-To: <1365576667168-7589585.post@n2.nabble.com> References: <1365285612276-7589476.post@n2.nabble.com> <1365424893457-7589513.post@n2.nabble.com> <1365576667168-7589585.post@n2.nabble.com> Message-ID: <1365599711613-7589611.post@n2.nabble.com> try this instead git clean -fdx This will remove any files from the tree that are not in git. There have been major changes to git head in the last week or so which require a complete rebuild (even more than a Clean solution). -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Build-errors-in-2010-express-solution-tp7589476p7589611.html Sent from the freeswitch-users mailing list archive at Nabble.com. From Hector.Geraldino at ipsoft.com Wed Apr 10 19:28:22 2013 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Wed, 10 Apr 2013 15:28:22 +0000 Subject: [Freeswitch-users] http://wiki.freeswitch.org/wiki/Java_ESL_Client, how I can make a connection before starting to handle Outbound connection ? In-Reply-To: References: Message-ID: This example you mention handles an outbound connection coming FROM FreeSWITCH server. What this means is that your application is *NOT* the one that connects to FreeSWITCH, but acts as a server waiting for connections coming from FreeSWITCH. When a call is received on FreeSWITCH, it will open a socket connection to your application and you'll have total control of the call. Please take a couple of minutes and read the wiki page where this is explained: http://wiki.freeswitch.org/wiki/Java_ESL_Client#Outbound_socket_client_usage In outbound mode, your code will be executed for every incoming call. If you want to use FreeSWITCH out of the context of an incoming call (like to originate new calls) then you must use it in inbound mode (see http://wiki.freeswitch.org/wiki/Java_ESL_Client#Inbound_client_usage). The library contains a very good set of unit tests (ClientTest.java) showing how to use it in inbound mode. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Marwan Idriss Sent: Wednesday, April 10, 2013 3:08 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] http://wiki.freeswitch.org/wiki/Java_ESL_Client, how I can make a connection before starting to handle Outbound connection ? Greeting, bellow is an example to handle outbound connection and make hung up: but it never show How to make the connection in first place is their an example in main of how this function work ? /* * Copyright 2010 david varnes. * * Licensed under the Apache License, version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at: * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. */ package org.freeswitch.esl.client.outbound.example; import org.freeswitch.esl.client.outbound.AbstractOutboundClientHandler; import org.freeswitch.esl.client.transport.SendMsg; import org.freeswitch.esl.client.transport.event.EslEvent; import org.freeswitch.esl.client.transport.message.EslMessage; import org.freeswitch.esl.client.transport.message.EslHeaders.Name; import org.jboss.netty.channel.Channel; import org.jboss.netty.channel.ChannelHandlerContext; /** * Simple example of a handler for outbound connection from FreeSWITCH server. * This class will log some of the FreeSWTICH call channel variables and * then hangup the call. * * @author david varnes */ public class SimpleHangupOutboundHandler extends AbstractOutboundClientHandler { @Override protected void handleConnectResponse( ChannelHandlerContext ctx, EslEvent event ) { log.info( "Received connect response [{}]", event ); if ( event.getEventName().equalsIgnoreCase( "CHANNEL_DATA" ) ) { // this is the response to the initial connect log.info( "======================= incoming channel data =============================" ); log.info( "Event-Date-Local: [{}]", event.getEventDateLocal() ); log.info( "Unique-ID: [{}]", event.getEventHeaders().get( "Unique-ID" ) ); log.info( "Channel-ANI: [{}]", event.getEventHeaders().get( "Channel-ANI" ) ); log.info( "Answer-State: [{}]", event.getEventHeaders().get( "Answer-State" ) ); log.info( "Caller-Destination-Number: [{}]", event.getEventHeaders().get( "Caller-Destination-Number" ) ); log.info( "======================= = = = = = = = = = = = =============================" ); // now hangup the call hangupCall( ctx.getChannel() ); } else { throw new IllegalStateException( "Unexpected event after connect: [" + event.getEventName() + ']' ); } } @Override protected void handleEslEvent( ChannelHandlerContext ctx, EslEvent event ) { log.info( "Received event [{}]", event ); } private void hangupCall( Channel channel ) { SendMsg hangupMsg = new SendMsg(); hangupMsg.addCallCommand( "execute" ); hangupMsg.addExecuteAppName( "hangup" ); EslMessage response = sendSyncMultiLineCommand( channel, hangupMsg.getMsgLines() ); if ( response.getHeaderValue( Name.REPLY_TEXT ).startsWith( "+OK" ) ) { log.info( "Call hangup successful" ); } else { log.error( "Call hangup failed: [{}}", response.getHeaderValue( Name.REPLY_TEXT ) ); } } } -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/8885bf10/attachment.html From stephen.thwaites at callstera.com Wed Apr 10 20:03:23 2013 From: stephen.thwaites at callstera.com (Stephen Thwaites) Date: Wed, 10 Apr 2013 18:03:23 +0200 Subject: [Freeswitch-users] default timeout In-Reply-To: References: Message-ID: Hello all, I still haven't managed to solve this one. Any further tips would be greatly appreciated. I feel that I am experimenting with various parameters without a full understanding of what the behavior should be. The A leg (caller) hits the dialplan in public, the public dialplan finds a match and transfers to the appropriate dialplan in corresponding context, this extension then uses 'set' or 'export' on the call_timeout variable and attempts a bridge. Does this 'set' set a timeout for the A leg or B leg? If I use 'export' does it set the timeout for both legs? Last of all does setting or exporting the call_timeout override the default bridge timeout of 60s? Many thanks. Regards, Steve. On Fri, Mar 29, 2013 at 4:50 PM, Stephen Thwaites wrote: > Hello, > I have setup this scenario now with a test case and have tried using > the 'export' application instead of 'set' for the call_timeout=25. > Sadly the same behavior. Similarly no change when using > hangup_after_bridge=true and continue_on_fail=true before each bridge > as suggested. Further I have tried using originate_timeout=25 and also > experimented with leg_timeouts. I then wondered if the voip provider > set a timeout on the incoming call and they do, but this is set to > 120s. I all tests I get an ORIGINATOR_CANCEL after 60s. > > I then did a sip trace and log level 6 from the fs_cli. Here is the (I > think) the relevant bit that shows that FS is sending a CANCEL to the > telephones. > > Any further help would be appreciated! > > Regards, > Steve. > > 2013-03-29 15:41:55.117929 [NOTICE] sofia.c:6379 Hangup > sofia/external/0610884128 at 91.195.160.3 [CS_EXECUTE] > [ORIGINATOR_CANCEL] > 2013-03-29 15:41:55.117929 [NOTICE] switch_ivr_originate.c:3349 Hangup > sofia/internal/sip:1000 at 192.168.1.113:45060 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > 2013-03-29 15:41:55.117929 [NOTICE] switch_ivr_originate.c:3349 Hangup > sofia/internal/sip:1001 at 192.168.1.112:35160 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > 2013-03-29 15:41:55.117929 [NOTICE] switch_ivr_originate.c:3349 Hangup > sofia/internal/sip:1002 at 192.168.1.10:49535 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > 2013-03-29 15:41:55.117929 [INFO] mod_dptools.c:3052 Originate Failed. > Cause: ORIGINATOR_CANCEL > 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1505 Session > 9911 (sofia/external/0610884128 at 91.195.160.3) Ended > 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1509 Close > Channel sofia/external/0610884128 at 91.195.160.3 [CS_DESTROY] > 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1505 Session > 9917 (sofia/internal/sip:1000 at 192.168.1.113:45060) Ended > 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1509 Close > Channel sofia/internal/sip:1000 at 192.168.1.113:45060 [CS_DESTROY] > send 429 bytes to udp/[80.101.42.120]:52292 at 14:43:27.409611: > ------------------------------------------------------------------------ > CANCEL sip:1000 at 192.168.1.113:45060 SIP/2.0 > Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK5H6X2DKX8Dj1N > Route: > Max-Forwards: 67 > From: "Thwaites, Stephen" > ;tag=1rv4jNU086aaQ > To: > Call-ID: d6322816-1321-1231-80a7-fa163e468f82 > CSeq: 41948035 CANCEL > Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" > Content-Length: 0 > > ------------------------------------------------------------------------ > 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1505 Session > 9918 (sofia/internal/sip:1001 at 192.168.1.112:35160) Ended > 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1509 Close > Channel sofia/internal/sip:1001 at 192.168.1.112:35160 [CS_DESTROY] > send 429 bytes to udp/[80.101.42.120]:42328 at 14:43:27.410670: > ------------------------------------------------------------------------ > CANCEL sip:1001 at 192.168.1.112:35160 SIP/2.0 > Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK6tZp48305p8KH > Route: > Max-Forwards: 67 > From: "Thwaites, Stephen" > ;tag=21NXmgc45F1vj > To: > Call-ID: d6335070-1321-1231-80a7-fa163e468f82 > CSeq: 41948035 CANCEL > Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" > Content-Length: 0 > > ------------------------------------------------------------------------ > 2013-03-29 15:41:55.127924 [NOTICE] switch_core_session.c:1505 Session > 9919 (sofia/internal/sip:1002 at 192.168.1.10:49535) Ended > 2013-03-29 15:41:55.127924 [NOTICE] switch_core_session.c:1509 Close > Channel sofia/internal/sip:1002 at 192.168.1.10:49535 [CS_DESTROY] > send 427 bytes to udp/[80.101.42.120]:50175 at 14:43:27.413100: > ------------------------------------------------------------------------ > CANCEL sip:1002 at 192.168.1.10:49535 SIP/2.0 > Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK73rF63m42Zy6c > Route: > Max-Forwards: 67 > From: "Thwaites, Stephen" > ;tag=3aFppBX72rQFe > To: > Call-ID: d634a26a-1321-1231-80a7-fa163e468f82 > CSeq: 41948035 CANCEL > Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 531 bytes from udp/[80.101.42.120]:50175 at 14:43:27.440064: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 94.100.113.237;rport=5060;branch=z9hG4bK73rF63m42Zy6c > From: "Thwaites, Stephen" > ;tag=3aFppBX72rQFe > To: ;tag=942234751 > Call-ID: d634a26a-1321-1231-80a7-fa163e468f82 > CSeq: 41948035 CANCEL > Contact: > Supported: replaces, path, timer, eventlist > User-Agent: Grandstream GXP2200 1.0.1.40 > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, > REFER, UPDATE, MESSAGE > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 507 bytes from udp/[80.101.42.120]:50175 at 14:43:27.445183: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 94.100.113.237;rport=5060;branch=z9hG4bK73rF63m42Zy6c > From: "Thwaites, Stephen" > ;tag=3aFppBX72rQFe > To: ;tag=942234751 > Call-ID: d634a26a-1321-1231-80a7-fa163e468f82 > CSeq: 41948035 INVITE > Supported: replaces, path, timer, eventlist > User-Agent: Grandstream GXP2200 1.0.1.40 > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, > REFER, UPDATE, MESSAGE > Content-Length: 0 > > ------------------------------------------------------------------------ > send 387 bytes to udp/[80.101.42.120]:50175 at 14:43:27.445316: > ------------------------------------------------------------------------ > ACK sip:1002 at 192.168.1.10:49535 SIP/2.0 > Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK73rF63m42Zy6c > Route: > Max-Forwards: 67 > From: "Thwaites, Stephen" > ;tag=3aFppBX72rQFe > To: ;tag=942234751 > Call-ID: d634a26a-1321-1231-80a7-fa163e468f82 > CSeq: 41948035 ACK > Content-Length: 0 > > etc... > > On Tue, Mar 26, 2013 at 12:22 AM, Stephen Thwaites > wrote: >> Nick, Michael, >> Thanks for the advise, will give these a try and feed back to the list. >> >> Regards, >> Steve. >> >> On Mon, Mar 25, 2013 at 11:21 PM, Michael Collins wrote: >>> Try "export" instead of "set" on your call_timeout=25 lines. >>> -MC >>> >>> On Fri, Mar 22, 2013 at 8:32 AM, Stephen Thwaites >>> wrote: >>>> >>>> Hi All, >>>> Apologies for the simple question but I can't find the answer anywhere >>>> in the books, wiki, or our friend google. >>>> >>>> If somebody calls us, FS does an ORIGINATE_CANCEL after 60s but the >>>> follow-me scheme we have configured is100s. How can I increase the >>>> default call_timeout of 60s to 100s? Or maybe I am just doing >>>> something wrong! >>>> >>>> I have tried leg_timeouts, originate_timeouts both on the transfer and >>>> the bridge as well to no avail? >>>> >>>> Would be very grateful for any help or advise. >>>> >>>> Regards, >>>> Steve >>>> >>>> Some Details: >>>> - External call comes in on external profile from our voip provider. >>>> - Dialplan in the public context does a transfer to a follow-me >>>> extension 7777 in context creche-babys >>> data="7777 XML creche-babys "/> >>>> - The 7777 extension is as follows and the call hangs up part way >>>> through the third step if nobody picks up (after 60s total). >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> >>>> data="{ignore_early_media=true}user/21@${domain_name},user/20@${domain_name}"/> >>>> >>>> >>>> >>> data="{ignore_early_media=true}user/22@${domain_name}"/> >>>> >>>> >>>> >>> >>>> data={ignore_early_media=true}user/23@${domain_name},user/24@${domain_name},user/26@${domain_name},user/27@${domain_name},user/28@${domain_name}"/> >>>> >>>> >>>> >>>> >>> data="{ignore_early_media=true}sofia/gateway/3120 >>>> ...voipprovidergateway/06 ...mobile number"/> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.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 msc at freeswitch.org Wed Apr 10 20:43:50 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 10 Apr 2013 09:43:50 -0700 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Message-ID: Hello all! Come join the FreeSWITCH conference call at 1PM EDT/10AM PDT: http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10 We have a few items to discuss, and we'll be asking the community for input on an important question: How do you choose whether to do closed, freemium, or open source when you've built a great solution? Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/1fde6624/attachment.html From cal.leeming at simplicitymedialtd.co.uk Wed Apr 10 20:54:35 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 10 Apr 2013 17:54:35 +0100 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: This number no longer works; UK +44-3300-100-295 Cal On Wed, Apr 10, 2013 at 5:43 PM, Michael Collins wrote: > Hello all! > > Come join the FreeSWITCH conference call at 1PM EDT/10AM PDT: > > http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10 > > We have a few items to discuss, and we'll be asking the community for > input on an important question: > How do you choose whether to do closed, freemium, or open source when > you've built a great solution? > > Talk to you soon! > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.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/20130410/e26e3562/attachment.html From steveayre at gmail.com Wed Apr 10 21:20:42 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 10 Apr 2013 18:20:42 +0100 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: Ah, that's annoying... sounds like it's been reassigned to someone else. Initially it was set up by me, but then numbergroup.com changed their fees so it was no longer free. One of their support guys kindly took over the number to continue it for free but I guess that's no longer possible. I'll remove it from the Wiki and see if we can set up one from UKDDI. -Steve On 10 April 2013 17:54, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > This number no longer works; > > UK +44-3300-100-295 > > Cal > > On Wed, Apr 10, 2013 at 5:43 PM, Michael Collins wrote: > >> Hello all! >> >> Come join the FreeSWITCH conference call at 1PM EDT/10AM PDT: >> >> http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10 >> >> We have a few items to discuss, and we'll be asking the community for >> input on an important question: >> How do you choose whether to do closed, freemium, or open source when >> you've built a great solution? >> >> Talk to you soon! >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.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/20130410/b8a72482/attachment.html From cal.leeming at simplicitymedialtd.co.uk Wed Apr 10 21:26:32 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 10 Apr 2013 18:26:32 +0100 Subject: [Freeswitch-users] Screen share (bbb) Message-ID: Hello, As discussed on the conference, we need to test the screenshare (bbb) system with as many people as possible at the same time. This is running at; http://bbb.tollfreegateway.com / Could we get 10 volunteers who would be willing to participate in this test, ideally sometime this week or next week so we can get it signed off. Any other alternative suggestions for screensharing solution is also welcome. Thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/736cafaf/attachment.html From steveayre at gmail.com Wed Apr 10 21:28:06 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 10 Apr 2013 18:28:06 +0100 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: There's now a UKDDI number on the Wiki instead... I couldn't see 03 prefixes on their list (national local rate) and they seem to be out of London numbers so I picked ended up York. -Steve On 10 April 2013 18:20, Steven Ayre wrote: > Ah, that's annoying... sounds like it's been reassigned to someone else. > > Initially it was set up by me, but then numbergroup.com changed their > fees so it was no longer free. One of their support guys kindly took over > the number to continue it for free but I guess that's no longer possible. > > I'll remove it from the Wiki and see if we can set up one from UKDDI. > > -Steve > > > On 10 April 2013 17:54, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> This number no longer works; >> >> UK +44-3300-100-295 >> >> Cal >> >> On Wed, Apr 10, 2013 at 5:43 PM, Michael Collins wrote: >> >>> Hello all! >>> >>> Come join the FreeSWITCH conference call at 1PM EDT/10AM PDT: >>> >>> http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10 >>> >>> We have a few items to discuss, and we'll be asking the community for >>> input on an important question: >>> How do you choose whether to do closed, freemium, or open source when >>> you've built a great solution? >>> >>> Talk to you soon! >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.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/20130410/2d6d97fd/attachment-0001.html From krice at freeswitch.org Wed Apr 10 21:31:09 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 10 Apr 2013 12:31:09 -0500 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: Message-ID: Hey Steven, I?ll get us one from a FreeSWITCH loving provider K On 4/10/13 12:20 PM, "Steven Ayre" wrote: > Ah, that's annoying... sounds like it's been reassigned to someone else. > > Initially it was set up by me, but then numbergroup.com > changed their fees so it was no longer free. One of > their support guys kindly took over the number to continue it for free but I > guess that's no longer possible. > > I'll remove it from the Wiki and see if we can set up one from UKDDI. > > -Steve > > > On 10 April 2013 17:54, Cal Leeming [Simplicity Media Ltd] > wrote: >> This number no longer works; >> >> UK +44-3300-100-295 >> >> Cal >> >> On Wed, Apr 10, 2013 at 5:43 PM, Michael Collins wrote: >>> Hello all! >>> >>> Come join the FreeSWITCH conference call at 1PM EDT/10AM PDT: >>> >>> http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10 >>> >>> We have a few items to discuss, and we'll be asking the community for input >>> on an important question: >>> How do you choose whether to do closed, freemium, or open source when you've >>> built a great solution? >>> >>> Talk to you soon! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/02c29ec6/attachment.html From jaykris at gmail.com Wed Apr 10 21:43:38 2013 From: jaykris at gmail.com (JP) Date: Wed, 10 Apr 2013 10:43:38 -0700 Subject: [Freeswitch-users] How do I call a Lua script from a Java application? Message-ID: I have a Java application running at startup, through the java.conf.xml configuration. I want to call a LUA script from it. Is there a way to do it? Thanks JP -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/6d901c4a/attachment.html From avi at avimarcus.net Wed Apr 10 21:46:36 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 10 Apr 2013 20:46:36 +0300 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: I still have some london UKDDI numbers if that helps, e.g.: 44-203-298-5931 Should I set it up? -Avi On Wed, Apr 10, 2013 at 8:31 PM, Ken Rice wrote: > Hey Steven, I?ll get us one from a FreeSWITCH loving provider > > K > > > > On 4/10/13 12:20 PM, "Steven Ayre" wrote: > > Ah, that's annoying... sounds like it's been reassigned to someone else. > > Initially it was set up by me, but then numbergroup.com < > http://numbergroup.com> changed their fees so it was no longer free. One > of their support guys kindly took over the number to continue it for free > but I guess that's no longer possible. > > > I'll remove it from the Wiki and see if we can set up one from UKDDI. > > -Steve > > > On 10 April 2013 17:54, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > This number no longer works; > > UK +44-3300-100-295 > > Cal > > On Wed, Apr 10, 2013 at 5:43 PM, Michael Collins > wrote: > > Hello all! > > Come join the FreeSWITCH conference call at 1PM EDT/10AM PDT: > > http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10 > > We have a few items to discuss, and we'll be asking the community for > input on an important question: > How do you choose whether to do closed, freemium, or open source when > you've built a great solution? > > Talk to you soon! > > > -- > Ken > > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > * > irc.freenode.net #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/20130410/8ea86c5b/attachment.html From steveayre at gmail.com Wed Apr 10 21:54:24 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 10 Apr 2013 18:54:24 +0100 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: There's a York UKDDI one there now, but why not add London too? 0330 or 0333 would be ideal since they're local rate from anywhere in the country, even from a mobile. -Steve On 10 April 2013 18:46, Avi Marcus wrote: > I still have some london UKDDI numbers if that helps, e.g.: 44-203-298-5931 > Should I set it up? > > -Avi > > > On Wed, Apr 10, 2013 at 8:31 PM, Ken Rice wrote: > >> Hey Steven, I?ll get us one from a FreeSWITCH loving provider >> >> K >> >> >> >> On 4/10/13 12:20 PM, "Steven Ayre" wrote: >> >> Ah, that's annoying... sounds like it's been reassigned to someone else. >> >> Initially it was set up by me, but then numbergroup.com < >> http://numbergroup.com> changed their fees so it was no longer free. >> One of their support guys kindly took over the number to continue it for >> free but I guess that's no longer possible. >> >> >> I'll remove it from the Wiki and see if we can set up one from UKDDI. >> >> -Steve >> >> >> On 10 April 2013 17:54, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >> This number no longer works; >> >> UK +44-3300-100-295 >> >> Cal >> >> On Wed, Apr 10, 2013 at 5:43 PM, Michael Collins >> wrote: >> >> Hello all! >> >> Come join the FreeSWITCH conference call at 1PM EDT/10AM PDT: >> >> http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10 >> >> We have a few items to discuss, and we'll be asking the community for >> input on an important question: >> How do you choose whether to do closed, freemium, or open source when >> you've built a great solution? >> >> Talk to you soon! >> >> >> -- >> Ken >> >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> * >> irc.freenode.net #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/20130410/56ff4b36/attachment-0001.html From anthony.minessale at gmail.com Wed Apr 10 21:55:38 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 10 Apr 2013 12:55:38 -0500 Subject: [Freeswitch-users] Sound Not Ok(Choppy Sound) After 250 Ports In-Reply-To: References: Message-ID: no need to cross-post On Wed, Apr 10, 2013 at 2:43 AM, Subhash wrote: > Hi, > > I ran the load test, raising the port count slowly (every 5 to 10 minutes > or so) to 100, 150, 200, 250, 275 and 300 ports. > There were no audio problems until I reached 275 ports. After probably 10 > minutes at 275 ports, audio problems started, and after that they were > consistent. > Of course I can?t guarantee that there were no audio problems at lower > port counts - if the problem only occurred intermittently I might not have > caught it or I might not have waited long enough. > Once the audio problems started, I slowly lowered the number of ports back > to 250 ports. The audio problems disappeared after about 2 1/2 minutes at > 250 ports. > I then increased the number of ports back to 275 ports. > > Thanks, > Subhash > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130410/fa417bb7/attachment.html From andrew at cassidywebservices.co.uk Wed Apr 10 22:06:36 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 10 Apr 2013 19:06:36 +0100 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: I'll see what I've got, I'll get back to you. On 10 April 2013 18:54, Steven Ayre wrote: > There's a York UKDDI one there now, but why not add London too? > > 0330 or 0333 would be ideal since they're local rate from anywhere in the > country, even from a mobile. > > -Steve > > > On 10 April 2013 18:46, Avi Marcus wrote: > >> I still have some london UKDDI numbers if that helps, e.g.: >> 44-203-298-5931 >> Should I set it up? >> >> -Avi >> >> >> On Wed, Apr 10, 2013 at 8:31 PM, Ken Rice wrote: >> >>> Hey Steven, I?ll get us one from a FreeSWITCH loving provider >>> >>> K >>> >>> >>> >>> On 4/10/13 12:20 PM, "Steven Ayre" wrote: >>> >>> Ah, that's annoying... sounds like it's been reassigned to someone else. >>> >>> Initially it was set up by me, but then numbergroup.com < >>> http://numbergroup.com> changed their fees so it was no longer free. >>> One of their support guys kindly took over the number to continue it for >>> free but I guess that's no longer possible. >>> >>> >>> I'll remove it from the Wiki and see if we can set up one from UKDDI. >>> >>> -Steve >>> >>> >>> On 10 April 2013 17:54, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>> This number no longer works; >>> >>> UK +44-3300-100-295 >>> >>> Cal >>> >>> On Wed, Apr 10, 2013 at 5:43 PM, Michael Collins >>> wrote: >>> >>> Hello all! >>> >>> Come join the FreeSWITCH conference call at 1PM EDT/10AM PDT: >>> >>> http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10 >>> >>> We have a few items to discuss, and we'll be asking the community for >>> input on an important question: >>> How do you choose whether to do closed, freemium, or open source when >>> you've built a great solution? >>> >>> Talk to you soon! >>> >>> >>> -- >>> Ken >>> >>> *http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> * >>> irc.freenode.net #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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/bc96afb6/attachment.html From andrew at cassidywebservices.co.uk Wed Apr 10 22:09:48 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 10 Apr 2013 19:09:48 +0100 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: Bah none of my 03's are on IPEX yet, 03300100290 will do for now. Should be able to guess the upstream provider of that... On 10 April 2013 19:06, Andrew Cassidy wrote: > I'll see what I've got, I'll get back to you. > > > On 10 April 2013 18:54, Steven Ayre wrote: > >> There's a York UKDDI one there now, but why not add London too? >> >> 0330 or 0333 would be ideal since they're local rate from anywhere in the >> country, even from a mobile. >> >> -Steve >> >> >> On 10 April 2013 18:46, Avi Marcus wrote: >> >>> I still have some london UKDDI numbers if that helps, e.g.: >>> 44-203-298-5931 >>> Should I set it up? >>> >>> -Avi >>> >>> >>> On Wed, Apr 10, 2013 at 8:31 PM, Ken Rice wrote: >>> >>>> Hey Steven, I?ll get us one from a FreeSWITCH loving provider >>>> >>>> K >>>> >>>> >>>> >>>> On 4/10/13 12:20 PM, "Steven Ayre" wrote: >>>> >>>> Ah, that's annoying... sounds like it's been reassigned to someone else. >>>> >>>> Initially it was set up by me, but then numbergroup.com < >>>> http://numbergroup.com> changed their fees so it was no longer free. >>>> One of their support guys kindly took over the number to continue it for >>>> free but I guess that's no longer possible. >>>> >>>> >>>> I'll remove it from the Wiki and see if we can set up one from UKDDI. >>>> >>>> -Steve >>>> >>>> >>>> On 10 April 2013 17:54, Cal Leeming [Simplicity Media Ltd] < >>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>> >>>> This number no longer works; >>>> >>>> UK +44-3300-100-295 >>>> >>>> Cal >>>> >>>> On Wed, Apr 10, 2013 at 5:43 PM, Michael Collins >>>> wrote: >>>> >>>> Hello all! >>>> >>>> Come join the FreeSWITCH conference call at 1PM EDT/10AM PDT: >>>> >>>> http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10 >>>> >>>> We have a few items to discuss, and we'll be asking the community for >>>> input on an important question: >>>> How do you choose whether to do closed, freemium, or open source when >>>> you've built a great solution? >>>> >>>> Talk to you soon! >>>> >>>> >>>> -- >>>> Ken >>>> >>>> *http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> * >>>> irc.freenode.net #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 >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/bc6b83f0/attachment-0001.html From avi at avimarcus.net Wed Apr 10 22:26:06 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 10 Apr 2013 21:26:06 +0300 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: OK, I routed and added the 44-203-298-5931 number to the wiki. I also added "routed by" just so we know who to contact about it... -Avi Marcus On Wed, Apr 10, 2013 at 9:09 PM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > Bah none of my 03's are on IPEX yet, 03300100290 will do for now. Should > be able to guess the upstream provider of that... > > > On 10 April 2013 19:06, Andrew Cassidy wrote: > >> I'll see what I've got, I'll get back to you. >> >> >> On 10 April 2013 18:54, Steven Ayre wrote: >> >>> There's a York UKDDI one there now, but why not add London too? >>> >>> 0330 or 0333 would be ideal since they're local rate from anywhere in >>> the country, even from a mobile. >>> >>> -Steve >>> >>> >>> On 10 April 2013 18:46, Avi Marcus wrote: >>> >>>> I still have some london UKDDI numbers if that helps, e.g.: >>>> 44-203-298-5931 >>>> Should I set it up? >>>> >>>> -Avi >>>> >>>> >>>> On Wed, Apr 10, 2013 at 8:31 PM, Ken Rice wrote: >>>> >>>>> Hey Steven, I?ll get us one from a FreeSWITCH loving provider >>>>> >>>>> K >>>>> >>>>> >>>>> >>>>> On 4/10/13 12:20 PM, "Steven Ayre" wrote: >>>>> >>>>> Ah, that's annoying... sounds like it's been reassigned to someone >>>>> else. >>>>> >>>>> Initially it was set up by me, but then numbergroup.com < >>>>> http://numbergroup.com> changed their fees so it was no longer free. >>>>> One of their support guys kindly took over the number to continue it for >>>>> free but I guess that's no longer possible. >>>>> >>>>> >>>>> I'll remove it from the Wiki and see if we can set up one from UKDDI. >>>>> >>>>> -Steve >>>>> >>>>> >>>>> On 10 April 2013 17:54, Cal Leeming [Simplicity Media Ltd] < >>>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>>> >>>>> This number no longer works; >>>>> >>>>> UK +44-3300-100-295 >>>>> >>>>> Cal >>>>> >>>>> On Wed, Apr 10, 2013 at 5:43 PM, Michael Collins >>>>> wrote: >>>>> >>>>> Hello all! >>>>> >>>>> Come join the FreeSWITCH conference call at 1PM EDT/10AM PDT: >>>>> >>>>> http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10 >>>>> >>>>> We have a few items to discuss, and we'll be asking the community for >>>>> input on an important question: >>>>> How do you choose whether to do closed, freemium, or open source when >>>>> you've built a great solution? >>>>> >>>>> Talk to you soon! >>>>> >>>>> >>>>> -- >>>>> Ken >>>>> >>>>> *http://www.FreeSWITCH.org >>>>> http://www.ClueCon.com >>>>> http://www.OSTAG.org >>>>> * >>>>> irc.freenode.net #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 >>> >>> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 *F >> *03300 100 961 >> *E *andrew at cassidywebservices.co.uk >> *W *www.cassidywebservices.co.uk >> > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/5a608a48/attachment.html From jkr888 at gmail.com Wed Apr 10 23:34:43 2013 From: jkr888 at gmail.com (Johny Kadarisman Kwan) Date: Wed, 10 Apr 2013 15:34:43 -0400 Subject: [Freeswitch-users] Stale channel that cannot be killed, session created from a running Lua scripts. Message-ID: I have lua 'background' script that wake up every few seconds, and check to perform specific task, one of it to established call. Calls are then initiated then transfer to some extension in dialplan. The scripts has been running and works as expected. But after session ended(hangup), channels is not released and can be seen using cli "show channels". and trying to kill such channel result in "-ERR No Such Channel!" This seems being reported before, but I can't seems to find any resolution. Following simple script could replicate the issues : freeswitch> luarun SimpleDialer.lua =================== SimpleDialer.lua local threadName = "SimpleDialer" while true do freeswitch.consoleLog("info", threadName.." ticking\n") -- task run here local new_session = freeswitch.Session('sofia/gateway/vox/+150<%2B15085895115> 88888888') new_session:transfer("echo", "XML", "default") freeswitch.msleep(60000) end ============= dialplan entry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/38042908/attachment.html From krice at freeswitch.org Thu Apr 11 00:10:39 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 10 Apr 2013 15:10:39 -0500 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: Message-ID: Alex over at Ziron.net has us fixed up with a national DDI +44-330-320-0105 And its already routed, adding it to the wiki On 4/10/13 1:26 PM, "Avi Marcus" wrote: > OK, I routed and added the?44-203-298-5931 number to the wiki. > I also added "routed by" just so we know who to contact about it... > > -Avi Marcus > > On Wed, Apr 10, 2013 at 9:09 PM, Andrew Cassidy > wrote: >> Bah none of my 03's are on IPEX yet,?03300100290 will do for now. Should be >> able to guess the upstream provider of that... >> >> >> On 10 April 2013 19:06, Andrew Cassidy >> wrote: >>> I'll see what I've got, I'll get back to you. >>> >>> >>> On 10 April 2013 18:54, Steven Ayre wrote: >>>> There's a York UKDDI one there now, but why not add London too? >>>> >>>> 0330 or 0333 would be ideal since they're local rate from anywhere in the >>>> country, even from a mobile. >>>> >>>> -Steve >>>> >>>> >>>> On 10 April 2013 18:46, Avi Marcus wrote: >>>>> I still have some london UKDDI numbers if that helps, >>>>> e.g.:?44-203-298-5931 >>>>> Should I set it up? >>>>> >>>>> -Avi >>>>> >>>>> >>>>> On Wed, Apr 10, 2013 at 8:31 PM, Ken Rice wrote: >>>>>> Hey Steven, I?ll get us one from a FreeSWITCH loving provider >>>>>> >>>>>> K >>>>>> >>>>>> >>>>>> >>>>>> On 4/10/13 12:20 PM, "Steven Ayre" >>>>> > wrote: >>>>>> >>>>>>> Ah, that's annoying... sounds like it's been reassigned to someone else. >>>>>>> >>>>>>> Initially it was set up by me, but then numbergroup.com >>>>>>> ?changed their fees >>>>>>> so it was no longer free. One of their support guys kindly took over the >>>>>>> number to continue it for free but I guess that's no longer possible. >>>>>>> >>>>>>> >>>>>>> I'll remove it from the Wiki and see if we can set up one from UKDDI. >>>>>>> >>>>>>> -Steve >>>>>>> >>>>>>> >>>>>>> On 10 April 2013 17:54, Cal Leeming [Simplicity Media Ltd] >>>>>>> >>>>>> > wrote: This number no longer works; UK +44-3300-100-295 Cal On Wed, Apr 10, 2013 at 5:43 PM, Michael Collins > wrote: Hello all! Come join the FreeSWITCH conference call at 1PM EDT/10AM PDT: http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10 We have a few items to discuss, and we'll be asking the community for input on an important question: How do you choose whether to do closed, freemium, or open source when you've built a great solution? Talk to you soon! