June 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

OpenSIPS Summit on Aug 4th.. come to cluecon early!
by Alex Goulis 18 Jun '14

18 Jun '14

The Windy City, Chi-Town, the Second City, the City of Big Shoulders, and the City That Works are just some of Chicago's many nicknames... *On August 4th, 2014 it will be the the "OpenSIPS" city!* If you've ever wondered how OpenSIPS can help you with your Freeswitch deployments, now's your chance. For all of you attending ClueCon, we'll be holding an OpenSIPS summit on Aug 4 running most of the day before ClueCon officially kicks off. *All ClueCon attendees receive a 25% discount by using the code "**cluecon2014".* Last year's summit was a tremendous success. Everyone that attended enjoyed interesting topics and discussions. Along with this announcement, we would like to announce next the two speakers for this years summit: *Steve Lorusso - Grasshopper - */"OpenSIPS in Production//"/ Engineers at Grasshopper recently deployed OpenSIPS as a production SBC for peering with multiple carriers. This session will cover what was done to setup, network, monitor, and operate OpenSIPS in an virtual HA configuration. Steve is VP of Engineering and Operations at Grasshopper, and oversees the technical direction, architecture, and implementation of software and services. He's got a track record of developing products from a 'startup' environment and guiding them to a mature business while building expert teams and processes. Before joining Grasshopper, Steve held technical management positions in Systems Engineering and Software Development. He was a day-1 core team member at Unisphere Networks, built SIP-based communications products for the Carrier marketplace, and then guided development through an acquisition (Siemens/Juniper) and subsequent global expansion into many countries. Steve is a seasoned engineering veteran with tier-1 Carrier and large Enterprise experience. He combines technical, business, and management experience with a passion for building innovative products. Steve has a number of patent applications relating to IP communications and redundant systems design. He holds a BSEE from Northeastern University and an MSEE degree from Boston University. *James Pearson - ViciDial - "*/Creating a routing and billing platform: How hard could it be?"/ Design and implementation of OpenSIPS into a billing and routing platform for a multi-tenanted high-capacity hosted call center. Topics include looking at what was required, design challenges and compromises, what wasn't known to be unknown, and other random craziness found along the way. James Pearson is a founder of The ViciDial Group in 2008 as well as the Director of Hosting for ViciHost.com. Currently maintains the OpenSIPS package on the OpenSuSE Build Service as well as contributing to other OBS projects. Experience with VoIP began in 2005 with Asterisk v.1.2 implementing customized PBXs for small business' as well as general network environment monitoring and design. Other areas of interest include networking, system automation, and new challenges You can get more information about the summit at: http://www.opensips.org/Community/Summit-2014Chicago Registration: http://www.opensips.org/Community/Summit-Registration Last year's summit highlights: http://www.opensips.org/Community/Summit-2013Chicago We look forward to seeing you! The OpenSIPS Team!

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DTMF I send are not recognized.
by Neo Haux 18 Jun '14