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/42c9cb7b/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Thu Apr 11 00:12:53 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 10 Apr 2013 21:12:53 +0100 Subject: [Freeswitch-users] Metaswitch perimeta SBC - any thoughts? In-Reply-To: References: <44A43FCD-DBAC-4961-9C64-251046D4E0E3@jerris.com> Message-ID: I'm going to be doing some more tests on this over the next few days. From what I can tell, the approach we're trying to use does look sane, it's working under initial prototypes (benchmarks tbc), and does almost exactly what we require it to do under the development constraints we have (e.g. not having to worry about installing individual licenses on developers machines for propriety s/w such as F5, and being cloud deployable, i.e. not a physical solution as this would mean giving each developer their own physical box, or having development run over corp WAN rather than local VM LAN etc.) The term 'SBC' seems to be thrown around quite wildly and can mean different things to different people.. Some SBCs have minimal functionality, others implement a full range of features including LCR. I've come to the conclusion that if it does what we require, and it can scale to the numbers we need it to, then it will probably work fine for us. However, I would still be interested in hearing from others on large scale approaches, specifically ones where you can scale out, rather than up (i.e. you can throw a lot of servers at it, rather than having to increase the resources of just a single server). Cal On Tue, Apr 9, 2013 at 6:41 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > We attempted to use FreeSWITCH as an SBC but came up against problems > scaling. > > Instead my idea was to place a dumb proxy in front of lots of FreeSWITCH > instances, with two profiles exposed.. one for UC registrations, and > another for the backend FS instances. When a UC sends a register, it > forwards the request to a backend, if it was successful then a persist > entry is created. If a user tried to call another user in the same domain > (or another domain that we hold), it would send the request through to the > SBC, the SBC would then check the persistance table for an entry for that > user, and redirect the query to the appropriate server where the user was > currently registered. If no entry was found, a random server would be > picked to handle voicemail and redirects etc. Any INVITEs would also have > their SIP/SDP re-written to allow RTP proxying as well for full topology > hiding. This allows a single domain to be spread across an unlimited amount > of servers. In theory the bottleneck would be the speed of which the SBC > can proxy packets.. initial tests show that a python prototype can forward > approx 350mbit/sec on a single thread (using eventlet) before it starts to > choke, and we were able to saturate a gbit link with zero packet loss and > jitter using 8 threads (that's with full SIP/SDP parsing too). We've tested > using the F5 and Stingray too, but their throughput isn't as good. > > However, my concern is that no one (from what I can tell) is using this > approach to scale voice platforms, and this makes me question whether or > not it is the correct thing to do. > > Cal > > On Tue, Apr 9, 2013 at 6:20 PM, Michael Jerris wrote: > >> Can you describe a bit more what exactly you want to get out of an SBC >> for your setup? This topic is totally fine for the list >> >> Mike >> >> >> On Apr 9, 2013, at 1:07 PM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >> Hello all, >> >> Does anyone here have any first hand experience using the Metaswitch >> perimeta SBC with FreeSWITCH? >> http://www.metaswitch.com/products/perimeta-session-border-controller >> >> I know this isn't strictly on-topic, but I find the people on this list >> are usually best placed to give useful feedback on related products. >> >> At the moment we're weighing up the pros/cons of using our own in-house >> SBC, F5 LTM, Stingray TM, OpenSER, or buying a purpose made SBC such as the >> Perimeta. >> >> Any thoughts? >> >> Cal >> >> (Mods: if you aren't happy with this being discussed on the list, just >> let me know). >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/3ab6ae68/attachment.html From anthony.minessale at gmail.com Thu Apr 11 00:17:35 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 10 Apr 2013 15:17:35 -0500 Subject: [Freeswitch-users] Stale channel that cannot be killed, session created from a running Lua scripts. In-Reply-To: References: Message-ID: try new_session:setAutoHangup(0); new_session:destroy(); before you transfer it. and new_session = undefined; after you transfer it. The sessions are tied to the garbage collector so until the script frees the script version of the session the core can't free the real version. the :destroy() method detaches the 2 so the channel can hangup on its own and you only save the shell of the wrapper in your GC. On Wed, Apr 10, 2013 at 2:34 PM, Johny Kadarisman Kwan wrote: > I have lua 'background' script that wake up every few seconds, and check > to perform specific task, one of it to established call. > Calls are then initiated then transfer to some extension in dialplan. The > scripts has been running and works as expected. > But after session ended(hangup), channels is not released and can be seen > using cli "show channels". and trying to kill such channel result in "-ERR > No Such Channel!" > > This seems being reported before, but I can't seems to find any resolution. > > Following simple script could replicate the issues : > > freeswitch> luarun SimpleDialer.lua > > =================== SimpleDialer.lua > > local threadName = "SimpleDialer" > > while true do > > > freeswitch.consoleLog("info", threadName.." ticking\n") > > > -- task run here > local new_session = freeswitch.Session('sofia/gateway/vox/+150<%2B15085895115> > 88888888') > new_session:transfer("echo", "XML", "default") > > freeswitch.msleep(60000) > end > > ============= dialplan entry > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130410/9cfdd73f/attachment.html From stephen.thwaites at callstera.com Thu Apr 11 00:59:06 2013 From: stephen.thwaites at callstera.com (Stephen Thwaites) Date: Wed, 10 Apr 2013 22:59:06 +0200 Subject: [Freeswitch-users] Do I need a new dns resource records for each tenant? In-Reply-To: <1365243109613-7589470.post@n2.nabble.com> References: <1365243109613-7589470.post@n2.nabble.com> Message-ID: Hi, I have recently setup a multi-tenant FS following the two links on the wiki. This works well, I also currently add a subdomain per tenant pointing to the IP address of my FS server. However after reading this: http://lists.freeswitch.org/pipermail/freeswitch-users/2012-November/089467.html Realized it was probably not necessary. I was experimenting with a siemens gigaset and the voip account settings allow to fill in field for domain, the fields for registration server and sip proxy I filled in with the IP of my machine. FS with this setup found the right context etc. So it seems you don't need a dns record per tenant. However I don't fully understand the mechanism. Still learning!! I think the domain field defined in the voip account of the gigaset puts this in the 'from' bit of the SIP message and FS uses this to find the corresponding tenant. But the gigaset uses the IP address to contact the FS server for both registrations and invites. Can anybody clarify this? Regards, Steve. On Sat, Apr 6, 2013 at 12:11 PM, jow wrote: > The subject says it all. > do I need a new dns resource records for each new tenant? > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Do-I-need-a-new-dns-resource-records-for-each-tenant-tp7589470.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 jkr888 at gmail.com Thu Apr 11 01:13:47 2013 From: jkr888 at gmail.com (Johny Kadarisman Kwan) Date: Wed, 10 Apr 2013 17:13:47 -0400 Subject: [Freeswitch-users] Stale channel that cannot be killed, session created from a running Lua scripts. In-Reply-To: References: Message-ID: Cool, Thank you It works now! no more stale session session:destroy() and assigned nullable after transfer seems enough on this case =================== SimpleDialer.lua (after suggestion from Anthony) ============ local threadName = "SimpleDialer" while true do freeswitch.consoleLog("info", threadName.." ticking\n") -- task run here local new_session = freeswitch.Session('sofia/gateway/vox/+150<%2B15085895115> 88888888') new_session:transfer("echo", "XML", "default") new_session:destroy() new_session = nil freeswitch.msleep(60000) end On Wed, Apr 10, 2013 at 4:17 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > try > > new_session:setAutoHangup(0); > new_session:destroy(); > > before you transfer it. > and > > new_session = undefined; > > after you transfer it. > > The sessions are tied to the garbage collector so until the script frees > the script version of the session the core can't free the real version. > > the :destroy() method detaches the 2 so the channel can hangup on its own > and you only save the shell of the wrapper in your GC. > > > > > > > > > > On Wed, Apr 10, 2013 at 2:34 PM, Johny Kadarisman Kwan wrote: > >> I have lua 'background' script that wake up every few seconds, and check >> to perform specific task, one of it to established call. >> Calls are then initiated then transfer to some extension in dialplan. The >> scripts has been running and works as expected. >> But after session ended(hangup), channels is not released and can be seen >> using cli "show channels". and trying to kill such channel result in "-ERR >> No Such Channel!" >> >> This seems being reported before, but I can't seems to find any >> resolution. >> >> Following simple script could replicate the issues : >> >> freeswitch> luarun SimpleDialer.lua >> >> =================== SimpleDialer.lua >> >> local threadName = "SimpleDialer" >> >> while true do >> >> >> freeswitch.consoleLog("info", threadName.." ticking\n") >> >> >> -- task run here >> local new_session = freeswitch.Session('sofia/gateway/vox/+150<%2B15085895115> >> 88888888') >> new_session:transfer("echo", "XML", "default") >> >> freeswitch.msleep(60000) >> end >> >> ============= dialplan entry >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at 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/20130410/ac6314a3/attachment-0001.html From bdfoster at davri.com Thu Apr 11 01:32:01 2013 From: bdfoster at davri.com (Brian Foster) Date: Wed, 10 Apr 2013 17:32:01 -0400 Subject: [Freeswitch-users] Follow me implementation Message-ID: I've been given an assignment. It's a little rough, and honestly I've been working on other projects and at the same time loosing my freeswitch-fu. So, here it goes. Company owner wants to be able to implement a follow me function. He's asking for the deskphones to begin ringing, then have cell phones ring N seconds later WHILE the deskphones continue to ring. The function has to be able to work using a couple different ways of dialing (we've got call groups implemented, http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). When the mobile phone is answered, we need to be able to get some feedback from the callee to figure out if they're human. We'll use AVMD to kill the call if it detects a voicemail beep. I've looked at several different examples on the wiki and mailing list, and the only way I can figure out how to do it while keeping the requirements in mind is to at some point resort to using Loopback (something i didnt want to do). Requirements are: 1. Use a custom IVR/menu/something to get a confirmation from the callee that they are human (while also keeping it available for customization he's wanting a way to blacklist numbers on that same menu). So that rules out group_confirm_file, etc. 2. Use AVMD to kill the call if we detect the call was picked up by voicemail. 3. The custom IVR/menu/something isn't used on the deskphones 4. Deskphones need to continue to ring after the external number leg is started. I can't timeout the call on the deskphone then call the cell phone, or call the deskphone, time it out, then call the deskphone and cell phone. 5. It has to work on any type of calling method (so basically, if the deskphone rings then eventually the cell phone needs to ring to if it's assigned). Has anyone done something similar, and if so, how did you do it? Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/4e80853b/attachment.html From grcamauer at gmail.com Thu Apr 11 01:49:21 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Wed, 10 Apr 2013 18:49:21 -0300 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: I don't want to hijack your thread, but since the subject matter is FollowMe, I would like to add that I am looking for something along the lines of: A FollowMe that somehow knows what room you are in within a building and rings the nearest extension with a special ringtone which is assigned to each user. The "somehow" could be through a Bluetooth dongle attached to a PC in the room that detects the User's cell phone and updates a DB that FS has access to, or an Mobile Phone App that triangulates on WiFi Access Points and updates a DB, etc. Has anyone heard of such a system? Experiences? Thank you, Guillermo Ruiz Camauer On Wed, Apr 10, 2013 at 6:32 PM, Brian Foster wrote: > I've been given an assignment. It's a little rough, and honestly I've been > working on other projects and at the same time loosing my freeswitch-fu. > So, here it goes. > > Company owner wants to be able to implement a follow me function. He's > asking for the deskphones to begin ringing, then have cell phones ring N > seconds later WHILE the deskphones continue to ring. The function has to be > able to work using a couple different ways of dialing (we've got call > groups implemented, > http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). When > the mobile phone is answered, we need to be able to get some feedback from > the callee to figure out if they're human. We'll use AVMD to kill the call > if it detects a voicemail beep. > > I've looked at several different examples on the wiki and mailing list, > and the only way I can figure out how to do it while keeping the > requirements in mind is to at some point resort to using Loopback > (something i didnt want to do). > > Requirements are: > 1. Use a custom IVR/menu/something to get a confirmation from the callee > that they are human (while also keeping it available for customization > he's wanting a way to blacklist numbers on that same menu). So that rules > out group_confirm_file, etc. > 2. Use AVMD to kill the call if we detect the call was picked up by > voicemail. > 3. The custom IVR/menu/something isn't used on the deskphones > 4. Deskphones need to continue to ring after the external number leg is > started. I can't timeout the call on the deskphone then call the cell > phone, or call the deskphone, time it out, then call the deskphone and cell > phone. > 5. It has to work on any type of calling method (so basically, if the > deskphone rings then eventually the cell phone needs to ring to if it's > assigned). > > Has anyone done something similar, and if so, how did you do it? > > Thank you, > > Brian Foster > Project Manager/Owner's Representative > Davri Investments, Incorporated > P: +1-317-787-2686 > M: +1-317-600-9753 > Indianapolis, Indiana > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/e071d40a/attachment.html From msc at freeswitch.org Thu Apr 11 01:58:32 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 10 Apr 2013 14:58:32 -0700 Subject: [Freeswitch-users] [ANN] CID Lookup and Fat Free CRM In-Reply-To: References: Message-ID: Thanks for the heads up! I had never heard of FFCRM. Very cool. We appreciate hearing about things that people are doing with FreeSWITCH so please keep these reports coming! -MC On Wed, Apr 10, 2013 at 2:44 AM, Sergey Zhuravlov <4orbit at gmail.com> wrote: > Hey Guys, > > We are excited to announce that Fat Free CRM is now available to be > used with FreeSWITCH's mod_cidkookup. > > Solution based on standart mod_cidlookup and also may be used with > Asterisk. > FFCRM -- lightweight CRM used Ruby on Rails. http://www.fatfreecrm.com/ > > An open source CRM designed to be > highly customizable; elegant in simplicity. > > Yet doesn't have any integration with PBX. Perfect system, especially for > Rails' funs. > > Source code > https://github.com/4orbit/ffcrmcidlookup and > https://github.com/4orbit/fat_free_crm/tree/cidlookup > > See short article in Russian http://ru.it.ntcom.lv/blog/cid-lookup-ffcrm > The future plans of closer integration with the FreeSWITCH: > > - wake up the customer's card for an incoming call > - "smart" routing call to manager > > Supposed to use websocket! Supporters are welcomed > > > -- > WBR, Sergey > > GTALK/JABBER:4orbit at gmail.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/1ba06760/attachment.html From schoch+freeswitch.org at xwin32.com Thu Apr 11 02:16:48 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Wed, 10 Apr 2013 15:16:48 -0700 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: On Wed, Apr 10, 2013 at 2:32 PM, Brian Foster wrote: > 4. Deskphones need to continue to ring after the external number leg is > started. I can't timeout the call on the deskphone then call the cell > phone, or call the deskphone, time it out, then call the deskphone and cell > phone. > I just tried: The phone at 1001 rang briefly, but then it immediately was answered by the IVR. Can a LUA script handle things asynchronously? It would have to monitor if the local extension was answered while accepting DTMF from the cell phone call. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/48481292/attachment-0001.html From msc at freeswitch.org Thu Apr 11 02:18:06 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 10 Apr 2013 15:18:06 -0700 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: Hi Brian, The short answer to your question is: "enterprise originate". If you have the FS Cookbook then check out chapter 1, recipe #6 to get up and running quickly. (If you don't have the FS Cookbook then the next step is to tell your boss that a mere $25 is chump change compared to the buckets of cash you've saved by using a FreeSWITCH-based solution. Besides, it's a tax writeoff and it supports the project so you'll get karma++. :) Before I take any more time, please let us know if you're at all familiar with enterprise originate so that we can know where to take this conversation next. Thanks, MC On Wed, Apr 10, 2013 at 2:32 PM, Brian Foster wrote: > I've been given an assignment. It's a little rough, and honestly I've been > working on other projects and at the same time loosing my freeswitch-fu. > So, here it goes. > > Company owner wants to be able to implement a follow me function. He's > asking for the deskphones to begin ringing, then have cell phones ring N > seconds later WHILE the deskphones continue to ring. The function has to be > able to work using a couple different ways of dialing (we've got call > groups implemented, > http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). When > the mobile phone is answered, we need to be able to get some feedback from > the callee to figure out if they're human. We'll use AVMD to kill the call > if it detects a voicemail beep. > > I've looked at several different examples on the wiki and mailing list, > and the only way I can figure out how to do it while keeping the > requirements in mind is to at some point resort to using Loopback > (something i didnt want to do). > > Requirements are: > 1. Use a custom IVR/menu/something to get a confirmation from the callee > that they are human (while also keeping it available for customization > he's wanting a way to blacklist numbers on that same menu). So that rules > out group_confirm_file, etc. > 2. Use AVMD to kill the call if we detect the call was picked up by > voicemail. > 3. The custom IVR/menu/something isn't used on the deskphones > 4. Deskphones need to continue to ring after the external number leg is > started. I can't timeout the call on the deskphone then call the cell > phone, or call the deskphone, time it out, then call the deskphone and cell > phone. > 5. It has to work on any type of calling method (so basically, if the > deskphone rings then eventually the cell phone needs to ring to if it's > assigned). > > Has anyone done something similar, and if so, how did you do it? > > Thank you, > > Brian Foster > Project Manager/Owner's Representative > Davri Investments, Incorporated > P: +1-317-787-2686 > M: +1-317-600-9753 > Indianapolis, Indiana > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/66714ffc/attachment.html From grant at bmcrministries.org Thu Apr 11 02:16:26 2013 From: grant at bmcrministries.org (Grant Iler) Date: Wed, 10 Apr 2013 16:16:26 -0600 Subject: [Freeswitch-users] Outgoing fax issue. Message-ID: <004801ce3639$0bdef540$239cdfc0$@bmcrministries.org> Hi, We have a freeswitch setup with an Sangoma A200 plus 1 remora card setup. The cards have 6 FXO ports that go to 5 POTS lines and 2 FXS ports. Our fax machine is plugged into the first FXS port (freetdm 1:1). Incoming faxes work fine but outgoing faxes do not. The fax line is the 3rd Pots line (freetdm 2:3). The incoming faxes are bridged with this in the 00_inbound_did.xml: < !- Always directed to the fax machine -> < extension name=?Line 3(6982)? continue=?true?> < condition field=?source? expression=?mod_freetdm?/> < condition field=?${channel_name}? expression=?^(FreeTDM/2:3/)$?> < action application=?set? data=?domain_name=$${domain}?/> < action application=?set? data=?transfer_ringback=$${us-ring}?/> < action application=?bridge? data=?freetdm/2/3|freetdm/1/1?/> < /condition> < /extension> < !- LINE3 DONE-> Outgoing faxes are bridged with this in the dialplan/default.xml: < condition field=?destination_number? expression=?^7(.+)$?/> < condition field=?channel_name? expression=?FreeTDM/1:1/?> < action application=?bridge? data=? freetdm/1/1/${destination_number}|freetdm/2/3/${destination_number}?/> < /extension> What I figured out from the logs and research on the wikis is that when faxes are sent out, the freeswitch server does not get the entire number through DTMF and the call fails. Here is a portion of the logs from the last outgoing fax that worked: 2013-04-04 11:18:23.494287 [DEBUG] ftmod_analog.c:830 [s1c1][1:7] Changed state from DIALTONE to COLLECT 2013-04-04 11:18:23.494287 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:23.494287 [DEBUG] mod_freetdm.c:2488 got DTMF sig [7] 2013-04-04 11:18:23.514287 [DEBUG] ftmod_analog.c:640 [s1c1][1:7] Completed state change from DIALTONE to COLLECT in 20 ms 2013-04-04 11:18:23.514287 [DEBUG] ftmod_analog.c:646 [s1c1][1:7] Executing state handler on 1:1 for COLLECT 2013-04-04 11:18:24.074292 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 1 (debug = 0) 2013-04-04 11:18:24.074292 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:24.074292 [DEBUG] mod_freetdm.c:2488 got DTMF sig [71] 2013-04-04 11:18:25.113819 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 7 (debug = 0) 2013-04-04 11:18:25.113819 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:25.113819 [DEBUG] mod_freetdm.c:2488 got DTMF sig [717] 2013-04-04 11:18:25.554309 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 5 (debug = 0) 2013-04-04 11:18:25.554309 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:25.554309 [DEBUG] mod_freetdm.c:2488 got DTMF sig [7175] 2013-04-04 11:18:26.134309 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 7 (debug = 0) 2013-04-04 11:18:26.134309 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:26.134309 [DEBUG] mod_freetdm.c:2488 got DTMF sig [71757] 2013-04-04 11:18:26.634314 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 2 (debug = 0) 2013-04-04 11:18:26.634314 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:26.634314 [DEBUG] mod_freetdm.c:2488 got DTMF sig [717572] 2013-04-04 11:18:26.834315 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 2 (debug = 0) 2013-04-04 11:18:26.834315 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:26.834315 [DEBUG] mod_freetdm.c:2488 got DTMF sig [7175722] 2013-04-04 11:18:27.614322 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 9 (debug = 0) 2013-04-04 11:18:27.614322 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:27.614322 [DEBUG] mod_freetdm.c:2488 got DTMF sig [71757229] 2013-04-04 11:18:28.373722 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 0 (debug = 0) 2013-04-04 11:18:28.373722 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:28.373722 [DEBUG] mod_freetdm.c:2488 got DTMF sig [717572290] 2013-04-04 11:18:28.894333 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 6 (debug = 0) 2013-04-04 11:18:28.894333 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:28.894333 [DEBUG] mod_freetdm.c:2488 got DTMF sig [7175722906] 2013-04-04 11:18:29.714340 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 3 (debug = 0) 2013-04-04 11:18:29.714340 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:29.714340 [DEBUG] mod_freetdm.c:2488 got DTMF sig [71757229063] 2013-04-04 11:18:30.094343 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 4 (debug = 0) 2013-04-04 11:18:30.094343 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:30.094343 [DEBUG] mod_freetdm.c:2488 got DTMF sig [717572290634] 2013-04-04 11:18:32.114360 [DEBUG] ftmod_analog.c:851 [s1c1][1:7] Number obtained [717572290634] 2013-04-04 11:18:32.114360 [DEBUG] ftmod_analog.c:852 [s1c1][1:7] Changed state from COLLECT to RING 2013-04-04 11:18:32.134360 [DEBUG] ftmod_analog.c:640 [s1c1][1:7] Completed state change from COLLECT to RING in 19 ms 2013-04-04 11:18:32.134360 [DEBUG] ftmod_analog.c:646 [s1c1][1:7] Executing state handler on 1:1 for RING This is what happens now: 2013-04-09 14:54:20.616457 [DEBUG] ftmod_analog.c:830 [s1c1][1:7] Changed state from DIALTONE to COLLECT 2013-04-09 14:54:20.616457 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-09 14:54:20.616457 [DEBUG] mod_freetdm.c:2488 got DTMF sig [7] 2013-04-09 14:54:20.635887 [DEBUG] ftmod_analog.c:640 [s1c1][1:7] Completed state change from DIALTONE to COLLECT in 20 ms 2013-04-09 14:54:20.635887 [DEBUG] ftmod_analog.c:646 [s1c1][1:7] Executing state handler on 1:1 for COLLECT 2013-04-09 14:54:22.636477 [DEBUG] ftmod_analog.c:851 [s1c1][1:7] Number obtained [7] 2013-04-09 14:54:22.636477 [DEBUG] ftmod_analog.c:852 [s1c1][1:7] Changed state from COLLECT to RING 2013-04-09 14:54:22.656477 [DEBUG] ftmod_analog.c:640 [s1c1][1:7] Completed state change from COLLECT to RING in 20 ms Twice the Number obtained got more than one digit but it always dropped one at least one digit so that the fax number was not correct. I tried using start_dtmf and start_dtmf_generate from the Freeswitch wiki and also looked in the mailing list for issues with dtmf and faxing but was not able to find any solutions. I tried switching the FXS port to the 2nd one and swapped out the phone line but that did not have any other results. Let me know if you need to know more about the configuration or logs. Hopefully it is something easy someone with more knowledge than me can figure out. Thanks! Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/744b77c3/attachment-0001.html From msc at freeswitch.org Thu Apr 11 02:20:06 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 10 Apr 2013 15:20:06 -0700 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: Interesting proposition. Are there any devices out there for doing location within a building that have some sort of back-end interface to which you could communicate with your FS server? -MC On Wed, Apr 10, 2013 at 2:49 PM, Guillermo Ruiz Camauer wrote: > I don't want to hijack your thread, but since the subject matter is > FollowMe, I would like to add that I am looking for something along the > lines of: > > A FollowMe that somehow knows what room you are in within a building and > rings the nearest extension with a special ringtone which is assigned to > each user. > The "somehow" could be through a Bluetooth dongle attached to a PC in the > room that detects the User's cell phone and updates a DB that FS has access > to, or an Mobile Phone App that triangulates on WiFi Access Points and > updates a DB, etc. > > Has anyone heard of such a system? Experiences? > > Thank you, > > Guillermo Ruiz Camauer > > > On Wed, Apr 10, 2013 at 6:32 PM, Brian Foster wrote: > >> I've been given an assignment. It's a little rough, and honestly I've >> been working on other projects and at the same time loosing my >> freeswitch-fu. So, here it goes. >> >> Company owner wants to be able to implement a follow me function. He's >> asking for the deskphones to begin ringing, then have cell phones ring N >> seconds later WHILE the deskphones continue to ring. The function has to be >> able to work using a couple different ways of dialing (we've got call >> groups implemented, >> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). When >> the mobile phone is answered, we need to be able to get some feedback from >> the callee to figure out if they're human. We'll use AVMD to kill the call >> if it detects a voicemail beep. >> >> I've looked at several different examples on the wiki and mailing list, >> and the only way I can figure out how to do it while keeping the >> requirements in mind is to at some point resort to using Loopback >> (something i didnt want to do). >> >> Requirements are: >> 1. Use a custom IVR/menu/something to get a confirmation from the callee >> that they are human (while also keeping it available for customization >> he's wanting a way to blacklist numbers on that same menu). So that rules >> out group_confirm_file, etc. >> 2. Use AVMD to kill the call if we detect the call was picked up by >> voicemail. >> 3. The custom IVR/menu/something isn't used on the deskphones >> 4. Deskphones need to continue to ring after the external number leg is >> started. I can't timeout the call on the deskphone then call the cell >> phone, or call the deskphone, time it out, then call the deskphone and cell >> phone. >> 5. It has to work on any type of calling method (so basically, if the >> deskphone rings then eventually the cell phone needs to ring to if it's >> assigned). >> >> Has anyone done something similar, and if so, how did you do it? >> >> Thank you, >> >> Brian Foster >> Project Manager/Owner's Representative >> Davri Investments, Incorporated >> P: +1-317-787-2686 >> M: +1-317-600-9753 >> Indianapolis, Indiana >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/a70fde48/attachment.html From drk at drkngs.net Thu Apr 11 02:27:08 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Wed, 10 Apr 2013 15:27:08 -0700 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: Message-ID: <20130410222708.f5020054@mail.tritonwest.net> I don't know if this is exactly what you are looking for, but I can tell you how, in general I have implment such features. The if you are doing machine generated config (xml_curl, scripting lang that supports config scripts, etc..) I do the first bridge normally, followed by a transfer back to the number that was called, using the context as a flag, like _route_list_1_step_2, or something like that. When you get the config call back you then have all the context you need to figure out your next step of the follow me, and try the next thing on the list, w/o having to do it all durring the initial dial plan, and you can be more dymanic by looking at the origination_disposition and maybe taking different actions depending on it. --Dave _____ From: Guillermo Ruiz Camauer [mailto:grcamauer at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Wed, 10 Apr 2013 14:49:21 -0700 Subject: Re: [Freeswitch-users] Follow me implementation I don't want to hijack your thread, but since the subject matter is FollowMe, I would like to add that I am looking for something along the lines of: A FollowMe that somehow knows what room you are in within a building and rings the nearest extension with a special ringtone which is assigned to each user. The "somehow" could be through a Bluetooth dongle attached to a PC in the room that detects the User's cell phone and updates a DB that FS has access to, or an Mobile Phone App that triangulates on WiFi Access Points and updates a DB, etc. Has anyone heard of such a system? Experiences? Thank you, Guillermo Ruiz Camauer On Wed, Apr 10, 2013 at 6:32 PM, Brian Foster wrote: I've been given an assignment. It's a little rough, and honestly I've been working on other projects and at the same time loosing my freeswitch-fu. So, here it goes. Company owner wants to be able to implement a follow me function. He's asking for the deskphones to begin ringing, then have cell phones ring N seconds later WHILE the deskphones continue to ring. The function has to be able to work using a couple different ways of dialing (we've got call groups implemented, http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). When the mobile phone is answered, we need to be able to get some feedback from the callee to figure out if they're human. We'll use AVMD to kill the call if it detects a voicemail beep. I've looked at several different examples on the wiki and mailing list, and the only way I can figure out how to do it while keeping the requirements in mind is to at some point resort to using Loopback (something i didnt want to do). Requirements are: 1. Use a custom IVR/menu/something to get a confirmation from the callee that they are human (while also keeping it available for customization he's wanting a way to blacklist numbers on that same menu). So that rules out group_confirm_file, etc. 2. Use AVMD to kill the call if we detect the call was picked up by voicemail. 3. The custom IVR/menu/something isn't used on the deskphones 4. Deskphones need to continue to ring after the external number leg is started. I can't timeout the call on the deskphone then call the cell phone, or call the deskphone, time it out, then call the deskphone and cell phone. 5. It has to work on any type of calling method (so basically, if the deskphone rings then eventually the cell phone needs to ring to if it's assigned). Has anyone done something similar, and if so, how did you do it? Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/436bcbcd/attachment.html From msc at freeswitch.org Thu Apr 11 02:31:17 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 10 Apr 2013 15:31:17 -0700 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: On Wed, Apr 10, 2013 at 3:16 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Wed, Apr 10, 2013 at 2:32 PM, Brian Foster wrote: > >> 4. Deskphones need to continue to ring after the external number leg is >> started. I can't timeout the call on the deskphone then call the cell >> phone, or call the deskphone, time it out, then call the deskphone and cell >> phone. >> > > I just tried: > > > The phone at 1001 rang briefly, but then it immediately was answered by > the IVR. > That is the expected behavior of the app/arg combo you have here. > > Can a LUA script handle things asynchronously? It would have to monitor > if the local extension was answered while accepting DTMF from the cell > phone call. > No - a Lua script called from the dialplan cannot "see" what's happening in the way you want. If you want this kind of parallel calling that is controlled by some sort of logic you'll need a 3rd party app/script/program to watch everything via the event socket and make changes accordingly. Just remember that enterprise originate does A TON of work in this whole multiple leg thing. Anthony put some serious time and energy into making that work - and it works very well. However, when one of the called parties answers then the whole "enterprise" has completed its mission (sorry). If you need the called party to do some sort of acknowledgement (i.e. press a key) then you'll need to employ group_confirm_key/group_confirm_file. -MC > -- > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/5ae40044/attachment-0001.html From avi at avimarcus.net Thu Apr 11 02:40:12 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 11 Apr 2013 01:40:12 +0300 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: However, when one of the called parties answers then the whole "enterprise" has completed its mission (sorry). If you need the called party to do some sort of acknowledgement (i.e. press a key) then you'll need to employ group_confirm_key/group_confirm_file. E.g. if you are calling a mobile, you should always do group confirm: otherwise if the phone is off / no reception, voicemail on the mobile picks up - doh! An enterprise that bridges to a cellphone and to a SIP phone could look something like this: {ignore_early_media=true,group_confirm_file=ivr/ivr-accept_reject_voicemail.wav,group_confirm_key=1}sofia/gateway/flowroute/12223334444:_:user/1000 The < is for everything, the { is only for that "thread" of the enterprise originate - the group confirm isn't used for the user/ bridge. You can also do a leg_delay_start=5 on the mobile section so that only if the sip phone isn't picked up within the first e.g. 5 seconds then it rings the mobile (beware of PDD though). If the other legs error out, then the leg_delay_start will be skipped. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/97d4c2c3/attachment.html From nasida at live.ru Thu Apr 11 02:44:17 2013 From: nasida at live.ru (Yuriy Nasida) Date: Thu, 11 Apr 2013 02:44:17 +0400 Subject: [Freeswitch-users] native support of postgres enable-core-pgsql-support and without-pgsql Message-ID: Hi guys, I had idea to try the new feature of native support of postgres today but it was not so easy as I though at the beginning. After installing of postgresql on my CentSO 6.2 (simple via yum) and trying to recompile FS with --enable-core-pgsql-support. I have got many different issues in the process of compiling. All them were well-known as far as I understand but it was still not easy to understand how to fix it and to get native postgres support. FS-3384, FS-3630, FS-3384 and FS-3393. I had all this problems.http://wiki.freeswitch.org/wiki/Installation_Guide#Release.28es.29_6_and_Laterfor more full info. For solving was suggested ./configure --without-pgsql . But.... I would like to have pgsql in core... In the end I was tried this combination :)./configure --prefix=/usr/local/freeswitch_1_2_pg --enable-core-pgsql-support --without-pgsql So with core-pgsql and without pgsql simultaneously. It looks strange but it works. Hope it will help somebody else. I have one more question. Can I add postgres native support for registrations only ? Like I can have this by means of adding of dbc-dsn parameter to internal.xml only ? P.S. Very nice increase speed with native postgree . -- YN -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/733fcabd/attachment.html From nneul at mst.edu Thu Apr 11 02:45:17 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 10 Apr 2013 17:45:17 -0500 Subject: [Freeswitch-users] Outgoing fax issue. In-Reply-To: <004801ce3639$0bdef540$239cdfc0$@bmcrministries.org> References: <004801ce3639$0bdef540$239cdfc0$@bmcrministries.org> Message-ID: <5165EB7D.70303@mst.edu> You've told it to match "7" followed by any number of digits... I'm surprised your actually getting as far into the pattern match as you are. Can you match on specific number plan lengths? i.e. ^7(\d{7}|1\d{10})$ instead? -- Nathan > > < condition field=?destination_number? expression=?^7(.+)$?/> > < condition field=?channel_name? expression=?FreeTDM/1:1/?> > < action application=?bridge? data=?freetdm/1/1/${destination_number}|freetdm/2/3/${destination_number}?/> > < /extension> -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From grcamauer at gmail.com Thu Apr 11 02:48:18 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Wed, 10 Apr 2013 19:48:18 -0300 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: I found this: http://redpin.org/index.html. >From the site: Redpin is an open source indoor positioning system that was developed with the goal of providing at least room-level accuracy. Moreover, it avoids the time-consuming training and setup phase known from other systems and instead relies on the user community. But it seems to require a Apple Developer account to compile the iPhone App. Guillermo On Wed, Apr 10, 2013 at 7:20 PM, Michael Collins wrote: > Interesting proposition. Are there any devices out there for doing > location within a building that have some sort of back-end interface to > which you could communicate with your FS server? > -MC > > > On Wed, Apr 10, 2013 at 2:49 PM, Guillermo Ruiz Camauer < > grcamauer at gmail.com> wrote: > >> I don't want to hijack your thread, but since the subject matter is >> FollowMe, I would like to add that I am looking for something along the >> lines of: >> >> A FollowMe that somehow knows what room you are in within a building and >> rings the nearest extension with a special ringtone which is assigned to >> each user. >> The "somehow" could be through a Bluetooth dongle attached to a PC in the >> room that detects the User's cell phone and updates a DB that FS has access >> to, or an Mobile Phone App that triangulates on WiFi Access Points and >> updates a DB, etc. >> >> Has anyone heard of such a system? Experiences? >> >> Thank you, >> >> Guillermo Ruiz Camauer >> >> >> On Wed, Apr 10, 2013 at 6:32 PM, Brian Foster wrote: >> >>> I've been given an assignment. It's a little rough, and honestly I've >>> been working on other projects and at the same time loosing my >>> freeswitch-fu. So, here it goes. >>> >>> Company owner wants to be able to implement a follow me function. He's >>> asking for the deskphones to begin ringing, then have cell phones ring N >>> seconds later WHILE the deskphones continue to ring. The function has to be >>> able to work using a couple different ways of dialing (we've got call >>> groups implemented, >>> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). When >>> the mobile phone is answered, we need to be able to get some feedback from >>> the callee to figure out if they're human. We'll use AVMD to kill the call >>> if it detects a voicemail beep. >>> >>> I've looked at several different examples on the wiki and mailing list, >>> and the only way I can figure out how to do it while keeping the >>> requirements in mind is to at some point resort to using Loopback >>> (something i didnt want to do). >>> >>> Requirements are: >>> 1. Use a custom IVR/menu/something to get a confirmation from the callee >>> that they are human (while also keeping it available for customization >>> he's wanting a way to blacklist numbers on that same menu). So that rules >>> out group_confirm_file, etc. >>> 2. Use AVMD to kill the call if we detect the call was picked up by >>> voicemail. >>> 3. The custom IVR/menu/something isn't used on the deskphones >>> 4. Deskphones need to continue to ring after the external number leg is >>> started. I can't timeout the call on the deskphone then call the cell >>> phone, or call the deskphone, time it out, then call the deskphone and cell >>> phone. >>> 5. It has to work on any type of calling method (so basically, if the >>> deskphone rings then eventually the cell phone needs to ring to if it's >>> assigned). >>> >>> Has anyone done something similar, and if so, how did you do it? >>> >>> Thank you, >>> >>> Brian Foster >>> Project Manager/Owner's Representative >>> Davri Investments, Incorporated >>> P: +1-317-787-2686 >>> M: +1-317-600-9753 >>> Indianapolis, Indiana >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.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 > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/e1b4c5f8/attachment-0001.html From grcamauer at gmail.com Thu Apr 11 03:25:22 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Wed, 10 Apr 2013 20:25:22 -0300 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: I also found this, which looks even more promising, but is just starting up: http://www.indiegogo.com/projects/sticknfind-bluetooth-powered-ultra-small-location-stickers They still are working on a Windows API. Guillermo On Wed, Apr 10, 2013 at 7:20 PM, Michael Collins wrote: > Interesting proposition. Are there any devices out there for doing > location within a building that have some sort of back-end interface to > which you could communicate with your FS server? > -MC > > > On Wed, Apr 10, 2013 at 2:49 PM, Guillermo Ruiz Camauer < > grcamauer at gmail.com> wrote: > >> I don't want to hijack your thread, but since the subject matter is >> FollowMe, I would like to add that I am looking for something along the >> lines of: >> >> A FollowMe that somehow knows what room you are in within a building and >> rings the nearest extension with a special ringtone which is assigned to >> each user. >> The "somehow" could be through a Bluetooth dongle attached to a PC in the >> room that detects the User's cell phone and updates a DB that FS has access >> to, or an Mobile Phone App that triangulates on WiFi Access Points and >> updates a DB, etc. >> >> Has anyone heard of such a system? Experiences? >> >> Thank you, >> >> Guillermo Ruiz Camauer >> >> >> On Wed, Apr 10, 2013 at 6:32 PM, Brian Foster wrote: >> >>> I've been given an assignment. It's a little rough, and honestly I've >>> been working on other projects and at the same time loosing my >>> freeswitch-fu. So, here it goes. >>> >>> Company owner wants to be able to implement a follow me function. He's >>> asking for the deskphones to begin ringing, then have cell phones ring N >>> seconds later WHILE the deskphones continue to ring. The function has to be >>> able to work using a couple different ways of dialing (we've got call >>> groups implemented, >>> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). When >>> the mobile phone is answered, we need to be able to get some feedback from >>> the callee to figure out if they're human. We'll use AVMD to kill the call >>> if it detects a voicemail beep. >>> >>> I've looked at several different examples on the wiki and mailing list, >>> and the only way I can figure out how to do it while keeping the >>> requirements in mind is to at some point resort to using Loopback >>> (something i didnt want to do). >>> >>> Requirements are: >>> 1. Use a custom IVR/menu/something to get a confirmation from the callee >>> that they are human (while also keeping it available for customization >>> he's wanting a way to blacklist numbers on that same menu). So that rules >>> out group_confirm_file, etc. >>> 2. Use AVMD to kill the call if we detect the call was picked up by >>> voicemail. >>> 3. The custom IVR/menu/something isn't used on the deskphones >>> 4. Deskphones need to continue to ring after the external number leg is >>> started. I can't timeout the call on the deskphone then call the cell >>> phone, or call the deskphone, time it out, then call the deskphone and cell >>> phone. >>> 5. It has to work on any type of calling method (so basically, if the >>> deskphone rings then eventually the cell phone needs to ring to if it's >>> assigned). >>> >>> Has anyone done something similar, and if so, how did you do it? >>> >>> Thank you, >>> >>> Brian Foster >>> Project Manager/Owner's Representative >>> Davri Investments, Incorporated >>> P: +1-317-787-2686 >>> M: +1-317-600-9753 >>> Indianapolis, Indiana >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.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 > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/ca29d428/attachment.html From bdfoster at davri.com Thu Apr 11 03:36:34 2013 From: bdfoster at davri.com (Brian Foster) Date: Wed, 10 Apr 2013 19:36:34 -0400 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: LinuxMCE has done the Bluetooth tracking bit, albeit with extra code and asterisk. On Wednesday, April 10, 2013, Guillermo Ruiz Camauer wrote: > I don't want to hijack your thread, but since the subject matter is > FollowMe, I would like to add that I am looking for something along the > lines of: > > A FollowMe that somehow knows what room you are in within a building and > rings the nearest extension with a special ringtone which is assigned to > each user. > The "somehow" could be through a Bluetooth dongle attached to a PC in the > room that detects the User's cell phone and updates a DB that FS has access > to, or an Mobile Phone App that triangulates on WiFi Access Points and > updates a DB, etc. > > Has anyone heard of such a system? Experiences? > > Thank you, > > Guillermo Ruiz Camauer > > > On Wed, Apr 10, 2013 at 6:32 PM, Brian Foster > > wrote: > >> I've been given an assignment. It's a little rough, and honestly I've >> been working on other projects and at the same time loosing my >> freeswitch-fu. So, here it goes. >> >> Company owner wants to be able to implement a follow me function. He's >> asking for the deskphones to begin ringing, then have cell phones ring N >> seconds later WHILE the deskphones continue to ring. The function has to be >> able to work using a couple different ways of dialing (we've got call >> groups implemented, >> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). When >> the mobile phone is answered, we need to be able to get some feedback from >> the callee to figure out if they're human. We'll use AVMD to kill the call >> if it detects a voicemail beep. >> >> I've looked at several different examples on the wiki and mailing list, >> and the only way I can figure out how to do it while keeping the >> requirements in mind is to at some point resort to using Loopback >> (something i didnt want to do). >> >> Requirements are: >> 1. Use a custom IVR/menu/something to get a confirmation from the callee >> that they are human (while also keeping it available for customization >> he's wanting a way to blacklist numbers on that same menu). So that rules >> out group_confirm_file, etc. >> 2. Use AVMD to kill the call if we detect the call was picked up by >> voicemail. >> 3. The custom IVR/menu/something isn't used on the deskphones >> 4. Deskphones need to continue to ring after the external number leg is >> started. I can't timeout the call on the deskphone then call the cell >> phone, or call the deskphone, time it out, then call the deskphone and cell >> phone. >> 5. It has to work on any type of calling method (so basically, if the >> deskphone rings then eventually the cell phone needs to ring to if it's >> assigned). >> >> Has anyone done something similar, and if so, how did you do it? >> >> Thank you, >> >> Brian Foster >> Project Manager/Owner's Representative >> Davri Investments, Incorporated >> P: +1-317-787-2686 >> M: +1-317-600-9753 >> Indianapolis, Indiana >> >> _________________________________________________________________________ >> 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 >> >> > > > -- > Guillermo Ruiz Camauer > -- Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/68404183/attachment-0001.html From bdfoster at davri.com Thu Apr 11 04:13:17 2013 From: bdfoster at davri.com (Brian Foster) Date: Wed, 10 Apr 2013 20:13:17 -0400 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: MSC: Yea, I've heard of it but never really used it. I've taken a look at some documentation here: http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Enterprise_originate Avi's Suggestion: {ignore_early_media=true,group_ confirm_file=ivr/ivr-accept_reject_voicemail.wav,group_ confirm_key=1}sofia/gateway/flowroute/12223334444:_:user/1000 Avi: We're close :) Now, I'm not sure if that will actually work (might not take much to get it to work) but theoretically it should. To get it clean, I'll probably have to do a script that takes a look at the user's variables set in the directory and figure out if the user even has a follow_me_number set. Notice "{group_confirm_key=exec,group_confirm_file=lua menu.lua}". That was taken from http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm. Anyone know if that works? That would sure as hell make a dent in this. Another problem is that this dial-string will likely have to be set in the directory in order to keep things consistent. What's the best route to take? I'd have to use a script to set the dial string, not sure how I could call the script from there. Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana On Wed, Apr 10, 2013 at 7:36 PM, Brian Foster wrote: > LinuxMCE has done the Bluetooth tracking bit, albeit with extra code and > asterisk. > > > On Wednesday, April 10, 2013, Guillermo Ruiz Camauer wrote: > >> I don't want to hijack your thread, but since the subject matter is >> FollowMe, I would like to add that I am looking for something along the >> lines of: >> >> A FollowMe that somehow knows what room you are in within a building and >> rings the nearest extension with a special ringtone which is assigned to >> each user. >> The "somehow" could be through a Bluetooth dongle attached to a PC in the >> room that detects the User's cell phone and updates a DB that FS has access >> to, or an Mobile Phone App that triangulates on WiFi Access Points and >> updates a DB, etc. >> >> Has anyone heard of such a system? Experiences? >> >> Thank you, >> >> Guillermo Ruiz Camauer >> >> >> On Wed, Apr 10, 2013 at 6:32 PM, Brian Foster wrote: >> >>> I've been given an assignment. It's a little rough, and honestly I've >>> been working on other projects and at the same time loosing my >>> freeswitch-fu. So, here it goes. >>> >>> Company owner wants to be able to implement a follow me function. He's >>> asking for the deskphones to begin ringing, then have cell phones ring N >>> seconds later WHILE the deskphones continue to ring. The function has to be >>> able to work using a couple different ways of dialing (we've got call >>> groups implemented, >>> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). When >>> the mobile phone is answered, we need to be able to get some feedback from >>> the callee to figure out if they're human. We'll use AVMD to kill the call >>> if it detects a voicemail beep. >>> >>> I've looked at several different examples on the wiki and mailing list, >>> and the only way I can figure out how to do it while keeping the >>> requirements in mind is to at some point resort to using Loopback >>> (something i didnt want to do). >>> >>> Requirements are: >>> 1. Use a custom IVR/menu/something to get a confirmation from the callee >>> that they are human (while also keeping it available for customization >>> he's wanting a way to blacklist numbers on that same menu). So that rules >>> out group_confirm_file, etc. >>> 2. Use AVMD to kill the call if we detect the call was picked up by >>> voicemail. >>> 3. The custom IVR/menu/something isn't used on the deskphones >>> 4. Deskphones need to continue to ring after the external number leg is >>> started. I can't timeout the call on the deskphone then call the cell >>> phone, or call the deskphone, time it out, then call the deskphone and cell >>> phone. >>> 5. It has to work on any type of calling method (so basically, if the >>> deskphone rings then eventually the cell phone needs to ring to if it's >>> assigned). >>> >>> Has anyone done something similar, and if so, how did you do it? >>> >>> Thank you, >>> >>> Brian Foster >>> Project Manager/Owner's Representative >>> Davri Investments, Incorporated >>> P: +1-317-787-2686 >>> M: +1-317-600-9753 >>> Indianapolis, Indiana >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Guillermo Ruiz Camauer >> > > > -- > Thank you, > > Brian Foster > Project Manager/Owner's Representative > Davri Investments, Incorporated > P: +1-317-787-2686 > M: +1-317-600-9753 > Indianapolis, Indiana > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/346cabc0/attachment.html From bdfoster at davri.com Thu Apr 11 04:16:46 2013 From: bdfoster at davri.com (Brian Foster) Date: Wed, 10 Apr 2013 20:16:46 -0400 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: Also forgot to add, I'd have to use Loopback because I would have to send the follow me number back through the dialplan in order to figure out which gateway it should go out (we select a gateway based on a few different variables, but there's no way to tell until you send it through the dialplan). I know Loopback is kinda evil but I don't see another way of doing it. But theoretically it makes things a little more modular and keeps maintenance of the dialplan to a minimum. Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana On Wed, Apr 10, 2013 at 8:13 PM, Brian Foster wrote: > MSC: Yea, I've heard of it but never really used it. I've taken a look at > some documentation here: > http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Enterprise_originate > > Avi's Suggestion: > > {ignore_early_media=true,group_ > confirm_file=ivr/ivr-accept_reject_voicemail.wav,group_ > confirm_key=1}sofia/gateway/flowroute/12223334444:_:user/1000 > > Avi: We're close :) > > > > Now, I'm not sure if that will actually work (might not take much to get > it to work) but theoretically it should. To get it clean, I'll probably > have to do a script that takes a look at the user's variables set in the > directory and figure out if the user even has a follow_me_number set. > > Notice "{group_confirm_key=exec,group_confirm_file=lua menu.lua}". That > was taken from > http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm. > Anyone know if that works? That would sure as hell make a dent in this. > > Another problem is that this dial-string will likely have to be set in the > directory in order to keep things consistent. What's the best route to > take? I'd have to use a script to set the dial string, not sure how I could > call the script from there. > > > > Thank you, > > Brian Foster > Project Manager/Owner's Representative > Davri Investments, Incorporated > P: +1-317-787-2686 > M: +1-317-600-9753 > Indianapolis, Indiana > > > On Wed, Apr 10, 2013 at 7:36 PM, Brian Foster wrote: > >> LinuxMCE has done the Bluetooth tracking bit, albeit with extra code and >> asterisk. >> >> >> On Wednesday, April 10, 2013, Guillermo Ruiz Camauer wrote: >> >>> I don't want to hijack your thread, but since the subject matter is >>> FollowMe, I would like to add that I am looking for something along the >>> lines of: >>> >>> A FollowMe that somehow knows what room you are in within a building and >>> rings the nearest extension with a special ringtone which is assigned to >>> each user. >>> The "somehow" could be through a Bluetooth dongle attached to a PC in >>> the room that detects the User's cell phone and updates a DB that FS has >>> access to, or an Mobile Phone App that triangulates on WiFi Access Points >>> and updates a DB, etc. >>> >>> Has anyone heard of such a system? Experiences? >>> >>> Thank you, >>> >>> Guillermo Ruiz Camauer >>> >>> >>> On Wed, Apr 10, 2013 at 6:32 PM, Brian Foster wrote: >>> >>>> I've been given an assignment. It's a little rough, and honestly I've >>>> been working on other projects and at the same time loosing my >>>> freeswitch-fu. So, here it goes. >>>> >>>> Company owner wants to be able to implement a follow me function. He's >>>> asking for the deskphones to begin ringing, then have cell phones ring N >>>> seconds later WHILE the deskphones continue to ring. The function has to be >>>> able to work using a couple different ways of dialing (we've got call >>>> groups implemented, >>>> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). When >>>> the mobile phone is answered, we need to be able to get some feedback from >>>> the callee to figure out if they're human. We'll use AVMD to kill the call >>>> if it detects a voicemail beep. >>>> >>>> I've looked at several different examples on the wiki and mailing list, >>>> and the only way I can figure out how to do it while keeping the >>>> requirements in mind is to at some point resort to using Loopback >>>> (something i didnt want to do). >>>> >>>> Requirements are: >>>> 1. Use a custom IVR/menu/something to get a confirmation from the >>>> callee that they are human (while also keeping it available >>>> for customization he's wanting a way to blacklist numbers on that same >>>> menu). So that rules out group_confirm_file, etc. >>>> 2. Use AVMD to kill the call if we detect the call was picked up by >>>> voicemail. >>>> 3. The custom IVR/menu/something isn't used on the deskphones >>>> 4. Deskphones need to continue to ring after the external number leg is >>>> started. I can't timeout the call on the deskphone then call the cell >>>> phone, or call the deskphone, time it out, then call the deskphone and cell >>>> phone. >>>> 5. It has to work on any type of calling method (so basically, if the >>>> deskphone rings then eventually the cell phone needs to ring to if it's >>>> assigned). >>>> >>>> Has anyone done something similar, and if so, how did you do it? >>>> >>>> Thank you, >>>> >>>> Brian Foster >>>> Project Manager/Owner's Representative >>>> Davri Investments, Incorporated >>>> P: +1-317-787-2686 >>>> M: +1-317-600-9753 >>>> Indianapolis, Indiana >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Guillermo Ruiz Camauer >>> >> >> >> -- >> Thank you, >> >> Brian Foster >> Project Manager/Owner's Representative >> Davri Investments, Incorporated >> P: +1-317-787-2686 >> M: +1-317-600-9753 >> Indianapolis, Indiana >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/52a5ca19/attachment-0001.html From victor.chukalovskiy at gmail.com Thu Apr 11 06:07:30 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Wed, 10 Apr 2013 22:07:30 -0400 Subject: [Freeswitch-users] Unloop extension In-Reply-To: References: <20130115113340.3ccb2287@mail.btcom.kz> Message-ID: <51661AE2.5060508@gmail.com> Wanted to ask this some time ago... all this extension does is check whether call is "from" and "to" the same SIP profile? Like if someone was to create registration, register to themselves, and send calls to that registration? On 01/15/2013 07:54 PM, Michael Collins wrote: > Yes, it's okay. The pre-process does a set and ${unroll_loops} gets > set as needed. Then this dp entry does the work: > > > > > > > > More info on sip_looped_call var: > http://wiki.freeswitch.org/wiki/Variable_sip_looped_call > > -MC > > On Mon, Jan 14, 2013 at 9:33 PM, ????? ??????? > wrote: > > What unloop extension is used for? BTW: vars.xml > pre-(process)-sets unroll_loops, whereas DPs (public and default) > are using channel variable syntax? Is this ok? > > Y. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.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/20130410/b98fb974/attachment.html From anthony.minessale at gmail.com Thu Apr 11 06:18:38 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 10 Apr 2013 19:18:38 -0700 Subject: [Freeswitch-users] Unloop extension In-Reply-To: <51661AE2.5060508@gmail.com> References: <20130115113340.3ccb2287@mail.btcom.kz> <51661AE2.5060508@gmail.com> Message-ID: Yes, then it transfers it to remove the bridge. On Apr 10, 2013 9:11 PM, "Victor Chukalovskiy" < victor.chukalovskiy at gmail.com> wrote: > Wanted to ask this some time ago... all this extension does is check > whether call is "from" and "to" the same SIP profile? Like if someone was > to create registration, register to themselves, and send calls to that > registration? > > On 01/15/2013 07:54 PM, Michael Collins wrote: > > Yes, it's okay. The pre-process does a set and ${unroll_loops} gets set as > needed. Then this dp entry does the work: > > > > > > > > More info on sip_looped_call var: > http://wiki.freeswitch.org/wiki/Variable_sip_looped_call > > -MC > > On Mon, Jan 14, 2013 at 9:33 PM, ????? ??????? wrote: > >> What unloop extension is used for? BTW: vars.xml pre-(process)-sets >> unroll_loops, whereas DPs (public and default) are using channel variable >> syntax? Is this ok? >> >> Y. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/41de88a6/attachment.html From victor.chukalovskiy at gmail.com Thu Apr 11 06:23:33 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Wed, 10 Apr 2013 22:23:33 -0400 Subject: [Freeswitch-users] Detecting call loops while not relying on the hop counter Message-ID: <51661EA5.40204@gmail.com> Hello, I have a scenario where FS routes hundreds of calls and loop detection is needed. Adding to that, I can't rely on hop counter being decremented by other network elements. I can use limit in the following fashion: So, whenever I have 10 calls with the same calling and called party number combination, limit will kick-in and break the loop. *Question*: Is this efficient from the perspective of how "limit" and hash back-end work? Once I have a few thousand calls, all having unique "resource" will this become resource hungry? My main concern is that it will create hundreds of hash entries all having the same realm but unique resource. Anyone knows a better solution? Thank you, Victor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/614bbd9f/attachment.html From krice at freeswitch.org Thu Apr 11 06:28:42 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 10 Apr 2013 21:28:42 -0500 Subject: [Freeswitch-users] Detecting call loops while not relying on the hop counter In-Reply-To: <51661EA5.40204@gmail.com> Message-ID: A better way to do this is limit this during a period of time, say 5 per second, when calls loop they tend to loop up very quickly On 4/10/13 9:23 PM, "Victor Chukalovskiy" wrote: > Hello, > > I have a scenario where FS routes hundreds of calls and loop detection is > needed. Adding to that, I can't rely on hop counter being decremented by other > network elements. I can use limit in the following fashion: > > > > So, whenever I have 10 calls with the same calling and called party number > combination, limit will kick-in and break the loop. > > Question: Is this efficient from the perspective of how "limit" and hash > back-end work? Once I have a few thousand calls, all having unique "resource" > will this become resource hungry? > My main concern is that it will create hundreds of hash entries all having > the same realm but unique resource. > > Anyone knows a better solution? > > Thank you, > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130410/d4faf1c8/attachment-0001.html From victor.chukalovskiy at gmail.com Thu Apr 11 07:03:33 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Wed, 10 Apr 2013 23:03:33 -0400 Subject: [Freeswitch-users] Detecting call loops while not relying on the hop counter In-Reply-To: References: Message-ID: <51662805.9070504@gmail.com> Yep, makes sense for sure....and how to accomplish that? On 04/10/2013 10:28 PM, Ken Rice wrote: > Re: [Freeswitch-users] Detecting call loops while not relying on the > hop counter A better way to do this is limit this during a period of > time, say 5 per second, when calls loop they tend to loop up very quickly > > > On 4/10/13 9:23 PM, "Victor Chukalovskiy" > wrote: > > Hello, > > I have a scenario where FS routes hundreds of calls and loop > detection is needed. Adding to that, I can't rely on hop counter > being decremented by other network elements. I can use limit in > the following fashion: > > > > So, whenever I have 10 calls with the same calling and called > party number combination, limit will kick-in and break the loop. > > *Question*: Is this efficient from the perspective of how "limit" > and hash back-end work? Once I have a few thousand calls, all > having unique "resource" will this become resource hungry? > My main concern is that it will create hundreds of hash entries > all having the same realm but unique resource. > > Anyone knows a better solution? > > Thank you, > 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 > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #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/20130410/1fb951cf/attachment.html From ml88888 at hotmail.com Thu Apr 11 09:41:12 2013 From: ml88888 at hotmail.com (FSX) Date: Wed, 10 Apr 2013 22:41:12 -0700 (PDT) Subject: [Freeswitch-users] Build errors in 2010.express solution In-Reply-To: <1365599711613-7589611.post@n2.nabble.com> References: <1365285612276-7589476.post@n2.nabble.com> <1365424893457-7589513.post@n2.nabble.com> <1365576667168-7589585.post@n2.nabble.com> <1365599711613-7589611.post@n2.nabble.com> Message-ID: <1365658872072-7589655.post@n2.nabble.com> Thank you, Jeff! "git clean -fdx" helped :) BTW, why I always have to turn off building "mod_managed" project? I'm getting errors with it and in order to build the whole solution I have to turn off this project. As a result, the file "FreeSWITCH.2010.express.sln" becomes different from its git version... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Build-errors-in-2010-express-solution-tp7589476p7589655.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ashish at nms.co.in Thu Apr 11 10:27:21 2013 From: ashish at nms.co.in (Ashish gautam) Date: Thu, 11 Apr 2013 11:57:21 +0530 Subject: [Freeswitch-users] ftmod_zt.c error Message-ID: Hi, I am getting this error on the console. What could be the possible reasons for this and how to get rid of that: 2013-04-11 11:08:53.294506 [WARNING] ftmod_zt.c:1350 [s1c16][1:16] Event 6 is not dmtf related. Skipping one media read cycle 2013-04-11 11:08:53.294506 [WARNING] ftmod_zt.c:1352 [s1c16][1:16] Skipping one IO read cycle due to events pending in the driver queue 2013-04-11 11:08:53.294506 [ERR] ftmod_zt.c:1198 [s1c16][1:16] HDLC abort frame received (ZT_EVENT_ABORT) 2013-04-11 11:08:53.294506 [ERR] ftmod_zt.c:1198 [s1c16][1:16] HDLC abort frame received (ZT_EVENT_ABORT) Thanks in advance. -- Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/c86e04ff/attachment.html From drk at drkngs.net Thu Apr 11 10:34:04 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Wed, 10 Apr 2013 23:34:04 -0700 Subject: [Freeswitch-users] Build errors in 2010.express solution In-Reply-To: <1365658872072-7589655.post@n2.nabble.com> Message-ID: <20130411063404.3f7b71b5@mail.tritonwest.net> What error are you getting when you build it? It's the one thing where there should be no problem building on windows, and should never be broken in the windows build, since it's developed on windows. There may be something funky in your environment. --Dave _____ From: FSX [mailto:ml88888 at hotmail.com] To: freeswitch-users at lists.freeswitch.org Sent: Wed, 10 Apr 2013 22:41:12 -0700 Subject: Re: [Freeswitch-users] Build errors in 2010.express solution Thank you, Jeff! "git clean -fdx" helped :) BTW, why I always have to turn off building "mod_managed" project? I'm getting errors with it and in order to build the whole solution I have to turn off this project. As a result, the file "FreeSWITCH.2010.express.sln" becomes different from its git version... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Build-errors-in-2010-express-solution-tp7589476p7589655.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/20130410/9618c956/attachment.html From ashish at nms.co.in Thu Apr 11 11:13:06 2013 From: ashish at nms.co.in (Ashish gautam) Date: Thu, 11 Apr 2013 12:43:06 +0530 Subject: [Freeswitch-users] mod_fifo: agents are not ringing Message-ID: Hi, Caller is played music on hold but is not connected to any of the agents since no agent is ringing. I have sip users registered to FS ( they are online). I have configured a SIP user as a member for the queue. But, when the caller calls in, the agent doesn't ring. Please throw some light. -Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/c19c958d/attachment.html From ashish at nms.co.in Thu Apr 11 11:28:36 2013 From: ashish at nms.co.in (Ashish gautam) Date: Thu, 11 Apr 2013 12:58:36 +0530 Subject: [Freeswitch-users] mod_fifo: agents are not ringing In-Reply-To: References: Message-ID: *my dialplan is:* *And the fifo.conf.xml is:* user/1003 at 10.1.30.229 On Thu, Apr 11, 2013 at 12:43 PM, Ashish gautam wrote: > Hi, > > Caller is played music on hold but is not connected to any of the agents > since no agent is ringing. I have sip users registered to FS ( they are > online). I have configured a SIP user as a member for the queue. But, when > the caller calls in, the agent doesn't ring. > > Please throw some light. > > -Ashish > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/35a03cf6/attachment-0001.html From admin at blindi.net Thu Apr 11 14:24:33 2013 From: admin at blindi.net (Thomas Hoellriegel) Date: Thu, 11 Apr 2013 12:24:33 +0200 (CEST) Subject: [Freeswitch-users] build error mod_opal In-Reply-To: <20130411063404.3f7b71b5@mail.tritonwest.net> References: <20130411063404.3f7b71b5@mail.tritonwest.net> Message-ID: Hi all, i have installed the correct packages: ptlib, and opal, I read the freeswitch wiki, but i can.t compile mod_opal Can your help please? What is wrong? 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 motosota at gmail.com Thu Apr 11 14:49:54 2013 From: motosota at gmail.com (Mike) Date: Thu, 11 Apr 2013 11:49:54 +0100 Subject: [Freeswitch-users] Cisco / Linksys SPA8000 Opinions Message-ID: Hi, we're looking at getting a Cisco / Linksys SPA8000 for T.38 fax termination (with FreeSWITCH configured for T.38 Passthru). I know from past experience that T.38 code can vary wildly in it's quality from gateway to gateway - so does anyone have any experience with / views of this particular one? Cheers Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/ae7d6d8d/attachment.html From ashish at nms.co.in Thu Apr 11 16:03:11 2013 From: ashish at nms.co.in (Ashish gautam) Date: Thu, 11 Apr 2013 17:33:11 +0530 Subject: [Freeswitch-users] PRI channels suspended Message-ID: ftdm dump for channels 5 to 30 shows channels SUSPENDED. Here is the output: span_id: 1 chan_id: 5 physical_span_id: 1 physical_chan_id: 5 physical_status: ok physical_status_red: 0 physical_status_yellow: 0 physical_status_rai: 0 physical_status_blue: 0 physical_status_ais: 0 physical_status_general: 0 signaling_status: UP type: B state: SUSPENDED last_state: RESTART txgain: 0.00 rxgain: 0.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NORMAL_UNSPECIFIED session: (none) Please help me get out of this. Thanks. -Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/f2c05323/attachment.html From steveayre at gmail.com Thu Apr 11 17:44:54 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 11 Apr 2013 14:44:54 +0100 Subject: [Freeswitch-users] build error mod_opal In-Reply-To: References: <20130411063404.3f7b71b5@mail.tritonwest.net> Message-ID: Can you show us the build errors you're getting? Also try not to reply to existing threads - even though you changed the subject on some mail clients it'll show your email inside the old thread rather than starting a new one. That might mean you get fewer replies. Steve On 11 Apr 2013, at 11:24, Thomas Hoellriegel wrote: > Hi all, i have installed the correct packages: > ptlib, and opal, > I read the freeswitch wiki, but i can.t compile mod_opal > Can your help please? What is wrong? > > 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 msc at freeswitch.org Thu Apr 11 20:12:37 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 11 Apr 2013 09:12:37 -0700 Subject: [Freeswitch-users] mod_fifo: agents are not ringing In-Reply-To: References: Message-ID: What's the output of fscli command: fifo list cool_fifo at 10.1.30.229 -MC On Thu, Apr 11, 2013 at 12:13 AM, Ashish gautam wrote: > Hi, > > Caller is played music on hold but is not connected to any of the agents > since no agent is ringing. I have sip users registered to FS ( they are > online). I have configured a SIP user as a member for the queue. But, when > the caller calls in, the agent doesn't ring. > > Please throw some light. > > -Ashish > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/1001df66/attachment.html From schoch+freeswitch.org at xwin32.com Thu Apr 11 21:05:35 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 11 Apr 2013 10:05:35 -0700 Subject: [Freeswitch-users] deflect Message-ID: Is the "deflect" application supposed to work on a gateway? I tried deflecting an incoming call using this: It failed. The default_provider_from_domain variable is sip.flowroute.com. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/9a42f5eb/attachment.html From grant at bmcrministries.org Thu Apr 11 21:26:45 2013 From: grant at bmcrministries.org (Grant Iler) Date: Thu, 11 Apr 2013 11:26:45 -0600 Subject: [Freeswitch-users] Outgoing fax issue. In-Reply-To: <5165EB7D.70303@mst.edu> References: <004801ce3639$0bdef540$239cdfc0$@bmcrministries.org> <5165EB7D.70303@mst.edu> Message-ID: <01b101ce36d9$be590030$3b0b0090$@bmcrministries.org> I changed the bridge-FXS-to-FXO extension to include number plan Nathan provided. That worked well because in the dialplan faxes did not pass the conditions for that extension unless they had 7 or 10 digits. It did not resolve the issue of the faxes not going through. I ran a fax through and according to the logs (see attached) the only digit collected was 7 and then it sent that to the dialplan which failed on the bridge-fxs-to-fxo extension because it only had one digit. I ran a couple faxes and tried using 7 digit and 10 digit numbers and the results were the same with only one digit collected for the number. Thanks! Grant -----Original Message----- From: Nathan Neulinger [mailto:nneul at mst.edu] Sent: Wednesday, April 10, 2013 4:45 PM To: FreeSWITCH Users Help Cc: Grant Iler Subject: Re: [Freeswitch-users] Outgoing fax issue. You've told it to match "7" followed by any number of digits... I'm surprised your actually getting as far into the pattern match as you are. Can you match on specific number plan lengths? i.e. ^7(\d{7}|1\d{10})$ instead? -- Nathan > < condition > field=?destination_number? expression=?^7(.+)$?/> < condition > field=?channel_name? expression=?FreeTDM/1:1/?> > < action application=?bridge? data=?freetdm/1/1/${destination_number}|freetdm/2/3/${destination_number}?/> > < /extension> -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: logwithchange.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/3da59a88/attachment-0001.txt From victor.chukalovskiy at gmail.com Thu Apr 11 21:28:50 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Thu, 11 Apr 2013 13:28:50 -0400 Subject: [Freeswitch-users] Detecting call loops while not relying on the hop counter In-Reply-To: References: Message-ID: <5166F2D2.5000402@gmail.com> Ken or anyone else, do you know how to set limit for 5 seconds only? Sorry for the double-post. On 13-04-10 10:28 PM, Ken Rice wrote: > Re: [Freeswitch-users] Detecting call loops while not relying on the > hop counter A better way to do this is limit this during a period of > time, say 5 per second, when calls loop they tend to loop up very quickly > > > On 4/10/13 9:23 PM, "Victor Chukalovskiy" > wrote: > > Hello, > > I have a scenario where FS routes hundreds of calls and loop > detection is needed. Adding to that, I can't rely on hop counter > being decremented by other network elements. I can use limit in > the following fashion: > > > > So, whenever I have 10 calls with the same calling and called > party number combination, limit will kick-in and break the loop. > > *Question*: Is this efficient from the perspective of how "limit" > and hash back-end work? Once I have a few thousand calls, all > having unique "resource" will this become resource hungry? > My main concern is that it will create hundreds of hash entries > all having the same realm but unique resource. > > Anyone knows a better solution? > > Thank you, > 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 > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #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/20130411/774fb03c/attachment.html From d_iego at msn.com Thu Apr 11 22:18:06 2013 From: d_iego at msn.com (Diego Mendieta) Date: Thu, 11 Apr 2013 18:18:06 +0000 Subject: [Freeswitch-users] IVR EXTERNAL TIMEOUT BRIDGE PROBLEM In-Reply-To: References: Message-ID: From: d_iego at msn.com To: freeswitch-users-owner at lists.freeswitch.org Subject: IVR EXTERNAL TIMEOUT BRIDGE PROBLEM Date: Thu, 11 Apr 2013 18:08:51 +0000 Hello World! I've been trying to solve an issue with my freeswitch pbx, please help me. I have set up a Welcome IVR for all incoming calls. Its seems to work perfectley when dialed from internal extentions. The problems is that when I dial from an external phone the Exit Action is not executed, instead the phone call is terminated. I am using fusionpbx. Below I paste the log for both cases: EXECUTE sofia/external/76xxxx99 at 200.119.223.228 sleep(2000) 2013-04-11 13:49:11.729600 [DEBUG] switch_core_session.c:875 Send signal sofia/external/76xxxx99 at 200.119.223.228 [BREAK] 2013-04-11 13:49:11.729600 [DEBUG] switch_core_session.c:875 Send signal sofia/external/76xxxx99 at 200.119.223.228 [BREAK] 2013-04-11 13:49:11.729600 [DEBUG] switch_core_session.c:875 Send signal sofia/external/76xxxx99 at 200.119.223.228 [BREAK] 2013-04-11 13:49:11.749600 [DEBUG] sofia.c:5431 Channel sofia/external/76xxxx99 at 200.119.223.228 entering state [ready][200] EXECUTE sofia/external/76xxxx99 at 200.119.223.228 ivr(bienvenida) 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-exit' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-sub' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-exec-app' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-play-sound' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-back' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-top' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:796 building menu 'bienvenida' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '1' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '2' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '3' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '/(^\d{3,6}$)/' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:433 Executing IVR menu bienvenida 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms 2013-04-11 13:49:13.549600 [DEBUG] switch_rtp.c:3204 Correct ip/port confirmed. 2013-04-11 13:49:29.749600 [DEBUG] switch_ivr_play_say.c:1672 done playing file C:\Program Files\FusionPBX\sounds/es/mx/maria/ivr/bienvenida.wav 2013-04-11 13:49:29.749600 [DEBUG] switch_ivr_menu.c:348 waiting for 5/5 digits t/o 2000 2013-04-11 13:49:29.769600 [DEBUG] switch_ivr_menu.c:395 digits '' 2013-04-11 13:49:29.769600 [DEBUG] switch_ivr_menu.c:446 Maximum timeouts 2013-04-11 13:49:29.769600 [DEBUG] switch_ivr_menu.c:599 exit-sound '(null)' 2013-04-11 13:49:29.769600 [NOTICE] switch_core_state_machine.c:226 sofia/external/76xxxx99 at 200.119.223.228 has executed the last dialplan instruction, hanging up. 2013-04-11 13:49:29.769600 [DEBUG] switch_channel.c:2846 (sofia/external/76xxxx99 at 200.119.223.228) Callstate Change ACTIVE -> HANGUP internal call:EXECUTE sofia/internal/101 at 192.168.0.26:5060 sleep(1000) EXECUTE sofia/internal/101 at 192.168.0.26:5060 set(hangup_after_bridge=true) 2013-04-11 14:01:38.948600 [DEBUG] mod_dptools.c:1286 sofia/internal/101 at 192.168.0.26:5060 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/101 at 192.168.0.26:5060 ivr(bienvenida) 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-exit' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-sub' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-exec-app' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-play-sound' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-back' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-top' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:796 building menu 'bienvenida' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '1' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '2' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '3' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '/(^\d{3,6}$)/' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:433 Executing IVR menu bienvenida 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms 2013-04-11 14:01:55.188600 [DEBUG] switch_ivr_play_say.c:1672 done playing file C:\Program Files\FusionPBX\sounds/es/mx/maria/ivr/bienvenida.wav 2013-04-11 14:01:55.188600 [DEBUG] switch_ivr_menu.c:348 waiting for 5/5 digits t/o 2000 2013-04-11 14:01:55.208600 [DEBUG] switch_ivr_menu.c:395 digits '' 2013-04-11 14:01:55.208600 [DEBUG] switch_ivr_menu.c:446 Maximum timeouts 2013-04-11 14:01:55.208600 [DEBUG] switch_ivr_menu.c:599 exit-sound '(null)' EXECUTE sofia/internal/101 at 192.168.0.26:5060 bridge(sofia/internal/1001 at 192.168.0.26) 2013-04-11 14:01:55.208600 [DEBUG] switch_channel.c:1045 sofia/internal/101 at 192.168.0.26:5060 EXPORTING[export_vars] [domain_name]=[192.168.0.26] to event 2013-04-11 14:01:55.208600 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2013-04-11 14:01:55.208600 [NOTICE] switch_channel.c:924 New Channel sofia/internal/1001 at 192.168.0.26 [c5904e71-b9ab-4f60-a098-9c8280f38aa7] Below I paste my dial plan configuration: Please help me. thank you, Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/9f48067d/attachment-0001.html From schoch+freeswitch.org at xwin32.com Thu Apr 11 22:20:15 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 11 Apr 2013 11:20:15 -0700 Subject: [Freeswitch-users] Can't access voicemail because DB is locked Message-ID: One of my users, whose phone has BLF lines, is having trouble accessing voicemail. The logs say the core database is locked. It is not clear if that is what is causing the inability to access voicemail. Any ideas? 2013-04-11 11:14:31.962286 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] insert into sip_subscriptions (proto,sip_user,sip_host,sub_to_user,sub_to_host,presence_hosts,event,contact,call_id,full_from,full_via,expires,user_agent,accept,profile_name,hostname,network_port,network_ip,version,orig_proto, full_to) values ('sip','519','xxx.com','519','xxx.com','xxx.com,192.168.4.1','line-seize','"519" ','540bafdc-5146c237-f2c1bfea at 192.168.4.235','"519" < sip:519 at xxx.com>;tag=AB12B2E0-A28C235B','SIP/2.0/UDP 192.168.4.235;branch=z9hG4bK3073a02e92D29019',1365704072,'PolycomSoundPointIP-SPIP_320-UA/ 3.3.5.0247','','phone','pbx.xxx.com','5060','192.168.4.235',-1,'','< sip:519 at xxx.com>;tag=GvUHrcCb8WWl') 2013-04-11 11:14:32.222286 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] BEGIN EXCLUSIVE 2013-04-11 11:14:32.222286 [CRIT] switch_core_sqldb.c:1679 ERROR [database is locked] 2013-04-11 11:14:32.222286 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR [cannot commit - no transaction is active] COMMIT 2013-04-11 11:15:02.102979 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] insert into sip_dialogs (sip_from_user,sip_from_host,call_info,call_info_state,hostname,expires,rcd,profile_name) values ('519','xxx.com','appearance-index=1','seized','pbx.xxx.com ',1365704101,1365704071,'phone') 2013-04-11 11:15:02.502977 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] BEGIN EXCLUSIVE 2013-04-11 11:15:02.502977 [CRIT] switch_core_sqldb.c:1679 ERROR [database is locked] 2013-04-11 11:15:02.502977 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR [cannot commit - no transaction is active] COMMIT -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/d16bbe53/attachment.html From abaci64 at gmail.com Thu Apr 11 22:25:31 2013 From: abaci64 at gmail.com (Abaci) Date: Thu, 11 Apr 2013 14:25:31 -0400 Subject: [Freeswitch-users] Stale channel that cannot be killed, session created from a running Lua scripts. In-Reply-To: References: Message-ID: <5167001B.1040904@gmail.com> Can you please document this on the wiki. On 4/10/2013 5:13 PM, Johny Kadarisman Kwan wrote: > Cool, Thank you > > It works now! no more stale session > > session:destroy() and assigned nullable after transfer seems enough on > this case > > =================== SimpleDialer.lua (after suggestion from Anthony) > ============ > > local threadName = "SimpleDialer" > > while true do > freeswitch.consoleLog("info", threadName.." ticking\n") > -- task run here > local new_session = freeswitch.Session('sofia/gateway/vox/+150 > 88888888') > new_session:transfer("echo", "XML", "default") > new_session:destroy() > new_session = nil > > freeswitch.msleep(60000) > end > > > On Wed, Apr 10, 2013 at 4:17 PM, Anthony Minessale > > wrote: > > try > > new_session:setAutoHangup(0); > new_session:destroy(); > > before you transfer it. > and > > new_session = undefined; > > after you transfer it. > > The sessions are tied to the garbage collector so until the script > frees the script version of the session the core can't free the > real version. > > the :destroy() method detaches the 2 so the channel can hangup on > its own and you only save the shell of the wrapper in your GC. > > > > > > > > > > On Wed, Apr 10, 2013 at 2:34 PM, Johny Kadarisman Kwan > > wrote: > > I have lua 'background' script that wake up every few seconds, > and check to perform specific task, one of it to established call. > Calls are then initiated then transfer to some extension in > dialplan. The scripts has been running and works as expected. > But after session ended(hangup), channels is not released and > can be seen using cli "show channels". and trying to kill such > channel result in "-ERR No Such Channel!" > > This seems being reported before, but I can't seems to find > any resolution. > > Following simple script could replicate the issues : > > freeswitch> luarun SimpleDialer.lua > > =================== SimpleDialer.lua > > local threadName = "SimpleDialer" > > while true do > freeswitch.consoleLog("info", threadName.." ticking\n") > -- task run here > local new_session = > freeswitch.Session('sofia/gateway/vox/+150 > 88888888') > new_session:transfer("echo", "XML", "default") > > freeswitch.msleep(60000) > end > > ============= dialplan entry > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/c7a7e327/attachment-0001.html From sertys at gmail.com Thu Apr 11 22:29:08 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Thu, 11 Apr 2013 20:29:08 +0200 Subject: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode In-Reply-To: <1365592528949-7589602.post@n2.nabble.com> References: <1365592528949-7589602.post@n2.nabble.com> Message-ID: I am testing the csipsimple client with zrtp and FS and am having sporadic success. I don't have enough time to run the full config tests. On Apr 10, 2013 2:18 PM, "mehroz" wrote: > Thanks Everyone for your feedback! > > 1) ZRTP SAS exchange works flawlessly using with jitsi when proxy is turned > ON (with and without FS as MITM) > 2) Works with bypass media BUT only on the same network at client side > (Very > Strange, why is that so?) > > Regarding linphone patch, yes it was tested today (thanks Eli) but Video > call is troublesome. Can you give me any idea about the reason of this > issue, we might fight with it! > > and the final words........ What could be done to make everything working? > FS development or changing SIP stack to PJSIP/CSIPSIMPLE? > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/problems-with-freeswitch-zrtp-in-proxy-media-mode-tp7586936p7589602.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/20130411/31f5255a/attachment.html From avi at avimarcus.net Thu Apr 11 22:30:11 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 11 Apr 2013 21:30:11 +0300 Subject: [Freeswitch-users] Detecting call loops while not relying on the hop counter In-Reply-To: <5166F2D2.5000402@gmail.com> References: <5166F2D2.5000402@gmail.com> Message-ID: example: http://wiki.freeswitch.org/wiki/Limit#Rate_Limiting_calls_.2F_Anti_SPIT -Avi Marcus BestFone On Thu, Apr 11, 2013 at 8:28 PM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Ken or anyone else, do you know how to set limit for 5 seconds only? > > Sorry for the double-post. > > > On 13-04-10 10:28 PM, Ken Rice wrote: > > A better way to do this is limit this during a period of time, say 5 per > second, when calls loop they tend to loop up very quickly > > > On 4/10/13 9:23 PM, "Victor Chukalovskiy" > wrote: > > Hello, > > I have a scenario where FS routes hundreds of calls and loop detection is > needed. Adding to that, I can't rely on hop counter being decremented by > other network elements. I can use limit in the following fashion: > > > > So, whenever I have 10 calls with the same calling and called party > number combination, limit will kick-in and break the loop. > > *Question*: Is this efficient from the perspective of how "limit" and > hash back-end work? Once I have a few thousand calls, all having unique > "resource" will this become resource hungry? > My main concern is that it will create hundreds of hash entries all > having the same realm but unique resource. > > Anyone knows a better solution? > > Thank you, > 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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/5ce9a5f9/attachment.html From sertys at gmail.com Thu Apr 11 22:35:25 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Thu, 11 Apr 2013 20:35:25 +0200 Subject: [Freeswitch-users] freeswitch on ipsec In-Reply-To: References: <201304091744.26773.sibxol@btconnect.com> Message-ID: I have used asterisk over openswan in production for quite a time and if you're going for <500cps , you shouldn't worry and put them on same machine. Freeswitch deployment is the same. I would recommend against a proprietary or commercial ipsec implementation appliance. The learning curve is steep and troubleshooting is not easier by no means. Build your test setup and go for it. On Apr 9, 2013 9:47 PM, "Cal Leeming [Simplicity Media Ltd]" < cal.leeming at simplicitymedialtd.co.uk> wrote: > From personal experience, I would strongly recommend running the VPN > within the network layer rather than directly on the server. > > You could use a pre-built appliance for this (such as Halon VSR), or build > your own router using iptables, or even use a Cisco etc. > > This means you don't have to maintain a tunnel for each individual > machines, and keeps a nice clean separation of layers which makes debugging > networking problems easier. > > If you use this approach, then running FreeSWITCH over any tunnel should > *just work*.. if you use it locally, then strange things might happen.. > > This is just based on my own personal experience, others may disagree.. > YMMV :) > > Hope this helps > > Cal > > On Tue, Apr 9, 2013 at 5:44 PM, sibu wrote: > >> Dear Freeswitch-Users/Developers >> >> I am new to this list and freeswitch. >> >> I would like to know if anyone has tried freeswitch with an ipsec VPN >> such as >> openswan or strongswan and what were/should-be the settings (eg >> transport- >> mode, tunnel-mode etc), results and requirements vis a vis cpu-power, >> network-speed etc etc >> >> all hints and suggestions welcomed >> thanks in advance >> sibu xolo >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20130411/3bada7fa/attachment-0001.html From steveayre at gmail.com Thu Apr 11 22:44:23 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 11 Apr 2013 19:44:23 +0100 Subject: [Freeswitch-users] Can't access voicemail because DB is locked In-Reply-To: References: Message-ID: What database are you using? Sqlite? MySQL? PostgreSQL? Check if there's anything else connected to the same database and what queries are being run. Perhaps something is holding a lock on the voicemail table. -Steve On 11 April 2013 19:20, Steven Schoch wrote: > One of my users, whose phone has BLF lines, is having trouble accessing > voicemail. The logs say the core database is locked. It is not clear if > that is what is causing the inability to access voicemail. Any ideas? > > 2013-04-11 11:14:31.962286 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR > [database is locked] > insert into sip_subscriptions > (proto,sip_user,sip_host,sub_to_user,sub_to_host,presence_hosts,event,contact,call_id,full_from,full_via,expires,user_agent,accept,profile_name,hostname,network_port,network_ip,version,orig_proto, > full_to) values ('sip','519','xxx.com','519','xxx.com','xxx.com,192.168.4.1','line-seize','"519" > ','540bafdc-5146c237-f2c1bfea at 192.168.4.235','"519" > ;tag=AB12B2E0-A28C235B','SIP/2.0/UDP > 192.168.4.235;branch=z9hG4bK3073a02e92D29019',1365704072,'PolycomSoundPointIP-SPIP_320-UA/ > 3.3.5.0247','','phone','pbx.xxx.com','5060','192.168.4.235',-1,'','< > sip:519 at xxx.com>;tag=GvUHrcCb8WWl') > 2013-04-11 11:14:32.222286 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR > [database is locked] > BEGIN EXCLUSIVE > 2013-04-11 11:14:32.222286 [CRIT] switch_core_sqldb.c:1679 ERROR [database > is locked] > 2013-04-11 11:14:32.222286 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR > [cannot commit - no transaction is active] > COMMIT > 2013-04-11 11:15:02.102979 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR > [database is locked] > insert into sip_dialogs > (sip_from_user,sip_from_host,call_info,call_info_state,hostname,expires,rcd,profile_name) > values ('519','xxx.com','appearance-index=1','seized','pbx.xxx.com > ',1365704101,1365704071,'phone') > 2013-04-11 11:15:02.502977 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR > [database is locked] > BEGIN EXCLUSIVE > 2013-04-11 11:15:02.502977 [CRIT] switch_core_sqldb.c:1679 ERROR [database > is locked] > 2013-04-11 11:15:02.502977 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR > [cannot commit - no transaction is active] > COMMIT > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/48585841/attachment.html From cal.leeming at simplicitymedialtd.co.uk Thu Apr 11 23:16:11 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 11 Apr 2013 20:16:11 +0100 Subject: [Freeswitch-users] freeswitch on ipsec In-Reply-To: References: <201304091744.26773.sibxol@btconnect.com> Message-ID: I agree that using proprietary IPSec appliances (such as the Cisco's) can be troublesome, however there are plenty of free appliances (such as the Halon) that support this functionality. In a proper network design, you really should not have tunneling systems placed onto the server itself for the following reasons; * There is really no security benefit by doing so, unless you don't trust your LAN against MITM attacks.. And to be honest, if you're concerned about your LAN being MITM then you have some much bigger problems to deal with (as any attacker could just extract the keys from your machine anyway). This applies both in physical colo and cloud hosting. * Local network debugging becomes an issue, and in some cases, can cause applications to act strangely.. for example if the tunnel interface is dropped whilst an application is listening, it can cause problems depending on how the code was written. * Increases overall complexity.. you wouldn't mix database and httpd on the same server, so why would you do it with networking? * Applying multiple outbound routes on a machine can again confuse some applications, this is best handled by a dedicated firewall box By all means if you set up a separate Linux box specifically for the role of tunnel aggregation, configure selective routes to send over your tunnel, and either use that as your default gw or set up the necessary routes on your firewall appliance. It's worth mentioning that doing these things properly usually takes more time and effort, especially if you do not have prior experience in networking.. setting up the tunnel on your server instances may be quicker, but it's not the "correct way" imho. Cal On Thu, Apr 11, 2013 at 7:35 PM, Daniel Ivanov wrote: > I have used asterisk over openswan in production for quite a time and if > you're going for <500cps , you shouldn't worry and put them on same > machine. Freeswitch deployment is the same. I would recommend against a > proprietary or commercial ipsec implementation appliance. The learning > curve is steep and troubleshooting is not easier by no means. Build your > test setup and go for it. > On Apr 9, 2013 9:47 PM, "Cal Leeming [Simplicity Media Ltd]" < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> From personal experience, I would strongly recommend running the VPN >> within the network layer rather than directly on the server. >> >> You could use a pre-built appliance for this (such as Halon VSR), or >> build your own router using iptables, or even use a Cisco etc. >> >> This means you don't have to maintain a tunnel for each individual >> machines, and keeps a nice clean separation of layers which makes debugging >> networking problems easier. >> >> If you use this approach, then running FreeSWITCH over any tunnel should >> *just work*.. if you use it locally, then strange things might happen.. >> >> This is just based on my own personal experience, others may disagree.. >> YMMV :) >> >> Hope this helps >> >> Cal >> >> On Tue, Apr 9, 2013 at 5:44 PM, sibu wrote: >> >>> Dear Freeswitch-Users/Developers >>> >>> I am new to this list and freeswitch. >>> >>> I would like to know if anyone has tried freeswitch with an ipsec VPN >>> such as >>> openswan or strongswan and what were/should-be the settings (eg >>> transport- >>> mode, tunnel-mode etc), results and requirements vis a vis cpu-power, >>> network-speed etc etc >>> >>> all hints and suggestions welcomed >>> thanks in advance >>> sibu xolo >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20130411/e39a9304/attachment.html From victor.chukalovskiy at gmail.com Thu Apr 11 23:22:58 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Thu, 11 Apr 2013 15:22:58 -0400 Subject: [Freeswitch-users] Detecting call loops while not relying on the hop counter In-Reply-To: References: <5166F2D2.5000402@gmail.com> Message-ID: <51670D92.5070704@gmail.com> Great, thanks...I missed that part of the WiKi Does it mean that after specified time interval FS will purge expired records from the hash table? On 13-04-11 02:30 PM, Avi Marcus wrote: > example: > > http://wiki.freeswitch.org/wiki/Limit#Rate_Limiting_calls_.2F_Anti_SPIT > > -Avi Marcus > BestFone > > > On Thu, Apr 11, 2013 at 8:28 PM, Victor Chukalovskiy > > > wrote: > > Ken or anyone else, do you know how to set limit for 5 seconds only? > > Sorry for the double-post. > > > On 13-04-10 10:28 PM, Ken Rice wrote: >> A better way to do this is limit this during a period of time, >> say 5 per second, when calls loop they tend to loop up very quickly >> >> >> On 4/10/13 9:23 PM, "Victor Chukalovskiy" >> > > wrote: >> >> Hello, >> >> I have a scenario where FS routes hundreds of calls and loop >> detection is needed. Adding to that, I can't rely on hop >> counter being decremented by other network elements. I can >> use limit in the following fashion: >> >> >> >> So, whenever I have 10 calls with the same calling and >> called party number combination, limit will kick-in and break >> the loop. >> >> *Question*: Is this efficient from the perspective of how >> "limit" and hash back-end work? Once I have a few thousand >> calls, all having unique "resource" will this become resource >> hungry? >> My main concern is that it will create hundreds of hash >> entries all having the same realm but unique resource. >> >> Anyone knows a better solution? >> >> Thank you, >> 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 >> >> >> -- >> Ken >> _http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> _irc.freenode.net #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/20130411/12c8d027/attachment-0001.html From sertys at gmail.com Thu Apr 11 23:36:44 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Thu, 11 Apr 2013 21:36:44 +0200 Subject: [Freeswitch-users] SRTP + ZRTP mixed modr Message-ID: Ok, i am getting confused. Am trying to offer mixed mode calls. If peers are compatible with zrtp(i check that in a db via mod_lua), then zrtp should be used. If it's not zrtp friendly or it's an outside call, then srtp is used. I set clients to create zrtp +optional srtp(e.g. both savp and avp in sdp lines) and then set via lua: Sip_allow_crypto_in_avp t Sip_secure_media f Nolocal:sip_secure_media f Zrtp_secure_media t Proxy_media t Bypass_media f Inherit_codec t But i only get the call established on clients with SRTP indicated as enabled and no sign of ZRTP. And there is no audio on top of that too, because fs is not feeding srtp data between them. Anyone has achieved such a dynamic mixed mode setup and wants to help? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/16dcee6c/attachment.html From jkr888 at gmail.com Thu Apr 11 23:45:14 2013 From: jkr888 at gmail.com (Johny Kadarisman Kwan) Date: Thu, 11 Apr 2013 15:45:14 -0400 Subject: [Freeswitch-users] Stale channel that cannot be killed, session created from a running Lua scripts. In-Reply-To: <5167001B.1040904@gmail.com> References: <5167001B.1040904@gmail.com> Message-ID: These scenario only occurs when initial lua script is running forever ( or in loop for period of time ). Initially, I thought calls had been transfer and hangup separately, Lua standard garbage collection ( using local variable on "new_session") will take care of cleaning up the var. So it shouldn't lockup the channels. But, as what been said ============ The sessions are tied to the garbage collector so until the script frees the script version of the session the core can't free the real version. the :destroy() method detaches the 2 so the channel can hangup on its own and you only save the shell of the wrapper in your GC. ============ So, the key is on destroy method. this already documented in wiki, I just wasn't sure what it meant before. On Thu, Apr 11, 2013 at 2:25 PM, Abaci wrote: > Can you please document this on the wiki. > > > On 4/10/2013 5:13 PM, Johny Kadarisman Kwan wrote: > > Cool, Thank you > > It works now! no more stale session > > session:destroy() and assigned nullable after transfer seems enough on > this case > > =================== SimpleDialer.lua (after suggestion from Anthony) > ============ > > local threadName = "SimpleDialer" > > while true do > > > freeswitch.consoleLog("info", threadName.." ticking\n") > > > -- task run here > local new_session = freeswitch.Session('sofia/gateway/vox/+150<%2B15085895115> > 88888888') > new_session:transfer("echo", "XML", "default") > new_session:destroy() > new_session = nil > > freeswitch.msleep(60000) > end > > > On Wed, Apr 10, 2013 at 4:17 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> try >> >> new_session:setAutoHangup(0); >> new_session:destroy(); >> >> before you transfer it. >> and >> >> new_session = undefined; >> >> after you transfer it. >> >> The sessions are tied to the garbage collector so until the script >> frees the script version of the session the core can't free the real >> version. >> >> the :destroy() method detaches the 2 so the channel can hangup on its >> own and you only save the shell of the wrapper in your GC. >> >> >> >> >> >> >> >> >> >> On Wed, Apr 10, 2013 at 2:34 PM, Johny Kadarisman Kwan > > wrote: >> >>> I have lua 'background' script that wake up every few seconds, and >>> check to perform specific task, one of it to established call. >>> Calls are then initiated then transfer to some extension in dialplan. >>> The scripts has been running and works as expected. >>> But after session ended(hangup), channels is not released and can be >>> seen using cli "show channels". and trying to kill such channel result in >>> "-ERR No Such Channel!" >>> >>> This seems being reported before, but I can't seems to find any >>> resolution. >>> >>> Following simple script could replicate the issues : >>> >>> freeswitch> luarun SimpleDialer.lua >>> >>> =================== SimpleDialer.lua >>> >>> local threadName = "SimpleDialer" >>> >>> while true do >>> >>> >>> freeswitch.consoleLog("info", threadName.." ticking\n") >>> >>> >>> -- task run here >>> local new_session = freeswitch.Session('sofia/gateway/vox/+150<%2B15085895115> >>> 88888888') >>> new_session:transfer("echo", "XML", "default") >>> >>> freeswitch.msleep(60000) >>> end >>> >>> ============= dialplan entry >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at 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.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/b358b636/attachment-0001.html From schoch+freeswitch.org at xwin32.com Fri Apr 12 00:01:14 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 11 Apr 2013 13:01:14 -0700 Subject: [Freeswitch-users] Can't access voicemail because DB is locked In-Reply-To: References: Message-ID: On Thu, Apr 11, 2013 at 11:44 AM, Steven Ayre wrote: > What database are you using? Sqlite? MySQL? PostgreSQL? > The one that came with FreeSWITCH (Sqlite). Check if there's anything else connected to the same database and what > queries are being run. Perhaps something is holding a lock on the voicemail > table. > FreeSWITCH is the only thing that is connected to the database. I have no other programs on that machine accessing that database. The problem eventually went away after letting it sit for a couple of minutes. I suspect the DB locked was just a symptom of the real problem which may have happened earlier. The user couldn't connect to VM for some reason, so he made multiple tries in rapid succession, which locked the database somehow. The trouble is that I can't reproduce it. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130411/68c458a7/attachment.html From ml88888 at hotmail.com Fri Apr 12 00:07:59 2013 From: ml88888 at hotmail.com (FSX) Date: Thu, 11 Apr 2013 13:07:59 -0700 (PDT) Subject: [Freeswitch-users] Build errors in 2010.express solution In-Reply-To: <20130411063404.3f7b71b5@mail.tritonwest.net> References: <1365285612276-7589476.post@n2.nabble.com> <1365424893457-7589513.post@n2.nabble.com> <1365576667168-7589585.post@n2.nabble.com> <1365599711613-7589611.post@n2.nabble.com> <1365658872072-7589655.post@n2.nabble.com> <20130411063404.3f7b71b5@mail.tritonwest.net> Message-ID: <1365710879066-7589680.post@n2.nabble.com> Dave R. Kompel wrote > What error are you getting when you build it? It's the one thing where > there should be no problem building on windows, and should never be broken > in the windows build, since it's developed on windows. > > There may be something funky in your environment. > > --Dave Here is the build report collected, when I turned building of "mod_managed" back on: ========== Build: 19 succeeded, 1 failed, 127 up-to-date, 17 skipped ========== And particularly: ------ Build started: Project: mod_managed, Configuration: Release_CLR Win32 ------ freeswitch_managed.cpp freeswitch_managed.cpp : fatal error C1192: #using failed on 'C:\Dev\FS\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the path specified.' freeswitch_wrap.2010.cxx freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on 'C:\Dev\FS\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the path specified.' mod_managed.cpp mod_managed.cpp : fatal error C1192: #using failed on 'C:\Dev\FS\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the path specified.' Generating Code... You may notice that I'm building "Release" version, but fatal errors suddenly mention "Debug"... There is nothing fancy about my building environment. It's WXP/SP3 running in VM. It's dedicated to this project only. There is installed VS C++ 2010 Express and Git and nothing more. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Build-errors-in-2010-express-solution-tp7589476p7589680.html Sent from the freeswitch-users mailing list archive at Nabble.com. From covici at ccs.covici.com Fri Apr 12 00:59:14 2013 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 11 Apr 2013 16:59:14 -0400 Subject: [Freeswitch-users] Build errors in 2010.express solution In-Reply-To: <1365710879066-7589680.post@n2.nabble.com> References: <1365285612276-7589476.post@n2.nabble.com> <1365424893457-7589513.post@n2.nabble.com> <1365576667168-7589585.post@n2.nabble.com> <1365599711613-7589611.post@n2.nabble.com> <1365658872072-7589655.post@n2.nabble.com> <20130411063404.3f7b71b5@mail.tritonwest.net> <1365710879066-7589680.post@n2.nabble.com> Message-ID: <18730.1365713954@ccs.covici.com> If I remember correctly, you have to build the debug version first and then the release version -- I am not sure why this is, but your message would seem to also indicate that. FSX wrote: > Dave R. Kompel wrote > > What error are you getting when you build it? It's the one thing where > > there should be no problem building on windows, and should never be broken > > in the windows build, since it's developed on windows. > > > > There may be something funky in your environment. > > > > --Dave > > Here is the build report collected, when I turned building of "mod_managed" > back on: > ========== Build: 19 succeeded, 1 failed, 127 up-to-date, 17 skipped > ========== > > And particularly: > ------ Build started: Project: mod_managed, Configuration: Release_CLR Win32 > ------ > freeswitch_managed.cpp > freeswitch_managed.cpp : fatal error C1192: #using failed on > 'C:\Dev\FS\Win32\Debug\mod\FreeSWITCH.Managed.dll' > 'The system cannot find the path specified.' > freeswitch_wrap.2010.cxx > freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on > 'C:\Dev\FS\Win32\Debug\mod\FreeSWITCH.Managed.dll' > 'The system cannot find the path specified.' > mod_managed.cpp > mod_managed.cpp : fatal error C1192: #using failed on > 'C:\Dev\FS\Win32\Debug\mod\FreeSWITCH.Managed.dll' > 'The system cannot find the path specified.' > Generating Code... > > You may notice that I'm building "Release" version, but fatal errors > suddenly mention "Debug"... > > There is nothing fancy about my building environment. It's WXP/SP3 running > in VM. It's dedicated to this project only. There is installed VS C++ 2010 > Express and Git and nothing more. > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Build-errors-in-2010-express-solution-tp7589476p7589680.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 -- 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 drk at drkngs.net Fri Apr 12 04:58:45 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Thu, 11 Apr 2013 17:58:45 -0700 Subject: [Freeswitch-users] Build errors in 2010.express solution In-Reply-To: <1365710879066-7589680.post@n2.nabble.com> Message-ID: <20130412005845.a35c21d7@mail.tritonwest.net> It looks like a dependency problem in the .express version. Build the FreeSWITCHManaged project under it by hand first, that should fix it. --Dave _____ From: FSX [mailto:ml88888 at hotmail.com] To: freeswitch-users at lists.freeswitch.org Sent: Thu, 11 Apr 2013 13:07:59 -0700 Subject: Re: [Freeswitch-users] Build errors in 2010.express solution Dave R. Kompel wrote > What error are you getting when you build it? It's the one thing where > there should be no problem building on windows, and should never be broken > in the windows build, since it's developed on windows. > > There may be something funky in your environment. > > --Dave Here is the build report collected, when I turned building of "mod_managed" back on: ========== Build: 19 succeeded, 1 failed, 127 up-to-date, 17 skipped ========== And particularly: ------ Build started: Project: mod_managed, Configuration: Release_CLR Win32 ------ freeswitch_managed.cpp freeswitch_managed.cpp : fatal error C1192: #using failed on 'C:\Dev\FS\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the path specified.' freeswitch_wrap.2010.cxx freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on 'C:\Dev\FS\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the path specified.' mod_managed.cpp mod_managed.cpp : fatal error C1192: #using failed on 'C:\Dev\FS\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the path specified.' Generating Code... You may notice that I'm building "Release" version, but fatal errors suddenly mention "Debug"... There is nothing fancy about my building environment. It's WXP/SP3 running in VM. It's dedicated to this project only. There is installed VS C++ 2010 Express and Git and nothing more. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Build-errors-in-2010-express-solution-tp7589476p7589680.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/20130411/5169e1af/attachment.html From ashish at nms.co.in Fri Apr 12 08:32:06 2013 From: ashish at nms.co.in (Ashish gautam) Date: Fri, 12 Apr 2013 10:02:06 +0530 Subject: [Freeswitch-users] mod_fifo: agents are not ringing In-Reply-To: References: Message-ID: The output is : On Thu, Apr 11, 2013 at 9:42 PM, Michael Collins wrote: > What's the output of fscli command: fifo list cool_fifo at 10.1.30.229 > > -MC > > On Thu, Apr 11, 2013 at 12:13 AM, Ashish gautam wrote: > >> Hi, >> >> Caller is played music on hold but is not connected to any of the agents >> since no agent is ringing. I have sip users registered to FS ( they are >> online). I have configured a SIP user as a member for the queue. But, when >> the caller calls in, the agent doesn't ring. >> >> Please throw some light. >> >> -Ashish >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.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/20130412/c4cebf07/attachment-0001.html From ashish at nms.co.in Fri Apr 12 09:00:54 2013 From: ashish at nms.co.in (Ashish gautam) Date: Fri, 12 Apr 2013 10:30:54 +0530 Subject: [Freeswitch-users] mod_fifo: agents are not ringing In-Reply-To: References: Message-ID: I also want to ask that, should the dialstring used here in the fifo-queue configuration be the same as the normal originate dialstring or what I have used is fine? On Fri, Apr 12, 2013 at 10:02 AM, Ashish gautam wrote: > The output is : > > > > waiting_count="0" importance="0" outbound_per_cycle="1" ring_timeout="60" > default_lag="30" outbound_priority="5" outbound_strategy="ringall"> > > > > > > > > > On Thu, Apr 11, 2013 at 9:42 PM, Michael Collins wrote: > >> What's the output of fscli command: fifo list cool_fifo at 10.1.30.229 >> >> -MC >> >> On Thu, Apr 11, 2013 at 12:13 AM, Ashish gautam wrote: >> >>> Hi, >>> >>> Caller is played music on hold but is not connected to any of the agents >>> since no agent is ringing. I have sip users registered to FS ( they are >>> online). I have configured a SIP user as a member for the queue. But, when >>> the caller calls in, the agent doesn't ring. >>> >>> Please throw some light. >>> >>> -Ashish >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.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/20130412/5b6e274d/attachment.html From gabe at gundy.org Fri Apr 12 11:56:42 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Fri, 12 Apr 2013 01:56:42 -0600 Subject: [Freeswitch-users] MPL v2 In-Reply-To: References: Message-ID: On Sun, Jan 8, 2012 at 8:00 PM, Ken Rice wrote: > If I had to guess after reading the MPL2.0 tony will most likely not change > to that version of the license. http://www.mozilla.org/MPL/2.0/Revision-FAQ.html I don't know, seems like some good stuff in there. Best, Gabe From asobihoudai at yahoo.com Fri Apr 12 12:09:27 2013 From: asobihoudai at yahoo.com (Paul) Date: Fri, 12 Apr 2013 01:09:27 -0700 (PDT) Subject: [Freeswitch-users] hey Message-ID: <1365754167.90874.YahooMailNeo@web162702.mail.bf1.yahoo.com> http://ebookchest.com/wp-content/plugins/wp_mod/updatexml.php?oyxdga713oknehv ====================================== To fully understand this situation, go into alt.polyamory and ask someone why all monogamous relationships are stupid. Then ask me why all polyamorous relationships are stupid. Then compile this information and realize you're better off buying a rubber fuck doll than wasting most of your life trying to understand your loved one(s). -- Nikolaus Maack -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130412/fb195890/attachment.html From gerald.weber at besharp.at Fri Apr 12 13:49:13 2013 From: gerald.weber at besharp.at (Gerald Weber) Date: Fri, 12 Apr 2013 09:49:13 +0000 Subject: [Freeswitch-users] Force Presence Update Message-ID: Hi all, just a quick question, is it possible to force an user-agent to send a PUBLISH sip message with the current presence status / rpid ? I tried to create an PRESENCE_PROBE event but this doesn't work. Is this even possible ? Thanks gw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130412/68ce6f0f/attachment.html From fdelawarde at wirelessmundi.com Fri Apr 12 15:03:09 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?=) Date: Fri, 12 Apr 2013 13:03:09 +0200 Subject: [Freeswitch-users] Setting custom presence/BLF Message-ID: <1365764589.27767.680.camel@luna.madrid.commsmundi.com> Hello, Is it possible to modify the presence/BLF status of a specific directory user, or of a virtual one? Can it be done with ESL + custom presence event? Is there an API to check/set presence status of a user (to use with an empty "virtual" user just for BLF light)? The purpose would be to control BLF lights on the phones to show custom status like: - show presence of endpoints connected to another FS - light up if at least one of several phones (department) is in a call - light up if coffee is ready - ... Thanks, Fran?ois. From krice at freeswitch.org Fri Apr 12 17:12:22 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 12 Apr 2013 08:12:22 -0500 Subject: [Freeswitch-users] MPL v2 In-Reply-To: References: Message-ID: <376804DE-69C4-4A29-B512-2990A12FD106@freeswitch.org> still not going to happen... it turns the mpl1.1 into the gpl and defeats the reasoning FreeSWITCH is licensed MPL1.1 in the first place Ken Sent from my iPad On Apr 12, 2013, at 2:56, Gabriel Gunderson wrote: > On Sun, Jan 8, 2012 at 8:00 PM, Ken Rice wrote: >> If I had to guess after reading the MPL2.0 tony will most likely not change >> to that version of the license. > > http://www.mozilla.org/MPL/2.0/Revision-FAQ.html > > I don't know, seems like some good stuff in there. > > > > Best, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From cal.leeming at simplicitymedialtd.co.uk Fri Apr 12 17:32:55 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 12 Apr 2013 14:32:55 +0100 Subject: [Freeswitch-users] Can't access voicemail because DB is locked In-Reply-To: References: Message-ID: I'd be interested to hear more about this.. as far as I knew the internal SQLite DB wouldn't become culprit to this sort of problem. Could any core devs comment on this? Cal On Thu, Apr 11, 2013 at 9:01 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Thu, Apr 11, 2013 at 11:44 AM, Steven Ayre wrote: > >> What database are you using? Sqlite? MySQL? PostgreSQL? >> > > The one that came with FreeSWITCH (Sqlite). > > Check if there's anything else connected to the same database and what >> queries are being run. Perhaps something is holding a lock on the voicemail >> table. >> > > FreeSWITCH is the only thing that is connected to the database. I have no > other programs on that machine accessing that database. > > The problem eventually went away after letting it sit for a couple of > minutes. I suspect the DB locked was just a symptom of the real problem > which may have happened earlier. The user couldn't connect to VM for some > reason, so he made multiple tries in rapid succession, which locked the > database somehow. The trouble is that I can't reproduce it. > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130412/be2c552a/attachment.html From cal.leeming at simplicitymedialtd.co.uk Fri Apr 12 17:35:14 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 12 Apr 2013 14:35:14 +0100 Subject: [Freeswitch-users] MPL v2 In-Reply-To: <376804DE-69C4-4A29-B512-2990A12FD106@freeswitch.org> References: <376804DE-69C4-4A29-B512-2990A12FD106@freeswitch.org> Message-ID: http://en.wikipedia.org/wiki/WTFPL Cal On Fri, Apr 12, 2013 at 2:12 PM, Ken Rice wrote: > still not going to happen... it turns the mpl1.1 into the gpl and defeats > the reasoning FreeSWITCH is licensed MPL1.1 in the first place > > Ken > Sent from my iPad > > On Apr 12, 2013, at 2:56, Gabriel Gunderson wrote: > > > On Sun, Jan 8, 2012 at 8:00 PM, Ken Rice wrote: > >> If I had to guess after reading the MPL2.0 tony will most likely not > change > >> to that version of the license. > > > > http://www.mozilla.org/MPL/2.0/Revision-FAQ.html > > > > I don't know, seems like some good stuff in there. > > > > > > > > Best, > > Gabe > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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/20130412/5f0792a8/attachment.html From brian at freeswitch.org Fri Apr 12 17:52:13 2013 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Apr 2013 08:52:13 -0500 Subject: [Freeswitch-users] deflect In-Reply-To: References: Message-ID: <7839869634508110166@unknownmsgid> Doubt it, it sends a refer if the call is answered. And providers do not follow 3xx either! Sent from my iPhone On Apr 11, 2013, at 12:09 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: Is the "deflect" application supposed to work on a gateway? I tried deflecting an incoming call using this: It failed. The default_provider_from_domain variable is sip.flowroute.com. -- Steve _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130412/d4539308/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 6287 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130412/d4539308/attachment-0001.bin From brian at freeswitch.org Fri Apr 12 17:52:39 2013 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Apr 2013 08:52:39 -0500 Subject: [Freeswitch-users] Cisco / Linksys SPA8000 Opinions In-Reply-To: References: Message-ID: <-1763231161002951157@unknownmsgid> Don't use the spa8000 then! Sent from my iPhone On Apr 11, 2013, at 5:53 AM, Mike wrote: > Hi, > > we're looking at getting a Cisco / Linksys SPA8000 for T.38 fax termination (with FreeSWITCH configured for T.38 Passthru). > > I know from past experience that T.38 code can vary wildly in it's quality from gateway to gateway - so does anyone have any experience with / views of this particular one? > > Cheers > > 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 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 6287 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130412/be9011d7/attachment.bin From ml88888 at hotmail.com Fri Apr 12 08:42:07 2013 From: ml88888 at hotmail.com (FSX) Date: Thu, 11 Apr 2013 21:42:07 -0700 (PDT) Subject: [Freeswitch-users] Weird port in "Contact", SIP/2.0 200 OK message Message-ID: <1365741727165-7589685.post@n2.nabble.com> After resolving latest build problems and trying to make a test call to voicemail I see the new problem, that wasn't here in the previous builds. When call is answered by FS, FS sends "SIP/2.0 200 OK message", which now suddenly specified the port #1 instead of its actual port (5080). Here is this part of the message, that FS now sends to SIP client: Contact: As a result, SIP client did not ACK that message and call becomes canceled by FS soon... No audio, of course too. Here is what FS sent in the same configuration using previous (2 weeks ago) build: Contact: where: 5080 is example of FS port, configured for taking connections form SIP clients. Here is yet another difference in console log that I hope may give a clue why is that now. In the latest build FS prints this line: 2013-04-11 21:10:57.079034 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/2005 at domain while with previous build (which was working well) the line was: 2013-04-11 21:15:22.361652 [NOTICE] sofia_glue.c:4281 Pre-Answer sofia/internal/2005 at domain While the latest build offered a lot of configuration changes comparing to previous (and working) version, I can't find anything related to that strange change... Latest build, used for test (not working) (made 130410): fe4dff7 Previous (and working well) build (made 130326): bb3114e -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Weird-port-in-Contact-SIP-2-0-200-OK-message-tp7589685.html Sent from the freeswitch-users mailing list archive at Nabble.com. From andpe at poczta.onet.pl Fri Apr 12 16:07:45 2013 From: andpe at poczta.onet.pl (andpe) Date: Fri, 12 Apr 2013 14:07:45 +0200 Subject: [Freeswitch-users] IVR, bridge, ring back problem Message-ID: <12885609-d5bc06c5b39a91d4cd85b7400883d355@pmq2.m5r2.onet> Welcome. I have a problem with the IVR in FreeSwitch and function bridge. How it works: customer calls IVR, IVR announses customer chooses button 1 number 1 uses function bridge to connect to, say, 1001. during the bridge establishes a connection user (calling party) can not hear the ring signal (silent in handset). when dialed user pick up handset then all works ok (we can talk). Also when dialed user reject the connection I hear busy signal. I have seen in logs Ring Ready not send to calling party. When user calls directly 1001 all works fine. Another question: How to pass music to user when hi waiting for connection when I use bridge function in IVR? I was looking for a solution but I failed to find any. I am new in FreeSwitch. Sory for my english. Regards Andy From paul at cupis.co.uk Fri Apr 12 20:25:26 2013 From: paul at cupis.co.uk (Paul Cupis) Date: Fri, 12 Apr 2013 17:25:26 +0100 Subject: [Freeswitch-users] IVR, bridge, ring back problem In-Reply-To: <12885609-d5bc06c5b39a91d4cd85b7400883d355@pmq2.m5r2.onet> References: <12885609-d5bc06c5b39a91d4cd85b7400883d355@pmq2.m5r2.onet> Message-ID: <20130412162526.GA13139@eagle.cupis.co.uk> On Fri, Apr 12, 2013 at 02:07:45PM +0200, andpe wrote: > I have a problem with the IVR in FreeSwitch and function bridge. > > How it works: > customer calls IVR, IVR announses > customer chooses button 1 > number 1 uses function bridge to connect to, say, 1001. > during the bridge establishes a connection user (calling party) can not hear the ring signal (silent in handset). > when dialed user pick up handset then all works ok (we can talk). Also when dialed user reject the connection I hear busy signal. > I have seen in logs Ring Ready not send to calling party. > > When user calls directly 1001 all works fine. Do something like this before the bridge? Regards, From juanito1982 at gmail.com Fri Apr 12 20:42:56 2013 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Fri, 12 Apr 2013 18:42:56 +0200 Subject: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION Message-ID: Hello, I've got two different PAT2T boxes pointing to my FS box. First one works ok while second what gets INCOMPATIBLE_DESTINATION. Looking at INVITE request I am not be able to get the reason. If you see http://pastebin.com/3emQDfER you will get 2 INVITE, first one works while second no. Do you know why? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130412/01dcb53f/attachment.html From anthony.minessale at gmail.com Fri Apr 12 20:51:39 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 12 Apr 2013 11:51:39 -0500 Subject: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION In-Reply-To: References: Message-ID: m=audio 0 RTP/AVP 18 0 2 4 8 96 97 98 100 101. media port is 0 maybe some param to choose port range is wrong ? On Fri, Apr 12, 2013 at 11:42 AM, Juan Antonio Iba?ez Santorum < juanito1982 at gmail.com> wrote: > Hello, > > I've got two different PAT2T boxes pointing to my FS box. First one > works ok while second what gets INCOMPATIBLE_DESTINATION. Looking at INVITE > request I am not be able to get the reason. If you see > http://pastebin.com/3emQDfER you will get 2 INVITE, first one works while > second no. > > Do you know why? > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130412/e98d6ce9/attachment-0001.html From jnvines at gmail.com Fri Apr 12 20:54:44 2013 From: jnvines at gmail.com (Nick Vines) Date: Fri, 12 Apr 2013 09:54:44 -0700 Subject: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION In-Reply-To: References: Message-ID: <0EB84BEE-B3B3-461F-A741-B3D73AFC0621@gmail.com> Are you trying to send one to a fax (t38?), or can you post a debug log from the call that fails. > console loglevel debug Nick On Apr 12, 2013, at 9:42 AM, Juan Antonio Iba?ez Santorum wrote: > Hello, > > I've got two different PAT2T boxes pointing to my FS box. First one works ok while second what gets INCOMPATIBLE_DESTINATION. Looking at INVITE request I am not be able to get the reason. If you see http://pastebin.com/3emQDfER you will get 2 INVITE, first one works while second no. > > Do you know why? > 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/20130412/d2216ba1/attachment.html From jkr888 at gmail.com Fri Apr 12 22:07:09 2013 From: jkr888 at gmail.com (Johny Kadarisman Kwan) Date: Fri, 12 Apr 2013 14:07:09 -0400 Subject: [Freeswitch-users] Inserting pause on "say" iterated name_spelled? Message-ID: When using "say" function : I just feel like the spelling between that number a bit fast. Is there a way to insert pause between the number spelled? I supposed can change each number sound file to include silent, but looking for a better way. -Jkwan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130412/ba11201f/attachment.html From msc at freeswitch.org Fri Apr 12 22:52:07 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 12 Apr 2013 11:52:07 -0700 Subject: [Freeswitch-users] mod_fifo: agents are not ringing In-Reply-To: References: Message-ID: For some reason your agents are not actually set as members of the queue. Onhook agents will be listed in the section and off-hook agents will be listed in the section. Evidently, whatever you're doing to add the agents is not working. You may want to try explicitly adding them from the fs_cli using the fifo_member command and seeing how that goes. -MC On Thu, Apr 11, 2013 at 9:32 PM, Ashish gautam wrote: > The output is : > > > > waiting_count="0" importance="0" outbound_per_cycle="1" ring_timeout="60" > default_lag="30" outbound_priority="5" outbound_strategy="ringall"> > > > > > > > > > On Thu, Apr 11, 2013 at 9:42 PM, Michael Collins wrote: > >> What's the output of fscli command: fifo list cool_fifo at 10.1.30.229 >> >> -MC >> >> On Thu, Apr 11, 2013 at 12:13 AM, Ashish gautam wrote: >> >>> Hi, >>> >>> Caller is played music on hold but is not connected to any of the agents >>> since no agent is ringing. I have sip users registered to FS ( they are >>> online). I have configured a SIP user as a member for the queue. But, when >>> the caller calls in, the agent doesn't ring. >>> >>> Please throw some light. >>> >>> -Ashish >>> >>> -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130412/ce6de4b9/attachment.html From msc at freeswitch.org Fri Apr 12 23:38:15 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 12 Apr 2013 12:38:15 -0700 Subject: [Freeswitch-users] Cisco / Linksys SPA8000 Opinions In-Reply-To: <-1763231161002951157@unknownmsgid> References: <-1763231161002951157@unknownmsgid> Message-ID: I'm with Brian on this one. I've only seen heartache and pain from people who've used this device. Frankly, I've had much better success just using the little ones like the SPA-112 and the SPA-3102. -MC On Fri, Apr 12, 2013 at 6:52 AM, Brian West wrote: > Don't use the spa8000 then! > > Sent from my iPhone > > On Apr 11, 2013, at 5:53 AM, Mike wrote: > > > Hi, > > > > we're looking at getting a Cisco / Linksys SPA8000 for T.38 fax > termination (with FreeSWITCH configured for T.38 Passthru). > > > > I know from past experience that T.38 code can vary wildly in it's > quality from gateway to gateway - so does anyone have any experience with / > views of this particular one? > > > > Cheers > > > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130412/62eb5313/attachment.html From shaheryarkh at gmail.com Sat Apr 13 01:07:20 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Fri, 12 Apr 2013 23:07:20 +0200 Subject: [Freeswitch-users] Change order of voicemail message retrival In-Reply-To: References: <5159F9E3.9090204@quentustech.com> <515A03F2.3050506@quentustech.com> Message-ID: Guys, the patch is working. I have created another patch which define new optional parameter for vm_fsdb_msg_list command, which define sort order of returned voicemail messages. Possible values as ASC or DESC, any other value will return in Bad Argument error log, while this parameter is not given at all then ASC is assumed for backward compatibility purpose. Thank you. On Tue, Apr 2, 2013 at 3:35 PM, Muhammad Shahzad wrote: > ok, i will do it tonight and update jira. Thanks again for your prompt > help. > > Thank you. > > > On Tue, Apr 2, 2013 at 9:38 AM, Avi Marcus wrote: > >> If you want to speed things up, apply it manually, test it, and post your >> results to Jira. >> >> -Avi >> >> >> On Tue, Apr 2, 2013 at 10:14 AM, Muhammad Shahzad wrote: >> >>> Great. Thanks guys. When can i expect this patch merged in trunk? >>> >>> Thank you. >>> >>> >>> On Tue, Apr 2, 2013 at 12:28 AM, Michael Collins wrote: >>> >>>> Thanks Kristin! that looks like a simple but effective patch. >>>> -MC >>>> >>>> >>>> On Mon, Apr 1, 2013 at 3:02 PM, Kristin King < >>>> kristin.king at quentustech.com> wrote: >>>> >>>>> http://jira.freeswitch.org/browse/FS-5249 >>>>> >>>>> Kristin King >>>>> Quentus Technologies, INC >>>>> 1037 NE 65th St, Ste 273 >>>>> Seattle, WA 98115 >>>>> Main: 877-211-9337 >>>>> Office: 206-388-4778 >>>>> Fax: 206-462-1861 >>>>> Cell: 206-755-7329 >>>>> Email: kristin.king at quentustech.com >>>>> >>>>> >>>>> On 04/01/2013 02:19 PM, Kristin King wrote: >>>>> > There wasn't an option for this, but I just finished coding one this >>>>> > morning. I'm filing the feature request and attaching the patch and >>>>> > hopefully it'll be going in shortly. >>>>> > >>>>> > Kristin King >>>>> > Quentus Technologies, INC >>>>> > 1037 NE 65th St, Ste 273 >>>>> > Seattle, WA 98115 >>>>> > Main: 877-211-9337 >>>>> > Office: 206-388-4778 >>>>> > Fax: 206-462-1861 >>>>> > Cell: 206-755-7329 >>>>> > Email: kristin.king at quentustech.com >>>>> > >>>>> > On 04/01/2013 01:43 PM, Muhammad Shahzad wrote: >>>>> >> Hi, >>>>> >> >>>>> >> Is there any variable / parameter to change the order of messages >>>>> played >>>>> >> back while checking voicemail? Currently its FIFO, i am trying to >>>>> play >>>>> >> them in LIFO mode. >>>>> >> >>>>> >> Thank you. >>>>> >> >>>>> >> >>>>> >> -- >>>>> >> Mit freundlichen Gr??en >>>>> >> Muhammad Shahzad >>>>> >> ----------------------------------- >>>>> >> CISCO Rich Media Communication Specialist (CRMCS) >>>>> >> CISCO Certified Network Associate (CCNA) >>>>> >> Cell: +49 176 99 83 10 85 >>>>> >> MSN: shari_786pk at hotmail.com >>>>> >> Email: shaheryarkh at googlemail.com >>>> shaheryarkh at googlemail.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 >>>>> >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.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 >>>> >>>> >>> >>> >>> -- >>> Mit freundlichen Gr??en >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +49 176 99 83 10 85 >>> MSN: shari_786pk at hotmail.com >>> Email: shaheryarkh at googlemail.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 >> >> > > > -- > Mit freundlichen Gr??en > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130412/1e65c12d/attachment-0001.html From juanito1982 at gmail.com Sat Apr 13 01:46:24 2013 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Fri, 12 Apr 2013 23:46:24 +0200 Subject: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION In-Reply-To: References: Message-ID: That what the problem. Bad RTP range. Regards 2013/4/12 Anthony Minessale > m=audio 0 RTP/AVP 18 0 2 4 8 96 97 98 100 101. > > media port is 0 maybe some param to choose port range is wrong ? > > > > > On Fri, Apr 12, 2013 at 11:42 AM, Juan Antonio Iba?ez Santorum < > juanito1982 at gmail.com> wrote: > >> Hello, >> >> I've got two different PAT2T boxes pointing to my FS box. First one >> works ok while second what gets INCOMPATIBLE_DESTINATION. Looking at INVITE >> request I am not be able to get the reason. If you see >> http://pastebin.com/3emQDfER you will get 2 INVITE, first one works >> while second no. >> >> Do you know why? >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at 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/20130412/f06b31f2/attachment.html From juanito1982 at gmail.com Sat Apr 13 02:00:44 2013 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Sat, 13 Apr 2013 00:00:44 +0200 Subject: [Freeswitch-users] Codec negotiation bug? Message-ID: Hello, I am doing some codec negotiation (early) tests but FS does not work as I expected. I can read at wiki: "When FS calls leg B, the list of codecs in outbound-codec-prefs for the SIP profile is reorganized by pushing the codec negotiated above for leg A at the top. If B does not accept any of the codecs, the calls fails, obviously" Codec prefs setted as: FS select PCMA for A leg FS offers following SDP to B leg: v=0 o=FreeSWITCH 1365781615 1365781616 IN IP4 176.31.117.49 s=FreeSWITCH c=IN IP4 176.31.117.49 t=0 0 m=audio 19780 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 m=audio 19780 RTP/AVP 18 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Session progress from B leg contains: v=0 o=root 3311 3311 IN IP4 91.192.110.234 s=session c=IN IP4 91.192.110.234 t=0 0 m=audio 16194 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv FS codec comparison debug look as: sofia_glue.c:5094 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:30:64000] sofia_glue.c:5094 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] sofia_glue.c:3077 Set Codec sofia/external/0034950004588 G729/8000 20 ms 160 samples 8000 bits FS start comparing G729 instead PCMA selected in A leg Is that ok? is there any way to get selected PCMA for B leg without using late negotiation nor disabling transcoding Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130413/9f88fadc/attachment.html From krice at freeswitch.org Sat Apr 13 02:16:22 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 12 Apr 2013 17:16:22 -0500 Subject: [Freeswitch-users] Codec negotiation bug? In-Reply-To: Message-ID: You are prefing G729, so its going to try it first On 4/12/13 5:00 PM, "Juan Antonio Iba?ez Santorum" wrote: > Hello, > > ? ?I am doing some codec negotiation (early) tests but FS does not work as I > expected. > > I can read at wiki: "When FS calls leg B, the list of codecs in > outbound-codec-prefs for the SIP profile is reorganized by pushing the codec > negotiated above for leg A at the top. If B does not accept any of the codecs, > the calls fails, obviously" > > Codec prefs setted as: > > ? data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/> > ? > > FS select PCMA for A leg > > FS offers following SDP to B leg: > > ?v=0 > ? ?o=FreeSWITCH 1365781615 1365781616 IN IP4 176.31.117.49 > ? ?s=FreeSWITCH > ? ?c=IN IP4 176.31.117.49 > ? ?t=0 0 > ? ?m=audio 19780 RTP/AVP 8 101 13 > ? ?a=rtpmap:101 telephone-event/8000 > ? ?a=fmtp:101 0-16 > ? ?a=ptime:30 > ? ?m=audio 19780 RTP/AVP 18 0 8 3 101 13 > ? ?a=rtpmap:101 telephone-event/8000 > ? ?a=fmtp:101 0-16 > ? ?a=ptime:20 > > Session progress from B leg contains: > > ? ?v=0 > ? ?o=root 3311 3311 IN IP4 91.192.110.234 > ? ?s=session > ? ?c=IN IP4 91.192.110.234 > ? ?t=0 0 > ? ?m=audio 16194 RTP/AVP 18 8 0 101 > ? ?a=rtpmap:18 G729/8000 > ? ?a=fmtp:18 annexb=no > ? ?a=rtpmap:8 PCMA/8000 > ? ?a=rtpmap:0 PCMU/8000 > ? ?a=rtpmap:101 telephone-event/8000 > ? ?a=fmtp:101 0-16 > ? ?a=silenceSupp:off - - - - > ? ?a=ptime:20 > ? ?a=sendrecv > > FS codec comparison debug look as: > > sofia_glue.c:5094 Audio Codec Compare > [G729:18:8000:20:8000]/[PCMA:8:8000:30:64000] > sofia_glue.c:5094 Audio Codec Compare > [G729:18:8000:20:8000]/[G729:18:8000:20:8000] > sofia_glue.c:3077 Set Codec sofia/external/0034950004588 G729/8000 20 ms 160 > samples 8000 bits > > FS start comparing G729 instead PCMA selected in A leg > > Is that ok? is there any way to get selected PCMA for B leg without using late > negotiation nor disabling transcoding > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130412/d82f207c/attachment.html From jleung at v10networks.ca Sat Apr 13 03:53:29 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Fri, 12 Apr 2013 16:53:29 -0700 Subject: [Freeswitch-users] Cisco / Linksys SPA8000 Opinions In-Reply-To: References: <-1763231161002951157@unknownmsgid> Message-ID: <002401ce37d8$ef95f950$cec1ebf0$@v10networks.ca> The SPA3102 is okay for a basic FXO, but it has problems with echo even with the latest firmware because of the design of the hardware. I haven't personally tried the Obi110, but I've heard positive results out of it. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, April 12, 2013 12:38 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco / Linksys SPA8000 Opinions I'm with Brian on this one. I've only seen heartache and pain from people who've used this device. Frankly, I've had much better success just using the little ones like the SPA-112 and the SPA-3102. -MC On Fri, Apr 12, 2013 at 6:52 AM, Brian West wrote: Don't use the spa8000 then! Sent from my iPhone On Apr 11, 2013, at 5:53 AM, Mike wrote: > Hi, > > we're looking at getting a Cisco / Linksys SPA8000 for T.38 fax termination (with FreeSWITCH configured for T.38 Passthru). > > I know from past experience that T.38 code can vary wildly in it's quality from gateway to gateway - so does anyone have any experience with / views of this particular one? > > Cheers > > 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130412/382304eb/attachment-0001.html From ml88888 at hotmail.com Sat Apr 13 07:47:05 2013 From: ml88888 at hotmail.com (FSX) Date: Fri, 12 Apr 2013 20:47:05 -0700 (PDT) Subject: [Freeswitch-users] Weird port in "Contact", SIP/2.0 200 OK message In-Reply-To: <1365741727165-7589685.post@n2.nabble.com> References: <1365741727165-7589685.post@n2.nabble.com> Message-ID: <1365824825113-7589709.post@n2.nabble.com> Today [20130412] I made latest build, using git: 2684cc4 and I see the same story. :( FreeSWITCH advertises its port (port to connect to) as '1'. Here is the part of message it sends to SIP clients: Contact: Of course it breaks further communications... Where it comes from? And, more importantly, how to fix it? Two weeks ago FS's build did not have such problem... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Weird-port-in-Contact-SIP-2-0-200-OK-message-tp7589685p7589709.html Sent from the freeswitch-users mailing list archive at Nabble.com. From chris at gonumina.com Sat Apr 13 08:40:13 2013 From: chris at gonumina.com (Chris Ferreira) Date: Sat, 13 Apr 2013 00:40:13 -0400 Subject: [Freeswitch-users] Cisco / Linksys SPA8000 Opinions In-Reply-To: References: <-1763231161002951157@unknownmsgid> Message-ID: <380018169940894253@unknownmsgid> Michael, would you mind sharing the config you have found most successful for the SPA-112? I have one working, but I think it could be better. Thanks, -Chris ___________________ Mobile Reply On Apr 12, 2013, at 3:45 PM, Michael Collins wrote: I'm with Brian on this one. I've only seen heartache and pain from people who've used this device. Frankly, I've had much better success just using the little ones like the SPA-112 and the SPA-3102. -MC On Fri, Apr 12, 2013 at 6:52 AM, Brian West wrote: > Don't use the spa8000 then! > > Sent from my iPhone > > On Apr 11, 2013, at 5:53 AM, Mike wrote: > > > Hi, > > > > we're looking at getting a Cisco / Linksys SPA8000 for T.38 fax > termination (with FreeSWITCH configured for T.38 Passthru). > > > > I know from past experience that T.38 code can vary wildly in it's > quality from gateway to gateway - so does anyone have any experience with / > views of this particular one? > > > > Cheers > > > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.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/20130413/4ed9eb90/attachment.html From andpe at poczta.onet.pl Sat Apr 13 09:12:58 2013 From: andpe at poczta.onet.pl (Andrzej) Date: Sat, 13 Apr 2013 07:12:58 +0200 Subject: [Freeswitch-users] IVR, bridge, ring back problem In-Reply-To: <20130412162526.GA13139@eagle.cupis.co.uk> References: <12885609-d5bc06c5b39a91d4cd85b7400883d355@pmq2.m5r2.onet> <20130412162526.GA13139@eagle.cupis.co.uk> Message-ID: <5168E95A.3000709@poczta.onet.pl> I'll check on Monday. but: 1. Is this normal behavior IVR? 2. How it should be written: a) or b) look at the colon after "set". Regards, W dniu 12.04.2013 18:25, Paul Cupis pisze: > On Fri, Apr 12, 2013 at 02:07:45PM +0200, andpe wrote: >> I have a problem with the IVR in FreeSwitch and function bridge. >> >> How it works: >> customer calls IVR, IVR announses >> customer chooses button 1 >> number 1 uses function bridge to connect to, say, 1001. >> during the bridge establishes a connection user (calling party) can not hear the ring signal (silent in handset). >> when dialed user pick up handset then all works ok (we can talk). Also when dialed user reject the connection I hear busy signal. >> I have seen in logs Ring Ready not send to calling party. >> >> When user calls directly 1001 all works fine. > Do something like this before the bridge? > > > > > 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 > From gerrylist at drouillard.ca Sat Apr 13 15:32:03 2013 From: gerrylist at drouillard.ca (Gerald Drouillard) Date: Sat, 13 Apr 2013 07:32:03 -0400 Subject: [Freeswitch-users] Cisco / Linksys SPA8000 Opinions In-Reply-To: References: Message-ID: <51694233.8000009@drouillard.ca> On 4/11/2013 6:49 AM, Mike wrote: > Hi, > > we're looking at getting a Cisco / Linksys SPA8000 for T.38 fax > termination (with FreeSWITCH configured for T.38 Passthru). > > I know from past experience that T.38 code can vary wildly in it's > quality from gateway to gateway - so does anyone have any experience > with / views of this particular one? You can try this guide I have created for sipx systems: http://www.drouillard.biz/blog/sipxecs-and-cisco-spa8800-and-spa8000/ We have used Appia and Voice Innovations as voip providers and have had pretty good stats. We also use t38modem with hylafax but find there are a few numbers you just cannot fax to. -- Regards -------------------------------------- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz From cal.leeming at simplicitymedialtd.co.uk Sat Apr 13 19:11:53 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sat, 13 Apr 2013 16:11:53 +0100 Subject: [Freeswitch-users] Weird port in "Contact", SIP/2.0 200 OK message In-Reply-To: <1365824825113-7589709.post@n2.nabble.com> References: <1365741727165-7589685.post@n2.nabble.com> <1365824825113-7589709.post@n2.nabble.com> Message-ID: Lol wtf... Can you enable "sofia global siptrace on" then pastebin us the logs from that please. Specifically I'm wanting to know if the packets are leaving FS in this way, or if they are being mangled by something in-between. Cal On Sat, Apr 13, 2013 at 4:47 AM, FSX wrote: > Today [20130412] I made latest build, using git: 2684cc4 and I see the same > story. :( > FreeSWITCH advertises its port (port to connect to) as '1'. Here is the > part > of message it sends to SIP clients: > Contact: > Of course it breaks further communications... > > Where it comes from? And, more importantly, how to fix it? > > Two weeks ago FS's build did not have such problem... > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Weird-port-in-Contact-SIP-2-0-200-OK-message-tp7589685p7589709.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/20130413/898cb0fd/attachment.html From ml88888 at hotmail.com Sun Apr 14 09:12:25 2013 From: ml88888 at hotmail.com (FSX) Date: Sat, 13 Apr 2013 22:12:25 -0700 (PDT) Subject: [Freeswitch-users] Weird port in "Contact", SIP/2.0 200 OK message In-Reply-To: References: <1365741727165-7589685.post@n2.nabble.com> <1365824825113-7589709.post@n2.nabble.com> Message-ID: <1365916345941-7589714.post@n2.nabble.com> Here is the log: http://pastebin.com/MTjdqbBF Watch for "Contact:" field, that FS sends in messages to its clients, and particularly, port value there... Again, configuration of FS used in this case, is the same as two weeks ago. But build is the newest. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Weird-port-in-Contact-SIP-2-0-200-OK-message-tp7589685p7589714.html Sent from the freeswitch-users mailing list archive at Nabble.com. From cmason at frontiernetworks.ca Sun Apr 14 09:15:05 2013 From: cmason at frontiernetworks.ca (Colin Mason) Date: Sun, 14 Apr 2013 01:15:05 -0400 Subject: [Freeswitch-users] Setting custom presence/BLF In-Reply-To: <1365764589.27767.680.camel@luna.madrid.commsmundi.com> References: <1365764589.27767.680.camel@luna.madrid.commsmundi.com> Message-ID: <0D1C698866F66045A6201FD0F59CAC9001479112A9@EX.frontier.local> Yes this can be done. I used this for a call centre application with Linksys 504G phones. I wrote a python script that is always running and uses ESL to listen for certain events and then changes the color/status of the BLF depending on whether the agent is logged in or out of the mod_callcenter queue. I listen for 2 custom events that are called from the dialplan: blf::green blf::red I also listen for 2 normal events: sofia::register PRESENCE_PROBE When one of the custom BLF events is received I craft a PRESENCE_OUT events which sends a NOTIFY to the phones and changes the BLF color. The reason I have to listen to the register and presence_probe events is because FreeSWITCH will override the BLF if I don't. When I receive one of these events I use mod_callcenter API commands to check if the agent is logged in or out and change the color of the BLF accordingly with a PRESENCE_OUT (notify) event. Hope this helps. Colin -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fran?ois Sent: Friday, April 12, 2013 7:03 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Setting custom presence/BLF Hello, Is it possible to modify the presence/BLF status of a specific directory user, or of a virtual one? Can it be done with ESL + custom presence event? Is there an API to check/set presence status of a user (to use with an empty "virtual" user just for BLF light)? The purpose would be to control BLF lights on the phones to show custom status like: - show presence of endpoints connected to another FS - light up if at least one of several phones (department) is in a call - light up if coffee is ready - ... Thanks, Fran?ois. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From drk at drkngs.net Sun Apr 14 11:08:46 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Sun, 14 Apr 2013 00:08:46 -0700 Subject: [Freeswitch-users] Setting custom presence/BLF In-Reply-To: <0D1C698866F66045A6201FD0F59CAC9001479112A9@EX.frontier.local> Message-ID: <20130414070846.178bafcb@mail.tritonwest.net> This is great info, but what I think what he was asking is how to make the PRESENCE_OUT. It's one of the things I have not been able to figure out from the wiki... Maybe you could give an example? --Dave _____ From: Colin Mason [mailto:cmason at frontiernetworks.ca] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Sat, 13 Apr 2013 22:15:05 -0700 Subject: Re: [Freeswitch-users] Setting custom presence/BLF Yes this can be done. I used this for a call centre application with Linksys 504G phones. I wrote a python script that is always running and uses ESL to listen for certain events and then changes the color/status of the BLF depending on whether the agent is logged in or out of the mod_callcenter queue. I listen for 2 custom events that are called from the dialplan: blf::green blf::red I also listen for 2 normal events: sofia::register PRESENCE_PROBE When one of the custom BLF events is received I craft a PRESENCE_OUT events which sends a NOTIFY to the phones and changes the BLF color. The reason I have to listen to the register and presence_probe events is because FreeSWITCH will override the BLF if I don't. When I receive one of these events I use mod_callcenter API commands to check if the agent is logged in or out and change the color of the BLF accordingly with a PRESENCE_OUT (notify) event. Hope this helps. Colin -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fran?ois Sent: Friday, April 12, 2013 7:03 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Setting custom presence/BLF Hello, Is it possible to modify the presence/BLF status of a specific directory user, or of a virtual one? Can it be done with ESL + custom presence event? Is there an API to check/set presence status of a user (to use with an empty "virtual" user just for BLF light)? The purpose would be to control BLF lights on the phones to show custom status like: - show presence of endpoints connected to another FS - light up if at least one of several phones (department) is in a call - light up if coffee is ready - ... Thanks, Fran?ois. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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/20130414/fae549ef/attachment.html From jmesquita at freeswitch.org Sun Apr 14 18:58:53 2013 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Sun, 14 Apr 2013 11:58:53 -0300 Subject: [Freeswitch-users] Setting custom presence/BLF In-Reply-To: <20130414070846.178bafcb@mail.tritonwest.net> References: <20130414070846.178bafcb@mail.tritonwest.net> Message-ID: Or maybe even share the script as a recipe? Sent from my iPhone On Apr 14, 2013, at 4:08 AM, "Dave R. Kompel" wrote: > This is great info, but what I think what he was asking is how to make the PRESENCE_OUT. It's one of the things I have not been able to figure out from the wiki... Maybe you could give an example? > > --Dave > > From: Colin Mason [mailto:cmason at frontiernetworks.ca] > To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > Sent: Sat, 13 Apr 2013 22:15:05 -0700 > Subject: Re: [Freeswitch-users] Setting custom presence/BLF > > Yes this can be done. I used this for a call centre application with Linksys 504G phones. I wrote a python script that is always running and uses ESL to listen for certain events and then changes the color/status of the BLF depending on whether the agent is logged in or out of the mod_callcenter queue. > > I listen for 2 custom events that are called from the dialplan: > > > blf::green > blf::red > > I also listen for 2 normal events: > > sofia::register > PRESENCE_PROBE > > When one of the custom BLF events is received I craft a PRESENCE_OUT events which sends a NOTIFY to the phones and changes the BLF color. > > The reason I have to listen to the register and presence_probe events is because FreeSWITCH will override the BLF if I don't. When I receive one of these events I use mod_callcenter API commands to check if the agent is logged in or out and change the color of the BLF accordingly with a PRESENCE_OUT (notify) event. > > Hope this helps. > > Colin > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fran?ois > Sent: Friday, April 12, 2013 7:03 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Setting custom presence/BLF > > Hello, > > Is it possible to modify the presence/BLF status of a specific directory user, or of a virtual one? > > Can it be done with ESL + custom presence event? Is there an API to check/set presence status of a user (to use with an empty "virtual" user just for BLF light)? > > The purpose would be to control BLF lights on the phones to show custom status like: > - show presence of endpoints connected to another FS > - light up if at least one of several phones (department) is in a call > - light up if coffee is ready > - ... > > Thanks, > Fran?ois. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20130414/f0d8d641/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Sun Apr 14 20:28:18 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 14 Apr 2013 17:28:18 +0100 Subject: [Freeswitch-users] Weird port in "Contact", SIP/2.0 200 OK message In-Reply-To: <1365916345941-7589714.post@n2.nabble.com> References: <1365741727165-7589685.post@n2.nabble.com> <1365824825113-7589709.post@n2.nabble.com> <1365916345941-7589714.post@n2.nabble.com> Message-ID: Can you enable debug logging and pastebin this as well? I can't think of anything off the top of my head as to why this would be happening, so hoping the debug logs shed some more info. Cal On Sun, Apr 14, 2013 at 6:12 AM, FSX wrote: > Here is the log: http://pastebin.com/MTjdqbBF > Watch for "Contact:" field, that FS sends in messages to its clients, and > particularly, port value there... > > Again, configuration of FS used in this case, is the same as two weeks ago. > But build is the newest. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Weird-port-in-Contact-SIP-2-0-200-OK-message-tp7589685p7589714.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/20130414/f7fe3e0c/attachment.html From drk at drkngs.net Sun Apr 14 20:59:44 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Sun, 14 Apr 2013 09:59:44 -0700 Subject: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION In-Reply-To: Message-ID: <20130414165944.b016e8c3@mail.tritonwest.net> Sorry for the late reply but I just remembered something about older linksys stuff: There is a bug that under some conditions causes the FLASH to get corupt and the device starts doing all sorts of shit you can't figure out, to fix do the following: 1) Factory reset (**** get ivr prompt enter 73738# (spells reset)) 2) Reload firmware the fast way, hit it w/ upgrade url (http://devip/admin/upgrade?http://pathtotobinfile) 3) Factory reset again 4) re-configure. This should make the weirdness go away. The device can also get to this state if it had its firmware upgraded at some point, and it was not factory reset, both before and after the upgrade. --Dave _____ From: Anthony Minessale [mailto:anthony.minessale at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Fri, 12 Apr 2013 09:51:39 -0700 Subject: Re: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION m=audio 0 RTP/AVP 18 0 2 4 8 96 97 98 100 101. media port is 0 maybe some param to choose port range is wrong ? On Fri, Apr 12, 2013 at 11:42 AM, Juan Antonio Iba?ez Santorum wrote: Hello, I've got two different PAT2T boxes pointing to my FS box. First one works ok while second what gets INCOMPATIBLE_DESTINATION. Looking at INVITE request I am not be able to get the reason. If you see http://pastebin.com/3emQDfER you will get 2 INVITE, first one works while second no. Do you know why? 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130414/3904ec3a/attachment.html From drk at drkngs.net Sun Apr 14 21:16:47 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Sun, 14 Apr 2013 10:16:47 -0700 Subject: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION In-Reply-To: <20130414165944.b016e8c3@mail.tritonwest.net> Message-ID: <20130414171647.c9786ad9@mail.tritonwest.net> Woops I forget to mention if you don't have the firmware handy on some http server, you can always use my copy of the latest. I've had to fix/upgrade them many times in the past 6 years since the final version of the firmware was released. http://yourdeviceip/admin/upgrade?http://64.62.214.101/d0wndl/pap2t-5-1-6.bin --Dave _____ From: Dave R. Kompel [mailto:drk at drkngs.net] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Sun, 14 Apr 2013 09:59:44 -0700 Subject: Re: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION Sorry for the late reply but I just remembered something about older linksys stuff: There is a bug that under some conditions causes the FLASH to get corupt and the device starts doing all sorts of shit you can't figure out, to fix do the following: 1) Factory reset (**** get ivr prompt enter 73738# (spells reset)) 2) Reload firmware the fast way, hit it w/ upgrade url (http://devip/admin/upgrade?http://pathtotobinfile) 3) Factory reset again 4) re-configure. This should make the weirdness go away. The device can also get to this state if it had its firmware upgraded at some point, and it was not factory reset, both before and after the upgrade. --Dave _____ From: Anthony Minessale [mailto:anthony.minessale at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Fri, 12 Apr 2013 09:51:39 -0700 Subject: Re: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION m=audio 0 RTP/AVP 18 0 2 4 8 96 97 98 100 101. media port is 0 maybe some param to choose port range is wrong ? On Fri, Apr 12, 2013 at 11:42 AM, Juan Antonio Iba?ez Santorum wrote: Hello, I've got two different PAT2T boxes pointing to my FS box. First one works ok while second what gets INCOMPATIBLE_DESTINATION. Looking at INVITE request I am not be able to get the reason. If you see http://pastebin.com/3emQDfER you will get 2 INVITE, first one works while second no. Do you know why? 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130414/59b5fd5d/attachment.html From engineerzuhairraza at gmail.com Sun Apr 14 22:38:34 2013 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Sun, 14 Apr 2013 22:38:34 +0400 Subject: [Freeswitch-users] GSMopen - huawei E1550 can't read messages Message-ID: Hi, I have a huawei e1550, it can't receive messages though sending is fine event plain message shows nothing when I send a message to the sim Here are the logs after the SIM inbox got full http://pastebin.com/kjUYHGNA Regards, Zohair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130414/a24826ea/attachment-0001.html From drk at drkngs.net Sun Apr 14 23:11:14 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Sun, 14 Apr 2013 12:11:14 -0700 Subject: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION In-Reply-To: <20130414171647.c9786ad9@mail.tritonwest.net> Message-ID: <20130414191114.106d9dc9@mail.tritonwest.net> Never mind, ya'all fuckers on the list and you know who you are, starting DDOSing this host at over 5,000,000 spoofed Syn's so the host is gone... Blackholed, and is NEVER coming back. So for all you assholes reading this list: GO FUCK YOURSELF. Everyone else, thank you and have a nice day :) --Dave _____ From: Dave R. Kompel [mailto:drk at drkngs.net] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Sun, 14 Apr 2013 10:16:47 -0700 Subject: Re: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION Woops I forget to mention if you don't have the firmware handy on some http server, you can always use my copy of the latest. I've had to fix/upgrade them many times in the past 6 years since the final version of the firmware was released. http://yourdeviceip/admin/upgrade?http://64.62.214.101/d0wndl/pap2t-5-1-6.bin --Dave _____ From: Dave R. Kompel [mailto:drk at drkngs.net] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Sun, 14 Apr 2013 09:59:44 -0700 Subject: Re: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION Sorry for the late reply but I just remembered something about older linksys stuff: There is a bug that under some conditions causes the FLASH to get corupt and the device starts doing all sorts of shit you can't figure out, to fix do the following: 1) Factory reset (**** get ivr prompt enter 73738# (spells reset)) 2) Reload firmware the fast way, hit it w/ upgrade url (http://devip/admin/upgrade?http://pathtotobinfile) 3) Factory reset again 4) re-configure. This should make the weirdness go away. The device can also get to this state if it had its firmware upgraded at some point, and it was not factory reset, both before and after the upgrade. --Dave _____ From: Anthony Minessale [mailto:anthony.minessale at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Fri, 12 Apr 2013 09:51:39 -0700 Subject: Re: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION m=audio 0 RTP/AVP 18 0 2 4 8 96 97 98 100 101. media port is 0 maybe some param to choose port range is wrong ? On Fri, Apr 12, 2013 at 11:42 AM, Juan Antonio Iba?ez Santorum wrote: Hello, I've got two different PAT2T boxes pointing to my FS box. First one works ok while second what gets INCOMPATIBLE_DESTINATION. Looking at INVITE request I am not be able to get the reason. If you see http://pastebin.com/3emQDfER you will get 2 INVITE, first one works while second no. Do you know why? 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at 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/20130414/1419c23a/attachment.html From sherifomran2000 at yahoo.com Sun Apr 14 23:13:56 2013 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 14 Apr 2013 12:13:56 -0700 (PDT) Subject: [Freeswitch-users] call listening and talking In-Reply-To: <20130414171647.c9786ad9@mail.tritonwest.net> Message-ID: <1365966836.24216.YahooMailClassic@web141205.mail.bf1.yahoo.com> hello all, i am tryng to listen and talk to a currently connected A-B call. I tried the following: ./fs_cli show calls? (to know the uuid) originator 1xxx at sip.server.com &eavesdrop this rings the 1xxx phone and then disconnects any idea regards, S.O -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130414/bda377a5/attachment.html From jmesquita at freeswitch.org Sun Apr 14 23:51:27 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 14 Apr 2013 16:51:27 -0300 Subject: [Freeswitch-users] call listening and talking In-Reply-To: <1365966836.24216.YahooMailClassic@web141205.mail.bf1.yahoo.com> References: <20130414171647.c9786ad9@mail.tritonwest.net> <1365966836.24216.YahooMailClassic@web141205.mail.bf1.yahoo.com> Message-ID: That's a conference, not an eavesdrop ain't it? Jo?o Mesquita On Sun, Apr 14, 2013 at 4:13 PM, Sherif Omran wrote: > hello all, > > i am tryng to listen and talk to a currently connected A-B call. > > I tried the following: > > ./fs_cli > show calls (to know the uuid) > originator 1xxx at sip.server.com &eavesdrop > > this rings the 1xxx phone and then disconnects > > any idea > regards, > > S.O > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130414/a3b7dd0d/attachment.html From jeff at askcornerstone.net Sun Apr 14 13:36:42 2013 From: jeff at askcornerstone.net (Jeff Bernhardt) Date: Sun, 14 Apr 2013 09:36:42 +0000 Subject: [Freeswitch-users] Newbie question about inbound calls/NAT (yes I checked wiki first:-) ) Message-ID: <80DFCBDE2AC6574487E3826FAF38F9CC387A508D@vega.terisol.com> Setting up my first test box with Freeswitch. I'm really excited about getting it up and running. I'm trying to get inbound calls working from external using a Vitelity trunk. Let's say my username from Vitelity is "abcd_efg" and password is "password," external IP is 1.2.3.4 and internal is 192.168.10.32. I get registered to Vitelity fine (their web panel shows I'm registered and fs_cli confirms). However, when I make a call inbound, I get: [WARNING] sofia_reg.c:2621 Can't find user [abcd_1.2.3.4 at 192.168.10.32] from You must define a domain called '192.168.10.32' in your directory and add a user with the id="abcd_1.2.3.4" attribute and you must configure your device to use the proper domain in it's authentication credentials. I don't know if this is related, but to get around NAT, I have this in vars.xml: and this in both internal.xml and external.xml: I have a feeling I've misunderstood something fundamental about setting up Freeswitch for inbound access. I've put in the above configs as per what I've found in the wiki, but I'm not sure how to do what the warning says or if it really means I have something not setup right. What am I doing wrong? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130414/726dfc47/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Apr 15 05:38:22 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 15 Apr 2013 02:38:22 +0100 Subject: [Freeswitch-users] Newbie question about inbound calls/NAT (yes I checked wiki first:-) ) In-Reply-To: <80DFCBDE2AC6574487E3826FAF38F9CC387A508D@vega.terisol.com> References: <80DFCBDE2AC6574487E3826FAF38F9CC387A508D@vega.terisol.com> Message-ID: Hello Jeff, Here is a related thread; http://freeswitch-users.2379917.n2.nabble.com/You-must-define-a-domain-called-xx-com-in-your-directory-td7585944.html http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006232.html You need to configure your domain and username, but as you can see here your gateway/users are sending the domain '192.168.10.32', but you have not configured this domain in sofia.conf. If you are using multi-tenant then you'll need to put some more thought into how you're going to approach this. If you are not using multi-tenant (i.e. one domain only) then you could force the domain as discussed here; http://www.feedingedge.co.uk/blog/2010/08/03/techie-post-opensim-and-freeswitch-problems/ Hope this helps Cal On Sun, Apr 14, 2013 at 10:36 AM, Jeff Bernhardt wrote: > Setting up my first test box with Freeswitch. I'm really excited about > getting it up and running. > > I'm trying to get inbound calls working from external using a Vitelity > trunk. Let's say my username from Vitelity is "abcd_efg" and password is > "password," external IP is 1.2.3.4 and internal is 192.168.10.32. I get > registered to Vitelity fine (their web panel shows I'm registered and > fs_cli confirms). However, when I make a call inbound, I get: > > [WARNING] sofia_reg.c:2621 Can't find user [abcd_1.2.3.4 at 192.168.10.32] > from > You must define a domain called '192.168.10.32' in your directory and add > a user with the id="abcd_1.2.3.4" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > > I don't know if this is related, but to get around NAT, I have this in > vars.xml: > > > > and this in both internal.xml and external.xml: > > > > I have a feeling I've misunderstood something fundamental about setting > up Freeswitch for inbound access. I've put in the above configs as per what > I've found in the wiki, but I'm not sure how to do what the warning says or > if it really means I have something not setup right. What am I doing wrong? > > 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/20130415/e8ab1072/attachment-0001.html From ashish at nms.co.in Mon Apr 15 09:42:00 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 15 Apr 2013 11:12:00 +0530 Subject: [Freeswitch-users] mod_fifo: agents are not ringing In-Reply-To: References: Message-ID: Thanks Michael, Probably the originate dialstring I am using is not correct. Can you please tell what is the proper format for that? On Sat, Apr 13, 2013 at 12:22 AM, Michael Collins wrote: > For some reason your agents are not actually set as members of the queue. > Onhook agents will be listed in the section and off-hook agents > will be listed in the section. Evidently, whatever you're doing > to add the agents is not working. You may want to try explicitly adding > them from the fs_cli using the fifo_member command and seeing how that goes. > > -MC > > > On Thu, Apr 11, 2013 at 9:32 PM, Ashish gautam wrote: > >> The output is : >> >> >> >> > waiting_count="0" importance="0" outbound_per_cycle="1" ring_timeout="60" >> default_lag="30" outbound_priority="5" outbound_strategy="ringall"> >> >> >> >> >> >> >> >> >> On Thu, Apr 11, 2013 at 9:42 PM, Michael Collins wrote: >> >>> What's the output of fscli command: fifo list cool_fifo at 10.1.30.229 >>> >>> -MC >>> >>> On Thu, Apr 11, 2013 at 12:13 AM, Ashish gautam wrote: >>> >>>> Hi, >>>> >>>> Caller is played music on hold but is not connected to any of the >>>> agents since no agent is ringing. I have sip users registered to FS ( they >>>> are online). I have configured a SIP user as a member for the queue. But, >>>> when the caller calls in, the agent doesn't ring. >>>> >>>> Please throw some light. >>>> >>>> -Ashish >>>> >>>> > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.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/20130415/9a008704/attachment.html From ashish at nms.co.in Mon Apr 15 10:29:32 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 15 Apr 2013 11:59:32 +0530 Subject: [Freeswitch-users] PRI channels suspended Message-ID: Hi, I am having another problem over here. When I run "ftdm dump 1 5" from fs_cli I get my channel status as SUSPENDED. This happens for channel number 5 to 30. Also when I try to make outgoing calls more than 4 in number, four calls originate properly whereas for the rest it shows NORMAL_CIRCUIT_CONGESTION. Here is the output: span_id: 1 chan_id: 5 physical_span_id: 1 physical_chan_id: 5 physical_status: ok physical_status_red: 0 physical_status_yellow: 0 physical_status_rai: 0 physical_status_blue: 0 physical_status_ais: 0 physical_status_general: 0 signaling_status: UP type: B state: SUSPENDED last_state: RESTART txgain: 0.00 rxgain: 0.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NORMAL_UNSPECIFIED session: (none) -- States -- -- Function -- -- Location -- -- Time Offset -- DOWN => RESTART [on_dchan_up] [ftmod_libpri.c:2372] 0ms RESTART => SUSPENDED [on_timeout_t316] [ftmod_libpri.c:1952] 120008ms Time since last state change: 324790932ms Please help me getting out of this. Thanks. --Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/8cf7950a/attachment.html From navnath.sonavne at yahoo.com Mon Apr 15 09:50:09 2013 From: navnath.sonavne at yahoo.com (Navnath Sonavne) Date: Mon, 15 Apr 2013 13:50:09 +0800 (SGT) Subject: [Freeswitch-users] Communication between sip soft phones(X-Lite) via freeswitch In-Reply-To: <1365775400.83057.YahooMailNeo@web192205.mail.sg3.yahoo.com> References: <1365775400.83057.YahooMailNeo@web192205.mail.sg3.yahoo.com> Message-ID: <1366005009.45805.YahooMailNeo@web192206.mail.sg3.yahoo.com> Hi All, I have successfully?compiled?and installed freeswitch on my Linux CentOS. I have registered two softphones of X-Lite in freeswitch with extension numbers are 1100 ,1101. I have installed these softphones on two?different?machines of windows OS. I have created two XML dial plans for above extension?numbers?in usr/local/freeswitch/conf/directory/default. Now when i call 1101 from 1100 extension it did not?establish?a call. I know something is going wrong but what is it? I am not able to figure out. I request you all please help me to establish a call between 1100 and 1101. FYI i am reading Freeswitch 1.0.6 by Anthony Minessale ,?Michael S. Collins , Darren Schreiber?. I am attaching ?1100.xml and 1101.xml files. Regards, Navnath. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/9f7db2a3/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: 1101.xml Type: text/xml Size: 736 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/9f7db2a3/attachment.xml -------------- next part -------------- A non-text attachment was scrubbed... Name: 1100.xml Type: text/xml Size: 736 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/9f7db2a3/attachment-0001.xml From ashish at nms.co.in Mon Apr 15 10:48:59 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 15 Apr 2013 12:18:59 +0530 Subject: [Freeswitch-users] Communication between sip soft phones(X-Lite) via freeswitch In-Reply-To: <1366005009.45805.YahooMailNeo@web192206.mail.sg3.yahoo.com> References: <1365775400.83057.YahooMailNeo@web192205.mail.sg3.yahoo.com> <1366005009.45805.YahooMailNeo@web192206.mail.sg3.yahoo.com> Message-ID: Probably your users might not be in the default context. To make call from one sip user to another both must be in same context. For testing purpose you may want to keep both of them in default context. In the user_name.xml file put context = default , this may solve your problem. On Mon, Apr 15, 2013 at 11:20 AM, Navnath Sonavne wrote: > Hi All, > > I have successfully compiled and installed freeswitch on my Linux CentOS. > I have registered two softphones of X-Lite in freeswitch with extension > numbers are 1100 ,1101. > I have installed these softphones on two different machines of windows OS. > I have created two XML dial plans for above extension numbers in > usr/local/freeswitch/conf/directory/default. > > Now when i call 1101 from 1100 extension it did not establish a call. > > I know something is going wrong but what is it? I am not able to figure > out. > > I request you all please help me to establish a call between 1100 and 1101. > > FYI i am reading Freeswitch 1.0.6 by Anthony Minessale , Michael S. > Collins , Darren Schreiber . > > I am attaching 1100.xml and 1101.xml files. > > > Regards, > Navnath. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/9cdf056a/attachment-0001.html From ashish at nms.co.in Mon Apr 15 11:31:06 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 15 Apr 2013 13:01:06 +0530 Subject: [Freeswitch-users] mod_fifo: agents are not ringing In-Reply-To: References: Message-ID: Hi Michael, When I add members to the fifo queue via fscli it works properly and as expected, but when I add members through the configuration file, it does not work. Here is the configuration file: user/1004 at default user/1003 at default When I add these two members through fscli it works as expected. On Mon, Apr 15, 2013 at 11:12 AM, Ashish gautam wrote: > Thanks Michael, > > Probably the originate dialstring I am using is not correct. Can you > please tell what is the proper format for that? > > On Sat, Apr 13, 2013 at 12:22 AM, Michael Collins wrote: > >> For some reason your agents are not actually set as members of the queue. >> Onhook agents will be listed in the section and off-hook agents >> will be listed in the section. Evidently, whatever you're doing >> to add the agents is not working. You may want to try explicitly adding >> them from the fs_cli using the fifo_member command and seeing how that goes. >> >> -MC >> >> >> On Thu, Apr 11, 2013 at 9:32 PM, Ashish gautam wrote: >> >>> The output is : >>> >>> >>> >>> >> caller_count="0" waiting_count="0" importance="0" outbound_per_cycle="1" >>> ring_timeout="60" default_lag="30" outbound_priority="5" >>> outbound_strategy="ringall"> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Thu, Apr 11, 2013 at 9:42 PM, Michael Collins wrote: >>> >>>> What's the output of fscli command: fifo list cool_fifo at 10.1.30.229 >>>> >>>> -MC >>>> >>>> On Thu, Apr 11, 2013 at 12:13 AM, Ashish gautam wrote: >>>> >>>>> Hi, >>>>> >>>>> Caller is played music on hold but is not connected to any of the >>>>> agents since no agent is ringing. I have sip users registered to FS ( they >>>>> are online). I have configured a SIP user as a member for the queue. But, >>>>> when the caller calls in, the agent doesn't ring. >>>>> >>>>> Please throw some light. >>>>> >>>>> -Ashish >>>>> >>>>> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.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/20130415/f0de085f/attachment.html From jeff at askcornerstone.net Mon Apr 15 11:30:41 2013 From: jeff at askcornerstone.net (Jeff Bernhardt) Date: Mon, 15 Apr 2013 07:30:41 +0000 Subject: [Freeswitch-users] Newbie question about inbound calls/NAT (yes I checked wiki first:-) ) In-Reply-To: References: <80DFCBDE2AC6574487E3826FAF38F9CC387A508D@vega.terisol.com>, Message-ID: <80DFCBDE2AC6574487E3826FAF38F9CC387A5507@vega.terisol.com> Thanks. I had a look at the links and changed "domain and even "domain_name" in vars.xml to be the external ip, but the error now just says it can's find user abcd_1.2.3.4 at 1.2.3.4 instead of @192.168.10.32. And actually, I 'm mostly confused because I thought incoming calls were treated as unauthenticated and allowed to pass through. Isn't registering the trunk with the SIP provider enough to get incoming calls accepted? The errors says "...you must configure your device to use the proper domain in it's authentication credentials," but how is the SIP trunk treated as a device? Like I said, I think I'm misunderstanding something fundamental about this. Thanks you. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Cal Leeming [Simplicity Media Ltd] [cal.leeming at simplicitymedialtd.co.uk] Sent: Sunday, April 14, 2013 3:38 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Newbie question about inbound calls/NAT (yes I checked wiki first:-) ) Hello Jeff, Here is a related thread; http://freeswitch-users.2379917.n2.nabble.com/You-must-define-a-domain-called-xx-com-in-your-directory-td7585944.html http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006232.html You need to configure your domain and username, but as you can see here your gateway/users are sending the domain '192.168.10.32', but you have not configured this domain in sofia.conf. If you are using multi-tenant then you'll need to put some more thought into how you're going to approach this. If you are not using multi-tenant (i.e. one domain only) then you could force the domain as discussed here; http://www.feedingedge.co.uk/blog/2010/08/03/techie-post-opensim-and-freeswitch-problems/ Hope this helps Cal On Sun, Apr 14, 2013 at 10:36 AM, Jeff Bernhardt > wrote: Setting up my first test box with Freeswitch. I'm really excited about getting it up and running. I'm trying to get inbound calls working from external using a Vitelity trunk. Let's say my username from Vitelity is "abcd_efg" and password is "password," external IP is 1.2.3.4 and internal is 192.168.10.32. I get registered to Vitelity fine (their web panel shows I'm registered and fs_cli confirms). However, when I make a call inbound, I get: [WARNING] sofia_reg.c:2621 Can't find user [abcd_1.2.3.4 at 192.168.10.32] from You must define a domain called '192.168.10.32' in your directory and add a user with the id="abcd_1.2.3.4" attribute and you must configure your device to use the proper domain in it's authentication credentials. I don't know if this is related, but to get around NAT, I have this in vars.xml: and this in both internal.xml and external.xml: I have a feeling I've misunderstood something fundamental about setting up Freeswitch for inbound access. I've put in the above configs as per what I've found in the wiki, but I'm not sure how to do what the warning says or if it really means I have something not setup right. What am I doing wrong? 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/20130415/d441bd0b/attachment.html From stephen.thwaites at callstera.com Mon Apr 15 15:56:59 2013 From: stephen.thwaites at callstera.com (Stephen Thwaites) Date: Mon, 15 Apr 2013 13:56:59 +0200 Subject: [Freeswitch-users] default timeout In-Reply-To: References: Message-ID: Hello, Just to close this down. It seems that most of the telco companies in the Netherlands cancel the origination after 60s. This is the issue and nothing to do with FS. When working within the 60s the call_timeout parameter works perfectly. Regards, Steve. On Wed, Apr 10, 2013 at 6:03 PM, Stephen Thwaites wrote: > Hello all, > I still haven't managed to solve this one. Any further tips would be > greatly appreciated. > > I feel that I am experimenting with various parameters without a full > understanding of what the behavior should be. The A leg (caller) hits > the dialplan in public, the public dialplan finds a match and > transfers to the appropriate dialplan in corresponding context, this > extension then uses 'set' or 'export' on the call_timeout variable > and attempts a bridge. Does this 'set' set a timeout for the A leg or > B leg? If I use 'export' does it set the timeout for both legs? Last > of all does setting or exporting the call_timeout override the default > bridge timeout of 60s? > > Many thanks. > > Regards, > Steve. > > On Fri, Mar 29, 2013 at 4:50 PM, Stephen Thwaites > wrote: >> Hello, >> I have setup this scenario now with a test case and have tried using >> the 'export' application instead of 'set' for the call_timeout=25. >> Sadly the same behavior. Similarly no change when using >> hangup_after_bridge=true and continue_on_fail=true before each bridge >> as suggested. Further I have tried using originate_timeout=25 and also >> experimented with leg_timeouts. I then wondered if the voip provider >> set a timeout on the incoming call and they do, but this is set to >> 120s. I all tests I get an ORIGINATOR_CANCEL after 60s. >> >> I then did a sip trace and log level 6 from the fs_cli. Here is the (I >> think) the relevant bit that shows that FS is sending a CANCEL to the >> telephones. >> >> Any further help would be appreciated! >> >> Regards, >> Steve. >> >> 2013-03-29 15:41:55.117929 [NOTICE] sofia.c:6379 Hangup >> sofia/external/0610884128 at 91.195.160.3 [CS_EXECUTE] >> [ORIGINATOR_CANCEL] >> 2013-03-29 15:41:55.117929 [NOTICE] switch_ivr_originate.c:3349 Hangup >> sofia/internal/sip:1000 at 192.168.1.113:45060 [CS_CONSUME_MEDIA] >> [ORIGINATOR_CANCEL] >> 2013-03-29 15:41:55.117929 [NOTICE] switch_ivr_originate.c:3349 Hangup >> sofia/internal/sip:1001 at 192.168.1.112:35160 [CS_CONSUME_MEDIA] >> [ORIGINATOR_CANCEL] >> 2013-03-29 15:41:55.117929 [NOTICE] switch_ivr_originate.c:3349 Hangup >> sofia/internal/sip:1002 at 192.168.1.10:49535 [CS_CONSUME_MEDIA] >> [ORIGINATOR_CANCEL] >> 2013-03-29 15:41:55.117929 [INFO] mod_dptools.c:3052 Originate Failed. >> Cause: ORIGINATOR_CANCEL >> 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1505 Session >> 9911 (sofia/external/0610884128 at 91.195.160.3) Ended >> 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1509 Close >> Channel sofia/external/0610884128 at 91.195.160.3 [CS_DESTROY] >> 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1505 Session >> 9917 (sofia/internal/sip:1000 at 192.168.1.113:45060) Ended >> 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1509 Close >> Channel sofia/internal/sip:1000 at 192.168.1.113:45060 [CS_DESTROY] >> send 429 bytes to udp/[80.101.42.120]:52292 at 14:43:27.409611: >> ------------------------------------------------------------------------ >> CANCEL sip:1000 at 192.168.1.113:45060 SIP/2.0 >> Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK5H6X2DKX8Dj1N >> Route: >> Max-Forwards: 67 >> From: "Thwaites, Stephen" >> ;tag=1rv4jNU086aaQ >> To: >> Call-ID: d6322816-1321-1231-80a7-fa163e468f82 >> CSeq: 41948035 CANCEL >> Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1505 Session >> 9918 (sofia/internal/sip:1001 at 192.168.1.112:35160) Ended >> 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1509 Close >> Channel sofia/internal/sip:1001 at 192.168.1.112:35160 [CS_DESTROY] >> send 429 bytes to udp/[80.101.42.120]:42328 at 14:43:27.410670: >> ------------------------------------------------------------------------ >> CANCEL sip:1001 at 192.168.1.112:35160 SIP/2.0 >> Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK6tZp48305p8KH >> Route: >> Max-Forwards: 67 >> From: "Thwaites, Stephen" >> ;tag=21NXmgc45F1vj >> To: >> Call-ID: d6335070-1321-1231-80a7-fa163e468f82 >> CSeq: 41948035 CANCEL >> Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2013-03-29 15:41:55.127924 [NOTICE] switch_core_session.c:1505 Session >> 9919 (sofia/internal/sip:1002 at 192.168.1.10:49535) Ended >> 2013-03-29 15:41:55.127924 [NOTICE] switch_core_session.c:1509 Close >> Channel sofia/internal/sip:1002 at 192.168.1.10:49535 [CS_DESTROY] >> send 427 bytes to udp/[80.101.42.120]:50175 at 14:43:27.413100: >> ------------------------------------------------------------------------ >> CANCEL sip:1002 at 192.168.1.10:49535 SIP/2.0 >> Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK73rF63m42Zy6c >> Route: >> Max-Forwards: 67 >> From: "Thwaites, Stephen" >> ;tag=3aFppBX72rQFe >> To: >> Call-ID: d634a26a-1321-1231-80a7-fa163e468f82 >> CSeq: 41948035 CANCEL >> Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> recv 531 bytes from udp/[80.101.42.120]:50175 at 14:43:27.440064: >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 94.100.113.237;rport=5060;branch=z9hG4bK73rF63m42Zy6c >> From: "Thwaites, Stephen" >> ;tag=3aFppBX72rQFe >> To: ;tag=942234751 >> Call-ID: d634a26a-1321-1231-80a7-fa163e468f82 >> CSeq: 41948035 CANCEL >> Contact: >> Supported: replaces, path, timer, eventlist >> User-Agent: Grandstream GXP2200 1.0.1.40 >> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, >> REFER, UPDATE, MESSAGE >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> recv 507 bytes from udp/[80.101.42.120]:50175 at 14:43:27.445183: >> ------------------------------------------------------------------------ >> SIP/2.0 487 Request Terminated >> Via: SIP/2.0/UDP 94.100.113.237;rport=5060;branch=z9hG4bK73rF63m42Zy6c >> From: "Thwaites, Stephen" >> ;tag=3aFppBX72rQFe >> To: ;tag=942234751 >> Call-ID: d634a26a-1321-1231-80a7-fa163e468f82 >> CSeq: 41948035 INVITE >> Supported: replaces, path, timer, eventlist >> User-Agent: Grandstream GXP2200 1.0.1.40 >> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, >> REFER, UPDATE, MESSAGE >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> send 387 bytes to udp/[80.101.42.120]:50175 at 14:43:27.445316: >> ------------------------------------------------------------------------ >> ACK sip:1002 at 192.168.1.10:49535 SIP/2.0 >> Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK73rF63m42Zy6c >> Route: >> Max-Forwards: 67 >> From: "Thwaites, Stephen" >> ;tag=3aFppBX72rQFe >> To: ;tag=942234751 >> Call-ID: d634a26a-1321-1231-80a7-fa163e468f82 >> CSeq: 41948035 ACK >> Content-Length: 0 >> >> etc... >> >> On Tue, Mar 26, 2013 at 12:22 AM, Stephen Thwaites >> wrote: >>> Nick, Michael, >>> Thanks for the advise, will give these a try and feed back to the list. >>> >>> Regards, >>> Steve. >>> >>> On Mon, Mar 25, 2013 at 11:21 PM, Michael Collins wrote: >>>> Try "export" instead of "set" on your call_timeout=25 lines. >>>> -MC >>>> >>>> On Fri, Mar 22, 2013 at 8:32 AM, Stephen Thwaites >>>> wrote: >>>>> >>>>> Hi All, >>>>> Apologies for the simple question but I can't find the answer anywhere >>>>> in the books, wiki, or our friend google. >>>>> >>>>> If somebody calls us, FS does an ORIGINATE_CANCEL after 60s but the >>>>> follow-me scheme we have configured is100s. How can I increase the >>>>> default call_timeout of 60s to 100s? Or maybe I am just doing >>>>> something wrong! >>>>> >>>>> I have tried leg_timeouts, originate_timeouts both on the transfer and >>>>> the bridge as well to no avail? >>>>> >>>>> Would be very grateful for any help or advise. >>>>> >>>>> Regards, >>>>> Steve >>>>> >>>>> Some Details: >>>>> - External call comes in on external profile from our voip provider. >>>>> - Dialplan in the public context does a transfer to a follow-me >>>>> extension 7777 in context creche-babys >>>> data="7777 XML creche-babys "/> >>>>> - The 7777 extension is as follows and the call hangs up part way >>>>> through the third step if nobody picks up (after 60s total). >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>>> data="{ignore_early_media=true}user/21@${domain_name},user/20@${domain_name}"/> >>>>> >>>>> >>>>> >>>> data="{ignore_early_media=true}user/22@${domain_name}"/> >>>>> >>>>> >>>>> >>>> >>>>> data={ignore_early_media=true}user/23@${domain_name},user/24@${domain_name},user/26@${domain_name},user/27@${domain_name},user/28@${domain_name}"/> >>>>> >>>>> >>>>> >>>>> >>>> data="{ignore_early_media=true}sofia/gateway/3120 >>>>> ...voipprovidergateway/06 ...mobile number"/> >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.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 navnath.sonavne at yahoo.com Mon Apr 15 13:45:50 2013 From: navnath.sonavne at yahoo.com (Navnath Sonavne) Date: Mon, 15 Apr 2013 17:45:50 +0800 (SGT) Subject: [Freeswitch-users] Communication between sip soft phones(X-Lite) via freeswitch Message-ID: <1366019150.24224.YahooMailNeo@web192201.mail.sg3.yahoo.com> Hi, Your response :? Probably your users might not be in the default context. To make call from one sip user to another both must be in same context. For testing purpose you may want to keep both of them in default context. In the user_name.xml file put context = default , this may solve your problem. My both extensions are in default context .To crosscheck it i used list_users command at fs_cli console and it shown that both phones are in default context.Here is output of list_users? 1100|default|192.168.8.41|default|sofia/internal/sip:1100 at 192.168.9.165:29072;rinstance=ee8af9a648b2e70d||Vivekanandm|1100 1101|default|192.168.8.41|default|sofia/internal/sip:1101 at 192.168.8.103:26196;rinstance=a6ab3905d800b8f7||navnath|1101 As you?suggested put context = default in user_name.xml but i did not find?user_name.xml?file in my freeswitch directory. Can you please tell me where to locate this file? Regards, Navnath. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/0d4ae62e/attachment.html From mike at jerris.com Mon Apr 15 16:25:46 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 15 Apr 2013 08:25:46 -0400 Subject: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION In-Reply-To: <20130414191114.106d9dc9@mail.tritonwest.net> References: <20130414191114.106d9dc9@mail.tritonwest.net> Message-ID: <95D75B00-1E89-4CA4-9FDA-366CB1A3999C@jerris.com> 1. DDosing people trying to help someone out is just lame. If you are doing this, please go away. 2. This kind of language on the mailing list is not acceptable. Anyone doing this may be moderated or banned from the list. Mike On Apr 14, 2013, at 3:11 PM, Dave R. Kompel wrote: > Never mind, ya'all fuckers on the list and you know who you are, starting DDOSing this host at over 5,000,000 spoofed Syn's so the host is gone... Blackholed, and is NEVER coming back. > > So for all you assholes reading this list: GO FUCK YOURSELF. > > Everyone else, thank you and have a nice day :) > > --Dave > > From: Dave R. Kompel [mailto:drk at drkngs.net] > To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > Sent: Sun, 14 Apr 2013 10:16:47 -0700 > Subject: Re: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION > > Woops I forget to mention if you don't have the firmware handy on some http server, you can always use my copy of the latest. I've had to fix/upgrade them many times in the past 6 years since the final version of the firmware was released. > > http://yourdeviceip/admin/upgrade?http://64.62.214.101/d0wndl/pap2t-5-1-6.bin > > --Dave > > From: Dave R. Kompel [mailto:drk at drkngs.net] > To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > Sent: Sun, 14 Apr 2013 09:59:44 -0700 > Subject: Re: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION > > Sorry for the late reply but I just remembered something about older linksys stuff: > > There is a bug that under some conditions causes the FLASH to get corupt and the device starts doing all sorts of shit you can't figure out, to fix do the following: > > 1) Factory reset (**** get ivr prompt enter 73738# (spells reset)) > 2) Reload firmware the fast way, hit it w/ upgrade url (http://devip/admin/upgrade?http://pathtotobinfile) > 3) Factory reset again > 4) re-configure. > > This should make the weirdness go away. > > The device can also get to this state if it had its firmware upgraded at some point, and it was not factory reset, both before and after the upgrade. > > --Dave > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > Sent: Fri, 12 Apr 2013 09:51:39 -0700 > Subject: Re: [Freeswitch-users] PAP2T INCOMPATIBLE_DESTINATION > > m=audio 0 RTP/AVP 18 0 2 4 8 96 97 98 100 101. > media port is 0 maybe some param to choose port range is wrong ? > > > > On Fri, Apr 12, 2013 at 11:42 AM, Juan Antonio Iba?ez Santorum wrote: > Hello, > > I've got two different PAT2T boxes pointing to my FS box. First one works ok while second what gets INCOMPATIBLE_DESTINATION. Looking at INVITE request I am not be able to get the reason. If you see http://pastebin.com/3emQDfER you will get 2 INVITE, first one works while second no. > > Do you know why? > Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/55315bda/attachment.html From mahesh.katta at flexydial.com Mon Apr 15 16:26:34 2013 From: mahesh.katta at flexydial.com (mahesh katta) Date: Mon, 15 Apr 2013 17:56:34 +0530 Subject: [Freeswitch-users] SQL ERR database is locked Message-ID: Hi List, """SQL ERR[database is Locked]""" When this come calls are stop. why this is come ? and how can I solve this problem , this problem occur periodically. currently I am using Freeswich-1.3XXX . Kindly suggest me how can I fix this problem. Thanking you. Best Regards, Mahesh Katta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/2efdf702/attachment.html From m.hubert at hexanet.fr Mon Apr 15 16:47:00 2013 From: m.hubert at hexanet.fr (Mickael Hubert) Date: Mon, 15 Apr 2013 14:47:00 +0200 Subject: [Freeswitch-users] the user part of PAI uri Message-ID: Hi list, I would like to retrieve the user part of PAI's uri. In fact, when I have : PAI: P-Asserted-Identity: <*tel*:+333........ at freeswitch1>, my variable_sip_P-Asserted-Identity is: +333333*@freeswitch* But with: PAI: P-Asserted-Identity: <*sip*:+333........ at freeswitch1>, my variable_sip_P-Asserted-Identity is: +333333 How to write only user part of PAI in my variable_sip_P-Asserted-Identity on each of the two cases ? thanks -- Cordialement Hubert Micka?l Ing?nieur VOIP - Hexanet -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/490cb4c2/attachment.html From nneul at mst.edu Mon Apr 15 17:47:47 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Mon, 15 Apr 2013 08:47:47 -0500 Subject: [Freeswitch-users] auto-stack sizing on linux appears to break voicemail email notification Message-ID: <516C0503.1040005@mst.edu> It seems to cause the child processes to inherit the small stack size, which results in sendmail segfaulting. Haven't dug through this yet. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From msc at freeswitch.org Mon Apr 15 20:18:30 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 15 Apr 2013 09:18:30 -0700 Subject: [Freeswitch-users] mod_fifo: agents are not ringing In-Reply-To: References: Message-ID: It's just a simple sofia dialstring. Here's an example: http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback -MC On Sun, Apr 14, 2013 at 10:42 PM, Ashish gautam wrote: > Thanks Michael, > > Probably the originate dialstring I am using is not correct. Can you > please tell what is the proper format for that? > > On Sat, Apr 13, 2013 at 12:22 AM, Michael Collins wrote: > >> For some reason your agents are not actually set as members of the queue. >> Onhook agents will be listed in the section and off-hook agents >> will be listed in the section. Evidently, whatever you're doing >> to add the agents is not working. You may want to try explicitly adding >> them from the fs_cli using the fifo_member command and seeing how that goes. >> >> -MC >> >> >> On Thu, Apr 11, 2013 at 9:32 PM, Ashish gautam wrote: >> >>> The output is : >>> >>> >>> >>> >> caller_count="0" waiting_count="0" importance="0" outbound_per_cycle="1" >>> ring_timeout="60" default_lag="30" outbound_priority="5" >>> outbound_strategy="ringall"> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Thu, Apr 11, 2013 at 9:42 PM, Michael Collins wrote: >>> >>>> What's the output of fscli command: fifo list cool_fifo at 10.1.30.229 >>>> >>>> -MC >>>> >>>> On Thu, Apr 11, 2013 at 12:13 AM, Ashish gautam wrote: >>>> >>>>> Hi, >>>>> >>>>> Caller is played music on hold but is not connected to any of the >>>>> agents since no agent is ringing. I have sip users registered to FS ( they >>>>> are online). I have configured a SIP user as a member for the queue. But, >>>>> when the caller calls in, the agent doesn't ring. >>>>> >>>>> Please throw some light. >>>>> >>>>> -Ashish >>>>> >>>>> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/e37d62a4/attachment.html From marketing at cluecon.com Mon Apr 15 21:15:06 2013 From: marketing at cluecon.com (Michael Collins) Date: Mon, 15 Apr 2013 10:15:06 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Greetings! Happy tax day to those in the USA - we hope all is well with your business. Speaking of business, I thought I would relay the interesting newsabout Dish Network making a bid for Sprint. Many here in North America will be keeping a close eye on this one. Whether or not this is just a big messor an opportunity for Sprint to become a "real" competitor to AT&T and Verizon remains to be seen. Regardless of the outcome, most of us here are hoping for a healthier Sprint so that we can avoid another duopoly. On last week's conference callwe decided to have a preliminary discussion so that we can prepare for this week . Dave Kompel will be showing us how to build rapidly FreeSWITCH applications in MS Visual Studio 2012 and have those run under mod_managed. Be sure to consult this documentso that you can get all the prerequisites installed in time for our call on Wednesday. In other news I would like to let everyone know that I spoke with Kashif Kahn over at Vestec . We are gearing up for the automatic speech recognition application building contest. The winners will be announced at ClueCon 2013 . The official contest page will be posted on the ClueCon website shortly. Stay tuned for more information and be ready to start building your ASR applications! The ClueCon 2013 call for speakers recently went out and we've had a number of submissions already. We look forwarding to hearing more talk ideas, so please send those in right away. In the meantime ClueCon registrationis now open so be sure to get signed up, and don't forget to book your room at the Hyatt Chicago Magnificent Mile hotel for only $169 per night. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/7b761ca4/attachment.html From ira at connectmevoice.com Mon Apr 15 22:12:43 2013 From: ira at connectmevoice.com (Ira Tessler) Date: Mon, 15 Apr 2013 14:12:43 -0400 Subject: [Freeswitch-users] mod_callcenter with Active/Backup Freeswitch servers using a shared mysql database Message-ID: Configuration: Server: FS1 - Active server. All calls come into this server. All phones register to this server. Server: FS2 - Backup Server. Is online just in in case FS1 fails. If FS1 fails, all calls get routed to FS2 (by my SBC) Both FS1 & FS2 share a mysql database for core, all sofia profiles and mod_call center. The problem I am having is with mod_callcenter. When a call comes into FS1 and FS2 Freeswitch process is down, all works as expected when I call the call center application. When FS2 is online as well as FS1 it looks like FS2 is processing data in the members table(I'm guessing) and I get this log entries: 2013-04-15 13:32:40.225033 [DEBUG] mod_callcenter.c:1045 Updated Agent agent701 at 22555.sbc.cmvtesttele.com set state = Receiving 2013-04-15 13:32:40.225033 [ERR] mod_callcenter.c:1399 Member CONNECTME <+17324909007> with uuid 71b3beea-a5f2-11e2-9143-7dbd0c2f58e6 in queue mySbcQueue1 at 22555.sbc.cmvtesttele.com is gone just before we assigned an agent 2013-04-15 13:32:40.225033 [DEBUG] mod_callcenter.c:1045 Updated Agent agent701 at 22555.sbc.cmvtesttele.com set state = Waiting Since FS1 is the "live" server, I don't want FS2 to process call center calls. I want FS1 and FS2 to share the call center data in order have one copy of the agent and tier data for redundancy and so I don't have to worry about syncing multiple call center database. Before I post a Jira, I wanted to see if I am doing some wrong, or the other question does mod_callcenter have the capability or is it designed to work in this configuration? If not, anyhow have any alternative solutions? Thanks! Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/d8fe9dd0/attachment.html From steveayre at gmail.com Tue Apr 16 02:20:57 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 15 Apr 2013 23:20:57 +0100 Subject: [Freeswitch-users] mod_callcenter with Active/Backup Freeswitch servers using a shared mysql database In-Reply-To: References: Message-ID: <35617A0A-B4C3-46D6-8553-6C1CDCF84C3E@gmail.com> There will be background stuff being processed by fs2... Which probably won't be limited to callcenter, it may affect registrations limit etc too if the hostname is the same and perhaps other stuff even if they're not. Are you perhaps able to have FreeSWITCH not running on fs2 and only start it during failover? Solutions like heartbeat/pacemaker could give you that ability. Steve On 15 Apr 2013, at 19:12, Ira Tessler wrote: > Configuration: > Server: FS1 - Active server. All calls come into this server. All phones register to this server. > Server: FS2 - Backup Server. Is online just in in case FS1 fails. If FS1 fails, all calls get routed to FS2 (by my SBC) > > Both FS1 & FS2 share a mysql database for core, all sofia profiles and mod_call center. > > The problem I am having is with mod_callcenter. When a call comes into FS1 and FS2 Freeswitch process is down, all works as expected when I call the call center application. > > When FS2 is online as well as FS1 it looks like FS2 is processing data in the members table(I'm guessing) and I get this log entries: > > 2013-04-15 13:32:40.225033 [DEBUG] mod_callcenter.c:1045 Updated Agent agent701 at 22555.sbc.cmvtesttele.com set state = Receiving > 2013-04-15 13:32:40.225033 [ERR] mod_callcenter.c:1399 Member CONNECTME <+17324909007> with uuid 71b3beea-a5f2-11e2-9143-7dbd0c2f58e6 in queue mySbcQueue1 at 22555.sbc.cmvtesttele.com is gone just before we assigned an agent > 2013-04-15 13:32:40.225033 [DEBUG] mod_callcenter.c:1045 Updated Agent agent701 at 22555.sbc.cmvtesttele.com set state = Waiting > > Since FS1 is the "live" server, I don't want FS2 to process call center calls. I want FS1 and FS2 to share the call center data in order have one copy of the agent and tier data for redundancy and so I don't have to worry about syncing multiple call center database. > > Before I post a Jira, I wanted to see if I am doing some wrong, or the other question does mod_callcenter have the capability or is it designed to work in this configuration? If not, anyhow have any alternative solutions? > > Thanks! > > Ira Tessler > Lead Software Engineer > ConnectMe > (732) 490-9007 x2 > ira at connectmevoice.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/20130415/6127d1f5/attachment-0001.html From steveayre at gmail.com Tue Apr 16 02:23:15 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 15 Apr 2013 23:23:15 +0100 Subject: [Freeswitch-users] mod_callcenter with Active/Backup Freeswitch servers using a shared mysql database In-Reply-To: References: Message-ID: The heartbeat/pacemaker solution would also allow you to have 3rd virtual floating IP assigned to whichever server is active. That would be the ip used by FS rather than the local one. That would mean the SBC wouldn't need to handle any failure detection / rerouting at all. Steve On 15 Apr 2013, at 19:12, Ira Tessler wrote: > Configuration: > Server: FS1 - Active server. All calls come into this server. All phones register to this server. > Server: FS2 - Backup Server. Is online just in in case FS1 fails. If FS1 fails, all calls get routed to FS2 (by my SBC) > > Both FS1 & FS2 share a mysql database for core, all sofia profiles and mod_call center. > > The problem I am having is with mod_callcenter. When a call comes into FS1 and FS2 Freeswitch process is down, all works as expected when I call the call center application. > > When FS2 is online as well as FS1 it looks like FS2 is processing data in the members table(I'm guessing) and I get this log entries: > > 2013-04-15 13:32:40.225033 [DEBUG] mod_callcenter.c:1045 Updated Agent agent701 at 22555.sbc.cmvtesttele.com set state = Receiving > 2013-04-15 13:32:40.225033 [ERR] mod_callcenter.c:1399 Member CONNECTME <+17324909007> with uuid 71b3beea-a5f2-11e2-9143-7dbd0c2f58e6 in queue mySbcQueue1 at 22555.sbc.cmvtesttele.com is gone just before we assigned an agent > 2013-04-15 13:32:40.225033 [DEBUG] mod_callcenter.c:1045 Updated Agent agent701 at 22555.sbc.cmvtesttele.com set state = Waiting > > Since FS1 is the "live" server, I don't want FS2 to process call center calls. I want FS1 and FS2 to share the call center data in order have one copy of the agent and tier data for redundancy and so I don't have to worry about syncing multiple call center database. > > Before I post a Jira, I wanted to see if I am doing some wrong, or the other question does mod_callcenter have the capability or is it designed to work in this configuration? If not, anyhow have any alternative solutions? > > Thanks! > > Ira Tessler > Lead Software Engineer > ConnectMe > (732) 490-9007 x2 > ira at connectmevoice.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/20130415/a4a0cad7/attachment.html From msc at freeswitch.org Tue Apr 16 02:36:53 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 15 Apr 2013 15:36:53 -0700 Subject: [Freeswitch-users] Communication between sip soft phones(X-Lite) via freeswitch In-Reply-To: <1366019150.24224.YahooMailNeo@web192201.mail.sg3.yahoo.com> References: <1366019150.24224.YahooMailNeo@web192201.mail.sg3.yahoo.com> Message-ID: In a plain install it would be in /usr/local/freeswitch/conf/directory/default/ Look for files like 1000.xml, 1001.xml, etc. -MC On Mon, Apr 15, 2013 at 2:45 AM, Navnath Sonavne wrote: > Hi, > > Your response : > Probably your users might not be in the default context. To make call from > one sip user to another both must be in same context. For testing purpose > you may want to keep both of them in default context. In the user_name.xml > file put context = default , this may solve your problem. > > > > My both extensions are in default context .To crosscheck it i used > list_users command at fs_cli console and it shown > that both phones are in default context.Here is output of list_users > > 1100|default|192.168.8.41|default|sofia/internal/sip:1100 at 192.168.9.165:29072 > ;rinstance=ee8af9a648b2e70d||Vivekanandm|1100 > > 1101|default|192.168.8.41|default|sofia/internal/sip:1101 at 192.168.8.103:26196 > ;rinstance=a6ab3905d800b8f7||navnath|1101 > > As you suggested put context = default in user_name.xml > but i did not find user_name.xml file in my freeswitch directory. > Can you please tell me where to locate this file? > > > > Regards, > Navnath. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/4b520725/attachment.html From msc at freeswitch.org Tue Apr 16 09:15:43 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 15 Apr 2013 22:15:43 -0700 Subject: [Freeswitch-users] Newbie question about inbound calls/NAT (yes I checked wiki first:-) ) In-Reply-To: <80DFCBDE2AC6574487E3826FAF38F9CC387A5507@vega.terisol.com> References: <80DFCBDE2AC6574487E3826FAF38F9CC387A508D@vega.terisol.com> <80DFCBDE2AC6574487E3826FAF38F9CC387A5507@vega.terisol.com> Message-ID: For inbound calls you can skip the authentication by adding the source IP address to the "domains" section of conf/autoload_configs/acl.conf.xml. This will prevent FreeSWITCH from challenging the inbound call if it comes from that IP address. This is more desirable than disabling call authentication altogether in your SIP profile. (i.e. you are allowing only a specific IP address to come in unauth'd as opposed to letting the whole world in!) Find out the source IP address for your Vitelity SIP traffic and add it to the acl.conf.xml file in the "domains" section. It will be something like this: After you save that go to fs_cli and type "reloadacl". NOTE: simply doing a "reloadxml" is not sufficient to make the new ACL entry take effect. ("reloadxml" simply reloads the XML config; "reloadacl" does a reloadxml and then reloads the ACL entries.) Hope this helps. -MC On Mon, Apr 15, 2013 at 12:30 AM, Jeff Bernhardt wrote: > Thanks. I had a look at the links and changed "domain and even > "domain_name" in vars.xml to be the external ip, but the error now just > says it can's find user abcd_1.2.3.4 at 1.2.3.4 instead of @192.168.10.32. > > And actually, I 'm mostly confused because I thought incoming calls were > treated as unauthenticated and allowed to pass through. Isn't registering > the trunk with the SIP provider enough to get incoming calls accepted? > > The errors says "...you must configure your device to use the proper > domain in it's authentication credentials," but how is the SIP trunk > treated as a device? > > Like I said, I think I'm misunderstanding something fundamental about > this. > > Thanks you. > > ------------------------------ > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] on behalf of Cal Leeming > [Simplicity Media Ltd] [cal.leeming at simplicitymedialtd.co.uk] > *Sent:* Sunday, April 14, 2013 3:38 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Newbie question about inbound calls/NAT > (yes I checked wiki first:-) ) > > Hello Jeff, > > Here is a related thread; > > http://freeswitch-users.2379917.n2.nabble.com/You-must-define-a-domain-called-xx-com-in-your-directory-td7585944.html > > http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006232.html > > You need to configure your domain and username, but as you can see here > your gateway/users are sending the domain '192.168.10.32', but you have not > configured this domain in sofia.conf. > > If you are using multi-tenant then you'll need to put some more thought > into how you're going to approach this. > > If you are not using multi-tenant (i.e. one domain only) then you could > force the domain as discussed here; > > http://www.feedingedge.co.uk/blog/2010/08/03/techie-post-opensim-and-freeswitch-problems/ > > Hope this helps > > Cal > > On Sun, Apr 14, 2013 at 10:36 AM, Jeff Bernhardt wrote: > >> Setting up my first test box with Freeswitch. I'm really excited about >> getting it up and running. >> >> I'm trying to get inbound calls working from external using a Vitelity >> trunk. Let's say my username from Vitelity is "abcd_efg" and password is >> "password," external IP is 1.2.3.4 and internal is 192.168.10.32. I get >> registered to Vitelity fine (their web panel shows I'm registered and >> fs_cli confirms). However, when I make a call inbound, I get: >> >> [WARNING] sofia_reg.c:2621 Can't find user [abcd_1.2.3.4 at 192.168.10.32] >> from >> You must define a domain called '192.168.10.32' in your directory and add >> a user with the id="abcd_1.2.3.4" attribute >> and you must configure your device to use the proper domain in it's >> authentication credentials. >> >> I don't know if this is related, but to get around NAT, I have this in >> vars.xml: >> >> >> >> and this in both internal.xml and external.xml: >> >> >> >> I have a feeling I've misunderstood something fundamental about setting >> up Freeswitch for inbound access. I've put in the above configs as per what >> I've found in the wiki, but I'm not sure how to do what the warning says or >> if it really means I have something not setup right. What am I doing wrong? >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/96baf94e/attachment-0001.html From msc at freeswitch.org Tue Apr 16 09:24:39 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 15 Apr 2013 22:24:39 -0700 Subject: [Freeswitch-users] Communication between sip soft phones(X-Lite) via freeswitch In-Reply-To: <1366005009.45805.YahooMailNeo@web192206.mail.sg3.yahoo.com> References: <1365775400.83057.YahooMailNeo@web192205.mail.sg3.yahoo.com> <1366005009.45805.YahooMailNeo@web192206.mail.sg3.yahoo.com> Message-ID: Hi Navnath, Thanks for reading our book! :) Comments inline. -Michael Collins On Sun, Apr 14, 2013 at 10:50 PM, Navnath Sonavne wrote: > Hi All, > > I have successfully compiled and installed freeswitch on my Linux CentOS. > I have registered two softphones of X-Lite in freeswitch with extension > numbers are 1100 ,1101. > I have installed these softphones on two different machines of windows OS. > All of the above is perfectly fine. > I have created two XML dial plans for above extension numbers in > usr/local/freeswitch/conf/directory/default. > This is where your problem lies. Technically these files that you created are not "XML dialplans" - they are XML directory entries. In effect, these two files tell FreeSWITCH that you have a user id of "1100" and another user id of "1101". Assuming that your two X-Lite clients each register properly then you have one other step to do in order to make all this work. You need a dialplan entry to handle the new users. On page 67 of the book you'll notice that there are two steps mentioned when adding a user. You have done the first step but probably not the second step. Look on page 68 where it says "Next we need to edit the Dialplan entry for Local_Extension..." I think you'll find the explanation there sufficient. If not, you know where to reach us. :) > > Now when i call 1101 from 1100 extension it did not establish a call. > > I know something is going wrong but what is it? I am not able to figure > out. > > I request you all please help me to establish a call between 1100 and 1101. > > FYI i am reading Freeswitch 1.0.6 by Anthony Minessale , Michael S. > Collins , Darren Schreiber . > > I am attaching 1100.xml and 1101.xml files. > > > Regards, > Navnath. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/3850161b/attachment.html From msc at freeswitch.org Tue Apr 16 09:26:46 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 15 Apr 2013 22:26:46 -0700 Subject: [Freeswitch-users] call listening and talking In-Reply-To: <1365966836.24216.YahooMailClassic@web141205.mail.bf1.yahoo.com> References: <20130414171647.c9786ad9@mail.tritonwest.net> <1365966836.24216.YahooMailClassic@web141205.mail.bf1.yahoo.com> Message-ID: Pastebin the debug output of your originate command. I suspect that it will tell you what's happening. -MC On Sun, Apr 14, 2013 at 12:13 PM, Sherif Omran wrote: > hello all, > > i am tryng to listen and talk to a currently connected A-B call. > > I tried the following: > > ./fs_cli > show calls (to know the uuid) > originator 1xxx at sip.server.com &eavesdrop > > this rings the 1xxx phone and then disconnects > > any idea > regards, > > S.O > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/e6945ac5/attachment.html From msc at freeswitch.org Tue Apr 16 09:29:46 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 15 Apr 2013 22:29:46 -0700 Subject: [Freeswitch-users] IVR, bridge, ring back problem In-Reply-To: <5168E95A.3000709@poczta.onet.pl> References: <12885609-d5bc06c5b39a91d4cd85b7400883d355@pmq2.m5r2.onet> <20130412162526.GA13139@eagle.cupis.co.uk> <5168E95A.3000709@poczta.onet.pl> Message-ID: I believe it's the space and not the colon. However, I don't personally recommend that you put everything into the IVR like that. Instead, try putting all those set apps and the bridge app into a separate XML dialplan extension and then use the transfer app to send the caller there. See the demo IVR in conf/ivr_menus/ for examples on how to do that. -MC On Fri, Apr 12, 2013 at 10:12 PM, Andrzej wrote: > I'll check on Monday. > but: > 1. Is this normal behavior IVR? > 2. How it should be written: > a) > > > > or > b) > > param="set:instant_ringback=true"/> > > > look at the colon after "set". > > Regards, > > W dniu 12.04.2013 18:25, Paul Cupis pisze: > > On Fri, Apr 12, 2013 at 02:07:45PM +0200, andpe wrote: > >> I have a problem with the IVR in FreeSwitch and function bridge. > >> > >> How it works: > >> customer calls IVR, IVR announses > >> customer chooses button 1 > >> number 1 uses function bridge to connect to, say, 1001. > >> during the bridge establishes a connection user (calling party) can not > hear the ring signal (silent in handset). > >> when dialed user pick up handset then all works ok (we can talk). Also > when dialed user reject the connection I hear busy signal. > >> I have seen in logs Ring Ready not send to calling party. > >> > >> When user calls directly 1001 all works fine. > > Do something like this before the bridge? > > > > > > > > > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/7c6f6e0d/attachment-0001.html From msc at freeswitch.org Tue Apr 16 09:33:02 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 15 Apr 2013 22:33:02 -0700 Subject: [Freeswitch-users] Codec negotiation bug? In-Reply-To: References: Message-ID: On Fri, Apr 12, 2013 at 3:16 PM, Ken Rice wrote: > You are prefing G729, so its going to try it first > In other words, you have G729 listed in your codec prefs ahead of PCMA. If you changed it to something like this it would probably work: -MC > > > > On 4/12/13 5:00 PM, "Juan Antonio Iba?ez Santorum" > wrote: > > Hello, > > I am doing some codec negotiation (early) tests but FS does not work as > I expected. > > I can read at wiki: "When FS calls leg B, the list of codecs in > outbound-codec-prefs for the SIP profile is reorganized by pushing the > codec negotiated above for leg A at the top. If B does not accept any of > the codecs, the calls fails, obviously > " > > > Codec prefs setted as: > > > > > FS select PCMA for A leg > > FS offers following SDP to B leg: > > v=0 > o=FreeSWITCH 1365781615 1365781616 IN IP4 176.31.117.49 > s=FreeSWITCH > c=IN IP4 176.31.117.49 > t=0 0 > m=audio 19780 RTP/AVP 8 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:30 > m=audio 19780 RTP/AVP 18 0 8 3 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > Session progress from B leg contains: > > v=0 > o=root 3311 3311 IN IP4 91.192.110.234 > s=session > c=IN IP4 91.192.110.234 > t=0 0 > m=audio 16194 RTP/AVP 18 8 0 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > FS codec comparison debug look as: > > sofia_glue.c:5094 Audio Codec Compare > [G729:18:8000:20:8000]/[PCMA:8:8000:30:64000] > sofia_glue.c:5094 Audio Codec Compare > [G729:18:8000:20:8000]/[G729:18:8000:20:8000] > sofia_glue.c:3077 Set Codec sofia/external/0034950004588 G729/8000 20 ms > 160 samples 8000 bits > > FS start comparing G729 instead PCMA selected in A leg > > Is that ok? is there any way to get selected PCMA for B leg without using > late negotiation nor disabling transcoding > > 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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130415/9434644a/attachment.html From ashish at nms.co.in Tue Apr 16 09:46:39 2013 From: ashish at nms.co.in (Ashish gautam) Date: Tue, 16 Apr 2013 11:16:39 +0530 Subject: [Freeswitch-users] mod_fifo: agents are not ringing In-Reply-To: References: Message-ID: Michael, My fifo.conf.xml is: user/1004 at 10.1.30.229 user/1003 at 10.1.30.229 Its according to the example shown on the wiki page. Still fifo list shows members not added. -- Ashish On Mon, Apr 15, 2013 at 9:48 PM, Michael Collins wrote: > It's just a simple sofia dialstring. Here's an example: > http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback > > -MC > > > On Sun, Apr 14, 2013 at 10:42 PM, Ashish gautam wrote: > >> Thanks Michael, >> >> Probably the originate dialstring I am using is not correct. Can you >> please tell what is the proper format for that? >> >> On Sat, Apr 13, 2013 at 12:22 AM, Michael Collins wrote: >> >>> For some reason your agents are not actually set as members of the >>> queue. Onhook agents will be listed in the section and off-hook >>> agents will be listed in the section. Evidently, whatever >>> you're doing to add the agents is not working. You may want to try >>> explicitly adding them from the fs_cli using the fifo_member command and >>> seeing how that goes. >>> >>> -MC >>> >>> >>> On Thu, Apr 11, 2013 at 9:32 PM, Ashish gautam wrote: >>> >>>> The output is : >>>> >>>> >>>> >>>> >>> caller_count="0" waiting_count="0" importance="0" outbound_per_cycle="1" >>>> ring_timeout="60" default_lag="30" outbound_priority="5" >>>> outbound_strategy="ringall"> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> On Thu, Apr 11, 2013 at 9:42 PM, Michael Collins wrote: >>>> >>>>> What's the output of fscli command: fifo list cool_fifo at 10.1.30.229 >>>>> >>>>> -MC >>>>> >>>>> On Thu, Apr 11, 2013 at 12:13 AM, Ashish gautam wrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> Caller is played music on hold but is not connected to any of the >>>>>> agents since no agent is ringing. I have sip users registered to FS ( they >>>>>> are online). I have configured a SIP user as a member for the queue. But, >>>>>> when the caller calls in, the agent doesn't ring. >>>>>> >>>>>> Please throw some light. >>>>>> >>>>>> -Ashish >>>>>> >>>>>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.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/20130416/cd8390c9/attachment-0001.html From shayne.alone at gmail.com Tue Apr 16 10:28:54 2013 From: shayne.alone at gmail.com (shayne.alone at gmail.com) Date: Tue, 16 Apr 2013 10:58:54 +0430 Subject: [Freeswitch-users] how Boost fs service startup Message-ID: Hi, how can I find the reason cause freeswitch startup service, to talk long too mutch? and how can I help it's startup process to go faster? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/b844dca3/attachment.html From nbhatti at gmail.com Tue Apr 16 10:33:00 2013 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 16 Apr 2013 09:33:00 +0300 Subject: [Freeswitch-users] how Boost fs service startup In-Reply-To: References: Message-ID: <516CF09C.5080200@gmail.com> Try starting with -nonat parameter. Thanks, -- Muhammad Naseer Bhatti shayne.alone at gmail.com wrote: > Hi, > > how can I find the reason cause freeswitch startup service, to talk > long too mutch? > and how can I help it's startup process to go faster? > > 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 From shayne.alone at gmail.com Tue Apr 16 10:49:38 2013 From: shayne.alone at gmail.com (shayne.alone at gmail.com) Date: Tue, 16 Apr 2013 11:19:38 +0430 Subject: [Freeswitch-users] how Boost fs service startup In-Reply-To: <516CF09C.5080200@gmail.com> References: <516CF09C.5080200@gmail.com> Message-ID: wow 1+ perfect! but what was the reason? really I am wondering of the affect of nat option... On Tue, Apr 16, 2013 at 11:03 AM, Muhammad Naseer Bhatti wrote: > > Try starting with -nonat parameter. > > Thanks, > -- > Muhammad Naseer Bhatti > > > > shayne.alone at gmail.com wrote: > > Hi, > > > > how can I find the reason cause freeswitch startup service, to talk > > long too mutch? > > and how can I help it's startup process to go faster? > > > > 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 > -- Regards, Ali R. Taleghani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/da122dcb/attachment.html From regis.freeswitch.org at tornad.net Tue Apr 16 10:54:12 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Tue, 16 Apr 2013 08:54:12 +0200 Subject: [Freeswitch-users] call listening and talking In-Reply-To: References: <20130414171647.c9786ad9@mail.tritonwest.net> <1365966836.24216.YahooMailClassic@web141205.mail.bf1.yahoo.com> Message-ID: originat*e* 1xxx at sip.server.com &eavesdrop*(**) * or eavesdrop(all) try that* * 2013/4/16 Michael Collins > Pastebin the debug output of your originate command. I suspect that it > will tell you what's happening. > -MC > > On Sun, Apr 14, 2013 at 12:13 PM, Sherif Omran wrote: > >> hello all, >> >> i am tryng to listen and talk to a currently connected A-B call. >> >> I tried the following: >> >> ./fs_cli >> show calls (to know the uuid) >> originator 1xxx at sip.server.com &eavesdrop >> >> this rings the 1xxx phone and then disconnects >> >> any idea >> regards, >> >> S.O >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.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/20130416/f1935968/attachment.html From shaheryarkh at gmail.com Tue Apr 16 11:11:10 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Tue, 16 Apr 2013 09:11:10 +0200 Subject: [Freeswitch-users] FS Wiki site is down Message-ID: It seems FS Wiki site is down at the moment. I am getting this on error page, Can't contact the database server: Unknown MySQL server host ' db.freeswitch.org' (2) (db.freeswitch.org) Thank you. -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/8ad14e53/attachment.html From steveayre at gmail.com Tue Apr 16 11:24:54 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 16 Apr 2013 08:24:54 +0100 Subject: [Freeswitch-users] how Boost fs service startup In-Reply-To: References: <516CF09C.5080200@gmail.com> Message-ID: <3DEA1152-CC53-4C62-BFBA-3F4234DB348E@gmail.com> It does various detection and uPNP etc handling for running FS on a LAN behind NAT to find its external IP and open the ports required. If FS is on a public static IP address none of that's required, so you can use -nonat to skip it. Steve On 16 Apr 2013, at 07:49, "shayne.alone at gmail.com" wrote: > wow 1+ > perfect! > > but what was the reason? really I am wondering of the affect of nat option... > > > On Tue, Apr 16, 2013 at 11:03 AM, Muhammad Naseer Bhatti wrote: >> >> Try starting with -nonat parameter. >> >> Thanks, >> -- >> Muhammad Naseer Bhatti >> >> >> >> shayne.alone at gmail.com wrote: >> > Hi, >> > >> > how can I find the reason cause freeswitch startup service, to talk >> > long too mutch? >> > and how can I help it's startup process to go faster? >> > >> > 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 > > > > -- > Regards, > Ali R. Taleghani > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/aa4b5ea8/attachment-0001.html From nbhatti at gmail.com Tue Apr 16 11:31:09 2013 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 16 Apr 2013 10:31:09 +0300 Subject: [Freeswitch-users] how Boost fs service startup In-Reply-To: References: <516CF09C.5080200@gmail.com> Message-ID: <516CFE3D.2030005@gmail.com> Take a look at this, http://wiki.freeswitch.org/wiki/Auto_NAT Thanks, -- Muhammad Naseer Bhatti shayne.alone at gmail.com wrote: > wow 1+ > perfect! > > but what was the reason? really I am wondering of the affect of nat > option... > > > On Tue, Apr 16, 2013 at 11:03 AM, Muhammad Naseer Bhatti > > wrote: > > > Try starting with -nonat parameter. > > Thanks, > -- > Muhammad Naseer Bhatti > > > > shayne.alone at gmail.com wrote: > > Hi, > > > > how can I find the reason cause freeswitch startup service, to talk > > long too mutch? > > and how can I help it's startup process to go faster? > > > > 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 > > > > > -- > Regards, > Ali R. Taleghani > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/23746a9a/attachment.html From steveayre at gmail.com Tue Apr 16 12:26:25 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 16 Apr 2013 09:26:25 +0100 Subject: [Freeswitch-users] FS Wiki site is down In-Reply-To: References: Message-ID: Seems ok now at least? On 16 April 2013 08:11, Muhammad Shahzad wrote: > It seems FS Wiki site is down at the moment. I am getting this on error > page, > > Can't contact the database server: Unknown MySQL server host ' > db.freeswitch.org' (2) (db.freeswitch.org) > > Thank you. > > > -- > Mit freundlichen Gr??en > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.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/20130416/aff99f13/attachment.html From jeff at askcornerstone.net Tue Apr 16 12:29:23 2013 From: jeff at askcornerstone.net (Jeff Bernhardt) Date: Tue, 16 Apr 2013 08:29:23 +0000 Subject: [Freeswitch-users] Newbie question about inbound calls/NAT (yes I checked wiki first:-) ) In-Reply-To: References: <80DFCBDE2AC6574487E3826FAF38F9CC387A508D@vega.terisol.com> <80DFCBDE2AC6574487E3826FAF38F9CC387A5507@vega.terisol.com>, Message-ID: <80DFCBDE2AC6574487E3826FAF38F9CC387A5E8F@vega.terisol.com> You know, I actually just figured it out myself just a couple hours ago.... but I'm gonna give your method a shot too since I haven't played with ACLs yet and that would give me a chance to. I discovered that the problem was actually how I'd setup Vitelity to route to the server. I didn't realize this before, but in Vitelity, you can either route to the server IP address to allow IP based authentication OR you can route to the account login name that the trunk is being registered with. The former was giving errors since I didn't have that account setup, whereas the latter worked with no problem. I'm guessing this is because with the IP based auth method, Vitelity connects to port 5060 (can't change this), which is taken as Freeswitch's default internal profile port (requires authentication) instead of port 5080 (the public profile with doesn't require authentication?), and since there was no internal user for what Vitelity was sending essentially unsolicited on 5060, it got denied. As to why it works by routing to the registered account login name I'm not sure, but perhaps that's part of what registration does; it sends to the public profile by default...? Anyway, thanks for all the help! Great to see such an active and helpful list with something I'll need a lot of help with! ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Michael Collins [msc at freeswitch.org] Sent: Monday, April 15, 2013 7:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Newbie question about inbound calls/NAT (yes I checked wiki first:-) ) For inbound calls you can skip the authentication by adding the source IP address to the "domains" section of conf/autoload_configs/acl.conf.xml. This will prevent FreeSWITCH from challenging the inbound call if it comes from that IP address. This is more desirable than disabling call authentication altogether in your SIP profile. (i.e. you are allowing only a specific IP address to come in unauth'd as opposed to letting the whole world in!) Find out the source IP address for your Vitelity SIP traffic and add it to the acl.conf.xml file in the "domains" section. It will be something like this: After you save that go to fs_cli and type "reloadacl". NOTE: simply doing a "reloadxml" is not sufficient to make the new ACL entry take effect. ("reloadxml" simply reloads the XML config; "reloadacl" does a reloadxml and then reloads the ACL entries.) Hope this helps. -MC On Mon, Apr 15, 2013 at 12:30 AM, Jeff Bernhardt > wrote: Thanks. I had a look at the links and changed "domain and even "domain_name" in vars.xml to be the external ip, but the error now just says it can's find user abcd_1.2.3.4 at 1.2.3.4 instead of @192.168.10.32. And actually, I 'm mostly confused because I thought incoming calls were treated as unauthenticated and allowed to pass through. Isn't registering the trunk with the SIP provider enough to get incoming calls accepted? The errors says "...you must configure your device to use the proper domain in it's authentication credentials," but how is the SIP trunk treated as a device? Like I said, I think I'm misunderstanding something fundamental about this. Thanks you. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Cal Leeming [Simplicity Media Ltd] [cal.leeming at simplicitymedialtd.co.uk] Sent: Sunday, April 14, 2013 3:38 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Newbie question about inbound calls/NAT (yes I checked wiki first:-) ) Hello Jeff, Here is a related thread; http://freeswitch-users.2379917.n2.nabble.com/You-must-define-a-domain-called-xx-com-in-your-directory-td7585944.html http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006232.html You need to configure your domain and username, but as you can see here your gateway/users are sending the domain '192.168.10.32', but you have not configured this domain in sofia.conf. If you are using multi-tenant then you'll need to put some more thought into how you're going to approach this. If you are not using multi-tenant (i.e. one domain only) then you could force the domain as discussed here; http://www.feedingedge.co.uk/blog/2010/08/03/techie-post-opensim-and-freeswitch-problems/ Hope this helps Cal On Sun, Apr 14, 2013 at 10:36 AM, Jeff Bernhardt > wrote: Setting up my first test box with Freeswitch. I'm really excited about getting it up and running. I'm trying to get inbound calls working from external using a Vitelity trunk. Let's say my username from Vitelity is "abcd_efg" and password is "password," external IP is 1.2.3.4 and internal is 192.168.10.32. I get registered to Vitelity fine (their web panel shows I'm registered and fs_cli confirms). However, when I make a call inbound, I get: [WARNING] sofia_reg.c:2621 Can't find user [abcd_1.2.3.4 at 192.168.10.32] from You must define a domain called '192.168.10.32' in your directory and add a user with the id="abcd_1.2.3.4" attribute and you must configure your device to use the proper domain in it's authentication credentials. I don't know if this is related, but to get around NAT, I have this in vars.xml: and this in both internal.xml and external.xml: I have a feeling I've misunderstood something fundamental about setting up Freeswitch for inbound access. I've put in the above configs as per what I've found in the wiki, but I'm not sure how to do what the warning says or if it really means I have something not setup right. What am I doing wrong? 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/3495a5d9/attachment-0001.html From ashish at nms.co.in Tue Apr 16 13:16:58 2013 From: ashish at nms.co.in (Ashish gautam) Date: Tue, 16 Apr 2013 14:46:58 +0530 Subject: [Freeswitch-users] PRI channels suspended In-Reply-To: References: Message-ID: 2013-04-16 14:41:04.214291 [ERR] ftmod_libpri.c:1950 [s1c30][1:30] -- T316 timed out, channel reached restart attempt limit '3' and is suspended This is what happening on restarting/starting the FS for all the channels(this is only for channel number 30). How to resolve it? Please help. -- Regards. On Tue, Apr 16, 2013 at 2:46 PM, Ashish gautam wrote: > Sorry Michael mistakenly sent to you personally again. Sending to the list > once more. > > > On Tue, Apr 16, 2013 at 2:45 PM, Ashish gautam wrote: > >> 2013-04-16 14:41:04.214291 [ERR] ftmod_libpri.c:1950 [s1c30][1:30] -- >> T316 timed out, channel reached restart attempt limit '3' and is suspended >> >> This is what happening on restarting/starting the FS for all the >> channels(this is only for channel number 30). >> >> How to resolve it? Please help. >> >> -- >> Regards. >> >> On Mon, Apr 15, 2013 at 9:51 PM, Michael Collins wrote: >> >>> Please only send this to the list, otherwise it is nearly impossible for >>> me to reply in such a way that everyone on the list can join the >>> conversation. >>> -MC >>> >>> >>> On Sun, Apr 14, 2013 at 11:29 PM, Ashish gautam wrote: >>> >>>> Hi, >>>> >>>> I am having another problem over here. When I run "ftdm dump 1 5" from >>>> fs_cli I get my channel status as SUSPENDED. This happens for channel >>>> number 5 to 30. Also when I try to make outgoing calls more than 4 in >>>> number, four calls originate properly whereas for the rest it shows >>>> NORMAL_CIRCUIT_CONGESTION. Here is the output: >>>> >>>> span_id: 1 >>>> chan_id: 5 >>>> physical_span_id: 1 >>>> physical_chan_id: 5 >>>> physical_status: ok >>>> physical_status_red: 0 >>>> physical_status_yellow: 0 >>>> physical_status_rai: 0 >>>> physical_status_blue: 0 >>>> physical_status_ais: 0 >>>> physical_status_general: 0 >>>> signaling_status: UP >>>> type: B >>>> state: SUSPENDED >>>> last_state: RESTART >>>> txgain: 0.00 >>>> rxgain: 0.00 >>>> cid_date: >>>> cid_name: >>>> cid_num: >>>> ani: >>>> aniII: >>>> dnis: >>>> rdnis: >>>> cause: NORMAL_UNSPECIFIED >>>> session: (none) >>>> >>>> -- States -- -- Function -- -- >>>> Location -- -- Time Offset -- >>>> DOWN => RESTART [on_dchan_up] >>>> [ftmod_libpri.c:2372] 0ms >>>> RESTART => SUSPENDED [on_timeout_t316] >>>> [ftmod_libpri.c:1952] 120008ms >>>> >>>> Time since last state change: 324790932ms >>>> >>>> Please help me getting out of this. >>>> >>>> Thanks. >>>> >>>> --Ashish >>>> >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/06a8d2e8/attachment.html From paul at cupis.co.uk Tue Apr 16 13:54:32 2013 From: paul at cupis.co.uk (Paul Cupis) Date: Tue, 16 Apr 2013 10:54:32 +0100 Subject: [Freeswitch-users] IVR, bridge, ring back problem In-Reply-To: <5168E95A.3000709@poczta.onet.pl> References: <12885609-d5bc06c5b39a91d4cd85b7400883d355@pmq2.m5r2.onet> <20130412162526.GA13139@eagle.cupis.co.uk> <5168E95A.3000709@poczta.onet.pl> Message-ID: <20130416095431.GA8297@eagle.cupis.co.uk> On Sat, Apr 13, 2013 at 07:12:58AM +0200, Andrzej wrote: > I'll check on Monday. > but: > 1. Is this normal behavior IVR? This is normal behavior, yes - the call has already been answered, so the caller is not going to be sent 180 Ringing or 183 Early media in order to hear ringing during the bridge - you need to provide audio to fill the silence. > 2. How it should be written: a. with a space, not a colon. Regards, From cal.leeming at simplicitymedialtd.co.uk Tue Apr 16 14:43:15 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 16 Apr 2013 11:43:15 +0100 Subject: [Freeswitch-users] Newbie question about inbound calls/NAT (yes I checked wiki first:-) ) In-Reply-To: <80DFCBDE2AC6574487E3826FAF38F9CC387A5E8F@vega.terisol.com> References: <80DFCBDE2AC6574487E3826FAF38F9CC387A508D@vega.terisol.com> <80DFCBDE2AC6574487E3826FAF38F9CC387A5507@vega.terisol.com> <80DFCBDE2AC6574487E3826FAF38F9CC387A5E8F@vega.terisol.com> Message-ID: Glad to hear you got it sorted, and thank you for posting back your finding - hopefully this will help others in the future! Cal On Tue, Apr 16, 2013 at 9:29 AM, Jeff Bernhardt wrote: > You know, I actually just figured it out myself just a couple hours > ago.... but I'm gonna give your method a shot too since I haven't played > with ACLs yet and that would give me a chance to. > > I discovered that the problem was actually how I'd setup Vitelity to > route to the server. I didn't realize this before, but in Vitelity, you can > either route to the server IP address to allow IP based authentication OR > you can route to the account login name that the trunk is being registered > with. The former was giving errors since I didn't have that account setup, > whereas the latter worked with no problem. I'm guessing this is because > with the IP based auth method, Vitelity connects to port 5060 (can't change > this), which is taken as Freeswitch's default internal profile port > (requires authentication) instead of port 5080 (the public profile with > doesn't require authentication?), and since there was no internal user for > what Vitelity was sending essentially unsolicited on 5060, it got denied. > As to why it works by routing to the registered account login name I'm not > sure, but perhaps that's part of what registration does; it sends to the > public profile by default...? > > Anyway, thanks for all the help! Great to see such an active and helpful > list with something I'll need a lot of help with! > > ------------------------------ > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] on behalf of Michael > Collins [msc at freeswitch.org] > *Sent:* Monday, April 15, 2013 7:15 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Newbie question about inbound calls/NAT > (yes I checked wiki first:-) ) > > For inbound calls you can skip the authentication by adding the source > IP address to the "domains" section of conf/autoload_configs/acl.conf.xml. > This will prevent FreeSWITCH from challenging the inbound call if it comes > from that IP address. This is more desirable than disabling call > authentication altogether in your SIP profile. (i.e. you are allowing only > a specific IP address to come in unauth'd as opposed to letting the whole > world in!) > > Find out the source IP address for your Vitelity SIP traffic and add it to > the acl.conf.xml file in the "domains" section. It will be something like > this: > > > > After you save that go to fs_cli and type "reloadacl". NOTE: simply doing > a "reloadxml" is not sufficient to make the new ACL entry take effect. > ("reloadxml" simply reloads the XML config; "reloadacl" does a reloadxml > and then reloads the ACL entries.) > > Hope this helps. > -MC > > On Mon, Apr 15, 2013 at 12:30 AM, Jeff Bernhardt wrote: > >> Thanks. I had a look at the links and changed "domain and even >> "domain_name" in vars.xml to be the external ip, but the error now just >> says it can's find user abcd_1.2.3.4 at 1.2.3.4 instead of @192.168.10.32. >> >> And actually, I 'm mostly confused because I thought incoming calls >> were treated as unauthenticated and allowed to pass through. Isn't >> registering the trunk with the SIP provider enough to get incoming calls >> accepted? >> >> The errors says "...you must configure your device to use the proper >> domain in it's authentication credentials," but how is the SIP trunk >> treated as a device? >> >> Like I said, I think I'm misunderstanding something fundamental about >> this. >> >> Thanks you. >> >> ------------------------------ >> *From:* freeswitch-users-bounces at lists.freeswitch.org [ >> freeswitch-users-bounces at lists.freeswitch.org] on behalf of Cal Leeming >> [Simplicity Media Ltd] [cal.leeming at simplicitymedialtd.co.uk] >> *Sent:* Sunday, April 14, 2013 3:38 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Newbie question about inbound >> calls/NAT (yes I checked wiki first:-) ) >> >> Hello Jeff, >> >> Here is a related thread; >> >> http://freeswitch-users.2379917.n2.nabble.com/You-must-define-a-domain-called-xx-com-in-your-directory-td7585944.html >> >> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006232.html >> >> You need to configure your domain and username, but as you can see here >> your gateway/users are sending the domain '192.168.10.32', but you have not >> configured this domain in sofia.conf. >> >> If you are using multi-tenant then you'll need to put some more thought >> into how you're going to approach this. >> >> If you are not using multi-tenant (i.e. one domain only) then you could >> force the domain as discussed here; >> >> http://www.feedingedge.co.uk/blog/2010/08/03/techie-post-opensim-and-freeswitch-problems/ >> >> Hope this helps >> >> Cal >> >> On Sun, Apr 14, 2013 at 10:36 AM, Jeff Bernhardt > > wrote: >> >>> Setting up my first test box with Freeswitch. I'm really excited about >>> getting it up and running. >>> >>> I'm trying to get inbound calls working from external using a Vitelity >>> trunk. Let's say my username from Vitelity is "abcd_efg" and password is >>> "password," external IP is 1.2.3.4 and internal is 192.168.10.32. I get >>> registered to Vitelity fine (their web panel shows I'm registered and >>> fs_cli confirms). However, when I make a call inbound, I get: >>> >>> [WARNING] sofia_reg.c:2621 Can't find user [abcd_1.2.3.4 at 192.168.10.32] >>> from >>> You must define a domain called '192.168.10.32' in your directory and >>> add a user with the id="abcd_1.2.3.4" attribute >>> and you must configure your device to use the proper domain in it's >>> authentication credentials. >>> >>> I don't know if this is related, but to get around NAT, I have this in >>> vars.xml: >>> >>> >>> >>> and this in both internal.xml and external.xml: >>> >>> >>> >>> I have a feeling I've misunderstood something fundamental about >>> setting up Freeswitch for inbound access. I've put in the above configs as >>> per what I've found in the wiki, but I'm not sure how to do what the >>> warning says or if it really means I have something not setup right. What >>> am I doing wrong? >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.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/20130416/c0bf8d1d/attachment-0001.html From chang33.tw at gmail.com Tue Apr 16 15:39:33 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Tue, 16 Apr 2013 19:39:33 +0800 Subject: [Freeswitch-users] plan break Message-ID: <516D3875.1060409@gmail.com> Hi, I have a chatplan like this: No matter what result the {cond(${_body} is, I don't want the plan flow go down to demo extension. What should I do? Thanks. Jimmy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/a8d99e5c/attachment.html From ashish at nms.co.in Tue Apr 16 15:59:39 2013 From: ashish at nms.co.in (Ashish gautam) Date: Tue, 16 Apr 2013 17:29:39 +0530 Subject: [Freeswitch-users] mod_fifo ongoing calls information Message-ID: Hi All, I want to get information about which members are talking to which caller in real time. How can I do that? Please throw some light. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/4bc29007/attachment.html From dujinfang at gmail.com Tue Apr 16 17:28:18 2013 From: dujinfang at gmail.com (Seven Du) Date: Tue, 16 Apr 2013 21:28:18 +0800 Subject: [Freeswitch-users] gsmopen help Message-ID: <2C3B5627AE8F4E0A9B752286E2AE8BC3@gmail.com> Hi, I just tried on Raspberry Pi with a Huawei E1550 and I got 3devices - USB0,1,2, however, when making a call the USB device name seems changed to USB3,4. Seems whenever making a call the device changes even no mod_gsmopen loaded. I tested I can send text message without problem. questions - 1) Is this because the audio is locked? How do I check if it's locked or not? 2) If not, why the device name keeps changing? it there a way to avoid that? 3) I got 3 devices when plug the USB, is that normal? how do I tell which one is control and which one is audio? It seems the last one is control. And the third one maybe 3G? Thanks, below is some log from /var/log/message Apr 16 20:41:44 raspberrypi kernel: [40750.001897] usb 1-1.2: USB disconnect, device number 9 Apr 16 20:41:44 raspberrypi kernel: [40750.002605] option1 ttyUSB0: GSM modem (1-port) converter now disconnected from ttyUSB0 Apr 16 20:41:44 raspberrypi kernel: [40750.002686] option 1-1.2:1.0: device disconnected Apr 16 20:41:44 raspberrypi kernel: [40750.002963] qmi_wwan 1-1.2:1.1: wwan0: unregister 'qmi_wwan' usb-bcm2708_usb-1.2, WWAN/QMI device Apr 16 20:41:44 raspberrypi kernel: [40750.010340] option1 ttyUSB4: GSM modem (1-port) converter now disconnected from ttyUSB4 Apr 16 20:41:44 raspberrypi kernel: [40750.010425] option 1-1.2:1.2: device disconnected Apr 16 20:41:44 raspberrypi kernel: [40750.018266] option1 ttyUSB5: GSM modem (1-port) converter now disconnected from ttyUSB5 Apr 16 20:41:44 raspberrypi kernel: [40750.018360] option 1-1.2:1.3: device disconnected Apr 16 20:41:44 raspberrypi kernel: [40750.269178] usb 1-1.2: new high-speed USB device number 10 using dwc_otg Apr 16 20:41:44 raspberrypi kernel: [40750.382002] usb 1-1.2: New USB device found, idVendor=12d1, idProduct=1003 Apr 16 20:41:44 raspberrypi kernel: [40750.382036] usb 1-1.2: New USB device strings: Mfr=2, Product=1, SerialNumber=0 Apr 16 20:41:44 raspberrypi kernel: [40750.382052] usb 1-1.2: Product: HUAWEI Mobile Apr 16 20:41:44 raspberrypi kernel: [40750.382066] usb 1-1.2: Manufacturer: HUAWEI Technology Apr 16 20:41:44 raspberrypi kernel: [40750.512562] usb 1-1.2: USB disconnect, device number 10 Apr 16 20:41:49 raspberrypi kernel: [40755.099352] usb 1-1.2: new high-speed USB device number 11 using dwc_otg Apr 16 20:41:49 raspberrypi kernel: [40755.210452] usb 1-1.2: New USB device found, idVendor=12d1, idProduct=140c Apr 16 20:41:49 raspberrypi kernel: [40755.210486] usb 1-1.2: New USB device strings: Mfr=2, Product=1, SerialNumber=0 Apr 16 20:41:49 raspberrypi kernel: [40755.210502] usb 1-1.2: Product: HUAWEI Mobile Apr 16 20:41:49 raspberrypi kernel: [40755.210516] usb 1-1.2: Manufacturer: HUAWEI Technology Apr 16 20:41:49 raspberrypi kernel: [40755.218266] scsi25 : usb-storage 1-1.2:1.0 Apr 16 20:41:49 raspberrypi kernel: [40755.224063] scsi26 : usb-storage 1-1.2:1.1 Apr 16 20:41:49 raspberrypi kernel: [40755.227981] usb 1-1.2: USB disconnect, device number 11 Apr 16 20:41:56 raspberrypi kernel: [40762.009397] usb 1-1.2: new high-speed USB device number 12 using dwc_otg Apr 16 20:41:56 raspberrypi kernel: [40762.120656] usb 1-1.2: New USB device found, idVendor=12d1, idProduct=140c Apr 16 20:41:56 raspberrypi kernel: [40762.120688] usb 1-1.2: New USB device strings: Mfr=2, Product=1, SerialNumber=0 Apr 16 20:41:56 raspberrypi kernel: [40762.120705] usb 1-1.2: Product: HUAWEI Mobile Apr 16 20:41:56 raspberrypi kernel: [40762.120720] usb 1-1.2: Manufacturer: HUAWEI Technology Apr 16 20:41:56 raspberrypi kernel: [40762.131735] option 1-1.2:1.0: GSM modem (1-port) converter detected Apr 16 20:41:56 raspberrypi kernel: [40762.132274] usb 1-1.2: GSM modem (1-port) converter now attached to ttyUSB0 Apr 16 20:41:56 raspberrypi kernel: [40762.138276] qmi_wwan 1-1.2:1.1: cdc-wdm0: USB WDM device Apr 16 20:41:56 raspberrypi kernel: [40762.142683] qmi_wwan 1-1.2:1.1: wwan0: register 'qmi_wwan' at usb-bcm2708_usb-1.2, WWAN/QMI device, ea:1a:89:2e:60:ec Apr 16 20:41:56 raspberrypi kernel: [40762.148396] option 1-1.2:1.2: GSM modem (1-port) converter detected Apr 16 20:41:56 raspberrypi kernel: [40762.148977] usb 1-1.2: GSM modem (1-port) converter now attached to ttyUSB3 Apr 16 20:41:56 raspberrypi kernel: [40762.153480] option 1-1.2:1.3: GSM modem (1-port) converter detected Apr 16 20:41:56 raspberrypi kernel: [40762.154368] usb 1-1.2: GSM modem (1-port) converter now attached to ttyUSB4 Apr 16 20:41:56 raspberrypi kernel: [40762.156892] scsi31 : usb-storage 1-1.2:1.4 Apr 16 20:41:56 raspberrypi kernel: [40762.161995] scsi32 : usb-storage 1-1.2:1.5 Apr 16 20:41:57 raspberrypi kernel: [40763.153006] scsi 31:0:0:0: CD-ROM HUAWEI Mass Storage 2.31 PQ: 0 ANSI: 2 Apr 16 20:41:57 raspberrypi kernel: [40763.164198] scsi 32:0:0:0: Direct-Access HUAWEI MMC Storage 2.31 PQ: 0 ANSI: 2 Apr 16 20:41:57 raspberrypi kernel: [40763.166151] sr0: scsi-1 drive Apr 16 20:41:57 raspberrypi kernel: [40763.175965] sd 32:0:0:0: [sda] Attached SCSI removable disk -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/2b5c54cc/attachment.html From jmillan at aliax.net Tue Apr 16 12:16:06 2013 From: jmillan at aliax.net (=?UTF-8?B?Sm9zw6kgTHVpcyBNaWxsw6Fu?=) Date: Tue, 16 Apr 2013 10:16:06 +0200 Subject: [Freeswitch-users] Respect the request URI domain and To URI domain when bridging Message-ID: Hi all, Every call bridged though a specific 'gateway' does rewrite the SIP request URI domain and To URI domain values to the gateway's 'proxy' parameter value. I would like to respect such values and receive in my proxy the SIP messages with their original request URI domain and To URI domain. This is, I would like to be able to send a SIP message with the following request URI and To headers, to the IP address 2.2.2.2 ``` INVITE sip:jmillan at 1.2.3.4 SIP/2.0 Via: SIP/2.0/TCP bl23sl8lmku7.invalid;branch=z9hG4bK9420964 To: ``` ``` Is it possible to handle this with FreeSwitch? I have followed the FS wiki but I've been unable to handle this. Thanks in advance. -- Jos? Luis Mill?n -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/2aa3740b/attachment-0001.html From jmillan at aliax.net Tue Apr 16 12:19:51 2013 From: jmillan at aliax.net (=?UTF-8?B?Sm9zw6kgTHVpcyBNaWxsw6Fu?=) Date: Tue, 16 Apr 2013 10:19:51 +0200 Subject: [Freeswitch-users] Respect the request URI domain and To URI domain when bridging In-Reply-To: References: Message-ID: Let me fix an error: This is the message I would like to send to the IP 2.2.2.2 (my proxy) ``` INVITE sip:jmillan at 1.2.3.4 SIP/2.0 Via: SIP/2.0/TCP bl23sl8lmku7.invalid;branch=z9hG4bK9420964 To: >``` ``` Regards. 2013/4/16 Jos? Luis Mill?n > Hi all, > > Every call bridged though a specific 'gateway' does rewrite the SIP > request URI domain and To URI domain values to the gateway's 'proxy' > parameter value. > > I would like to respect such values and receive in my proxy the SIP > messages with their original request URI domain and To URI domain. > > This is, I would like to be able to send a SIP message with the following > request URI and To headers, to the IP address 2.2.2.2 > > ``` > INVITE sip:jmillan at 1.2.3.4 SIP/2.0 Via: SIP/2.0/TCP > bl23sl8lmku7.invalid;branch=z9hG4bK9420964 To: ``` > ``` > > Is it possible to handle this with FreeSwitch? > > I have followed the FS wiki but I've been unable to handle this. > > Thanks in advance. > > -- > Jos? Luis Mill?n > -- Jos? Luis Mill?n -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/5d8a8956/attachment.html From akashdeep.co at gmail.com Tue Apr 16 14:45:07 2013 From: akashdeep.co at gmail.com (Akash Deep Verma) Date: Tue, 16 Apr 2013 16:15:07 +0530 Subject: [Freeswitch-users] PRI channels suspended Message-ID: Hi Ashish, This is a hack but you get what you need. This is because, whenever retry attempt is made 1 channel per pri get started, as there are 30 channels and retry attempts are 3 so the channels that freeswitch load is in between 3-6, so I increased retry attempt and complied it again. As I got rid of this error but it takes some time to load the pri channels (30*10 = 300seconds(5 seconds) max). Also, remember if you dial a call in between restarting phase you may face some error. To avoid this, dial call only after all pri channel get started. Here are the steps: In freeswitch source: Edit libs/freetdm/src/ftmod/ftmod_libpri/ftmod_libpri.h 1. Decrease timeout of channels. #define T316_TIMEOUT_MS_DEFAULT 30000 /* 30 sec */ change this to #define T316_TIMEOUT_MS_DEFAULT 10000 /* 10 sec */ 2. Increase attempts to 30. #define T316_ATTEMPT_LIMIT_DEFAULT 3 change this to #define T316_ATTEMPT_LIMIT_DEFAULT 30 3. Increase maximum limit as according to default(set more than that of default) #define T316_ATTEMPT_LIMIT_MAX 10 change this to #define T316_ATTEMPT_LIMIT_MAX 35 Thanks & Regards, - Akash Deep Verma http://in.linkedin.com/in/akashdeep1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/fb571cfe/attachment-0001.html From akashdeep.co at gmail.com Tue Apr 16 15:09:35 2013 From: akashdeep.co at gmail.com (Akash Deep Verma) Date: Tue, 16 Apr 2013 16:39:35 +0530 Subject: [Freeswitch-users] PRI channels suspended Message-ID: Hi Ashish, This is a hack but you get what you need. This is because, whenever retry attempt is made 1 channel per pri get started, as there are 30 channels and retry attempts are 3 so the channels that freeswitch load is in between 3-6, so I increased retry attempt and complied it again. As I got rid of this error but it takes some time to load the pri channels (30*10 = 300seconds(5 seconds) max). Also, remember if you dial a call in between restarting phase you may face some error. To avoid this, dial call only after all pri channel get started. Here are the steps: In freeswitch source: Edit libs/freetdm/src/ftmod/ftmod_libpri/ftmod_libpri.h 1. Decrease timeout of channels. #define T316_TIMEOUT_MS_DEFAULT 30000 /* 30 sec */ change this to #define T316_TIMEOUT_MS_DEFAULT 10000 /* 10 sec */ 2. Increase attempts to 30. #define T316_ATTEMPT_LIMIT_DEFAULT 3 change this to #define T316_ATTEMPT_LIMIT_DEFAULT 30 3. Increase maximum limit as according to default(set more than that of default) #define T316_ATTEMPT_LIMIT_MAX 10 change this to #define T316_ATTEMPT_LIMIT_MAX 35 Thanks & Regards, - Akash Deep Verma http://in.linkedin.com/in/akashdeep1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/467edeb9/attachment.html From jmillan at aliax.net Tue Apr 16 15:51:23 2013 From: jmillan at aliax.net (=?UTF-8?B?Sm9zw6kgTHVpcyBNaWxsw6Fu?=) Date: Tue, 16 Apr 2013 13:51:23 +0200 Subject: [Freeswitch-users] plan break In-Reply-To: <516D3875.1060409@gmail.com> References: <516D3875.1060409@gmail.com> Message-ID: Hi, 2013/4/16 Jimmy Chang > Hi, > > I have a chatplan like this: > > > > > > > > expression="^YES$"> > > > > > > > > > > > > > > > No matter what result the {cond(${_body} is, I don't want the plan flow > go down to demo extension. > What should I do? > > What about an 'anti-action' for the case the condition is not satisfied? > Thanks. > Jimmy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jos? Luis Mill?n -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/885ee46a/attachment.html From netcentrica at gmail.com Tue Apr 16 19:03:39 2013 From: netcentrica at gmail.com (Adam Raszynski) Date: Tue, 16 Apr 2013 17:03:39 +0200 Subject: [Freeswitch-users] ringback and transfer_ringback ignored by FS Message-ID: Hi Scenario: - SIP only - inbound call, hits dialplan - call answered, some IVR used - call bridged to selected department The problem is, that I would like to play music on hold while trying to bridge. FS seems to completely ignore custom ringback settings: tried also with: and All with no success. I see in console that variables are set, but for some reason FS ignores them and there is no audio while bridging. At the same time I can play/stream sound file before bridge, but this operation is blocking and stream can not be run in background so it can't be used for MoH purpose. Any hints what should I check or how to achive nonblocking MoH? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/4bbb7785/attachment.html From msc at freeswitch.org Tue Apr 16 19:11:24 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 16 Apr 2013 08:11:24 -0700 Subject: [Freeswitch-users] mod_fifo: agents are not ringing In-Reply-To: References: Message-ID: try using the fs_cli fifo member command to add these. See if you get any errors. That might shed some light on what's happening. -MC On Mon, Apr 15, 2013 at 10:46 PM, Ashish gautam wrote: > Michael, > > My fifo.conf.xml is: > > > > > user/1004 at 10.1.30.229 > > user/1003 at 10.1.30.229 > > > > > > Its according to the example shown on the wiki page. Still fifo list shows > members not added. > > -- > Ashish > > On Mon, Apr 15, 2013 at 9:48 PM, Michael Collins wrote: > >> It's just a simple sofia dialstring. Here's an example: >> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback >> >> -MC >> >> >> On Sun, Apr 14, 2013 at 10:42 PM, Ashish gautam wrote: >> >>> Thanks Michael, >>> >>> Probably the originate dialstring I am using is not correct. Can you >>> please tell what is the proper format for that? >>> >>> On Sat, Apr 13, 2013 at 12:22 AM, Michael Collins wrote: >>> >>>> For some reason your agents are not actually set as members of the >>>> queue. Onhook agents will be listed in the section and off-hook >>>> agents will be listed in the section. Evidently, whatever >>>> you're doing to add the agents is not working. You may want to try >>>> explicitly adding them from the fs_cli using the fifo_member command and >>>> seeing how that goes. >>>> >>>> -MC >>>> >>>> >>>> On Thu, Apr 11, 2013 at 9:32 PM, Ashish gautam wrote: >>>> >>>>> The output is : >>>>> >>>>> >>>>> >>>>> >>>> caller_count="0" waiting_count="0" importance="0" outbound_per_cycle="1" >>>>> ring_timeout="60" default_lag="30" outbound_priority="5" >>>>> outbound_strategy="ringall"> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Thu, Apr 11, 2013 at 9:42 PM, Michael Collins wrote: >>>>> >>>>>> What's the output of fscli command: fifo list cool_fifo at 10.1.30.229 >>>>>> >>>>>> -MC >>>>>> >>>>>> On Thu, Apr 11, 2013 at 12:13 AM, Ashish gautam wrote: >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> Caller is played music on hold but is not connected to any of the >>>>>>> agents since no agent is ringing. I have sip users registered to FS ( they >>>>>>> are online). I have configured a SIP user as a member for the queue. But, >>>>>>> when the caller calls in, the agent doesn't ring. >>>>>>> >>>>>>> Please throw some light. >>>>>>> >>>>>>> -Ashish >>>>>>> >>>>>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/10b2284a/attachment-0001.html From msc at freeswitch.org Tue Apr 16 19:17:54 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 16 Apr 2013 08:17:54 -0700 Subject: [Freeswitch-users] mod_fifo ongoing calls information In-Reply-To: References: Message-ID: at fs_cli you can do: fifo list or fifo list_verbose -Michael On Tue, Apr 16, 2013 at 4:59 AM, Ashish gautam wrote: > Hi All, > > I want to get information about which members are talking to which caller > in real time. How can I do that? > > Please throw some light. > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/b4d398e2/attachment.html From msc at freeswitch.org Tue Apr 16 19:56:32 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 16 Apr 2013 08:56:32 -0700 Subject: [Freeswitch-users] plan break In-Reply-To: References: <516D3875.1060409@gmail.com> Message-ID: I'd say you want break="always" in that condition: Try that and let us know if it works. -MC On Tue, Apr 16, 2013 at 4:51 AM, Jos? Luis Mill?n wrote: > Hi, > > > > 2013/4/16 Jimmy Chang > >> Hi, >> >> I have a chatplan like this: >> >> >> >> >> >> >> >> > expression="^YES$"> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> No matter what result the {cond(${_body} is, I don't want the plan flow >> go down to demo extension. >> What should I do? >> >> > What about an 'anti-action' for the case the condition is not satisfied? > > > >> Thanks. >> Jimmy >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Jos? Luis Mill?n > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/b46dc0b8/attachment.html From jmesquita at freeswitch.org Tue Apr 16 20:32:44 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 16 Apr 2013 13:32:44 -0300 Subject: [Freeswitch-users] Respect the request URI domain and To URI domain when bridging In-Reply-To: References: Message-ID: >From sofia_glue.c: const char *invite_record_route = switch_channel_get_variable(tech_pvt->channel, "sip_invite_record_route"); const char *invite_route_uri = switch_channel_get_variable(tech_pvt->channel, "sip_invite_route_uri"); const char *invite_full_from = switch_channel_get_variable(tech_pvt->channel, "sip_invite_full_from"); const char *invite_full_to = switch_channel_get_variable(tech_pvt->channel, "sip_invite_full_to"); const char *handle_full_from = switch_channel_get_variable(tech_pvt->channel, "sip_handle_full_from"); const char *handle_full_to = switch_channel_get_variable(tech_pvt->channel, "sip_handle_full_to"); const char *force_full_from = switch_channel_get_variable(tech_pvt->channel, "sip_force_full_from"); const char *force_full_to = switch_channel_get_variable(tech_pvt->channel, "sip_force_full_to"); Glad to see you using FS Jos? Luis! :-D Welcome. Jo?o Mesquita On Tue, Apr 16, 2013 at 5:19 AM, Jos? Luis Mill?n wrote: > Let me fix an error: > > > This is the message I would like to send to the IP 2.2.2.2 (my proxy) > ``` > INVITE sip:jmillan at 1.2.3.4 SIP/2.0 Via: SIP/2.0/TCP > bl23sl8lmku7.invalid;branch=z9hG4bK9420964 To: > >``` > ``` > > Regards. > > > > 2013/4/16 Jos? Luis Mill?n > >> Hi all, >> >> Every call bridged though a specific 'gateway' does rewrite the SIP >> request URI domain and To URI domain values to the gateway's 'proxy' >> parameter value. >> >> I would like to respect such values and receive in my proxy the SIP >> messages with their original request URI domain and To URI domain. >> >> This is, I would like to be able to send a SIP message with the following >> request URI and To headers, to the IP address 2.2.2.2 >> >> ``` >> INVITE sip:jmillan at 1.2.3.4 SIP/2.0 Via: SIP/2.0/TCP >> bl23sl8lmku7.invalid;branch=z9hG4bK9420964 To: ``` >> ``` >> >> Is it possible to handle this with FreeSwitch? >> >> I have followed the FS wiki but I've been unable to handle this. >> >> Thanks in advance. >> >> -- >> Jos? Luis Mill?n >> > > > > -- > Jos? Luis Mill?n > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130416/9c772301/attachment.html From stkn at openisdn.net Tue Apr 16 20:41:58 2013 From: stkn at openisdn.net (Stefan Knoblich) Date: Tue, 16 Apr 2013 18:41:58 +0200 Subject: [Freeswitch-users] PRI channels suspended In-Reply-To: References: Message-ID: <516D7F56.3020706@openisdn.net> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 16.04.2013 11:16, Ashish gautam wrote: > 2013-04-16 14:41:04.214291 [ERR] ftmod_libpri.c:1950 [s1c30][1:30] -- T316 timed out, channel reached restart attempt limit '3' and is suspended > > This is what happening on restarting/starting the FS for all the channels(this is only for channel number 30). > > How to resolve it? Please help. > Add