18 Jun '14

Hi all, I have an issue to send DTMF to an external extension (goes through my PSTN) My actual config is: |--|FXS-phone FS------LAN-----CISCO SPA3000--| |--| FXO-to-pstn Either I make a call from FXS of a soft SIP phone in my LAN the DTMF I send are not recognized at all by the number I call. When I send the DTMF I see in the console: freeswitch@internal> 2014-06-17 15:19:59.846809 [DEBUG] switch_rtp.c:5788 RTP RECV DTMF 1:1600 2014-06-17 15:19:59.846809 [DEBUG] switch_channel.c:486 RECV DTMF 1:1600 2014-06-17 15:19:59.846809 [DEBUG] switch_ivr_bridge.c:470 Send signal sofia/internal/5142252398@192.168.0.102:5061 [BREAK] 2014-06-17 15:19:59.846809 [DEBUG] switch_rtp.c:4314 Send start packet for [1] ts=1562418126 dur=160/160/1600 seq=48292 lw=1562418126 2014-06-17 15:19:59.866877 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=320/320/1600 seq=48293 lw=1562418286 2014-06-17 15:19:59.886848 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=480/480/1600 seq=48294 lw=1562418446 2014-06-17 15:19:59.906840 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=640/640/1600 seq=48295 lw=1562418606 2014-06-17 15:19:59.926964 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=800/800/1600 seq=48296 lw=1562418766 2014-06-17 15:19:59.946846 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=960/960/1600 seq=48297 lw=1562418926 2014-06-17 15:19:59.966838 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=1120/1120/1600 seq=48298 lw=1562419086 2014-06-17 15:19:59.986853 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=1280/1280/1600 seq=48299 lw=1562419246 2014-06-17 15:20:00.006849 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=1440/1440/1600 seq=48300 lw=1562419406 2014-06-17 15:20:00.026839 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=1562418126 dur=1600/1600/1600 seq=48301 lw=1562419406 2014-06-17 15:20:00.026839 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=1562418126 dur=1600/1600/1600 seq=48302 lw=1562419406 2014-06-17 15:20:00.046848 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=1562418126 dur=1600/1600/1600 seq=48303 lw=1562419406 freeswitch@internal> freeswitch@internal> freeswitch@internal> freeswitch@internal> freeswitch@internal> freeswitch@internal> freeswitch@internal> freeswitch@internal> 2014-06-17 15:20:04.926811 [DEBUG] switch_rtp.c:5788 RTP RECV DTMF 1:1600 2014-06-17 15:20:04.926811 [DEBUG] switch_channel.c:486 RECV DTMF 1:1600 2014-06-17 15:20:04.926811 [DEBUG] switch_ivr_bridge.c:470 Send signal sofia/internal/5142252398@192.168.0.102:5061 [BREAK] 2014-06-17 15:20:04.926811 [DEBUG] switch_rtp.c:4314 Send start packet for [1] ts=1562459406 dur=160/160/1600 seq=48541 lw=1562459406 2014-06-17 15:20:04.946882 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=320/320/1600 seq=48542 lw=1562459566 2014-06-17 15:20:04.966839 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=480/480/1600 seq=48543 lw=1562459726 2014-06-17 15:20:04.986837 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=640/640/1600 seq=48544 lw=1562459886 2014-06-17 15:20:05.006847 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=800/800/1600 seq=48545 lw=1562460046 2014-06-17 15:20:05.026846 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=960/960/1600 seq=48546 lw=1562460206 2014-06-17 15:20:05.046839 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=1120/1120/1600 seq=48547 lw=1562460366 2014-06-17 15:20:05.066839 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=1280/1280/1600 seq=48548 lw=1562460526 2014-06-17 15:20:05.086839 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=1440/1440/1600 seq=48549 lw=1562460686 2014-06-17 15:20:05.106839 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=1562459406 dur=1600/1600/1600 seq=48550 lw=1562460686 2014-06-17 15:20:05.106839 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=1562459406 dur=1600/1600/1600 seq=48551 lw=1562460686 2014-06-17 15:20:05.126850 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=1562459406 dur=1600/1600/1600 seq=48552 lw=1562460686 freeswitch@internal> freeswitch@internal> freeswitch@internal> When I receive a call and the caller click on digits (send me DTMF) I see this in the console: freeswitch@internal> 2014-06-17 15:18:06.466803 [DEBUG] switch_rtp.c:5788 RTP RECV DTMF 1:1604 2014-06-17 15:18:06.466803 [DEBUG] switch_channel.c:486 RECV DTMF 1:1604 2014-06-17 15:18:06.466803 [DEBUG] switch_ivr_bridge.c:470 Send signal sofia/internal/sip:1000@192.168.0.102:5060 [BREAK] 2014-06-17 15:18:06.486822 [DEBUG] switch_rtp.c:4314 Send start packet for [1] ts=407942050 dur=240/240/1604 seq=40146 lw=407942050 2014-06-17 15:18:06.506818 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=407942050 dur=480/480/1604 seq=40147 lw=407942290 2014-06-17 15:18:06.546821 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=407942050 dur=720/720/1604 seq=40148 lw=407942530 2014-06-17 15:18:06.566821 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=407942050 dur=960/960/1604 seq=40149 lw=407942770 2014-06-17 15:18:06.606824 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=407942050 dur=1200/1200/1604 seq=40150 lw=407943010 2014-06-17 15:18:06.626820 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=407942050 dur=1440/1440/1604 seq=40151 lw=407943250 2014-06-17 15:18:06.666823 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=407942050 dur=1680/1680/1604 seq=40152 lw=407943250 2014-06-17 15:18:06.666823 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=407942050 dur=1680/1680/1604 seq=40153 lw=407943250 2014-06-17 15:18:06.666823 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=407942050 dur=1680/1680/1604 seq=40154 lw=407943250 I have no isea where to look. Is it a config to change within FS or SPA3000 ? What type of conf should I check ? In the past I had a Digium card+ asterisk and I had never had an issue with dtmf since the default config did work. I am new to FS so any idea is welcome. Thanks

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DNS name as the public IP respond 403
by Shahzad Bhatti 18 Jun '14

18 Jun '14

> > here i post 2 sip traces first for successive registration with same > credentials but using static ip > with headings as on line 40 Sip Trace of Successive Call using Public IP and error sip trace on line 288 SIP TRACE OF USING DNS CALL FOR REGISTER (having problem) any further information is required to trace the issue i will happy to send also regards Shahzad Bhatti ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 17 Jun 2014 12:59:02 -0500 > Subject: Re: [Freeswitch-users] DNS name as the public IP respond 403 > Raise the issue with your provider 403 usually means your credentials are > invalid. > > -- > Brian West > brian(a)freeswitch.org > > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > On Jun 17, 2014, at 4:57 AM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> > wrote: > > Output of sofia status profile and sip trace log is here > > http://pastebin.freeswitch.com/22750 > > Regards > > Shahzad Bhatti > > ---------- Forwarded message ---------- >> From: Brian West <brian(a)freeswitch.org> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Mon, 16 Jun 2014 13:41:44 -0500 >> Subject: Re: [Freeswitch-users] DNS name as the public IP respond 403 >> Please pastebin a sip trace of this. And the output of 'sofia status >> profile internal' or what ever the profile name is. >> -- >> Brian West >> brian(a)freeswitch.org >> >> <whmcslogo.png> >> >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> On Jun 16, 2014, at 12:47 PM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> >> wrote: >> >> Hi, i am trying to use DNS Name and for that i start my Freeswitch Using >> -nonat and in sip profile create a new profile with the following changes >> >> <param name="sip-port" value="8080"/> >> >> and >> >> <param name="ext-rtp-ip" value="host:sip.mydomain-name.com"/> >> >> rest conf is same as internal and when i try to register i got 403 >> response while on internal profile it works fine >> i am using >> FreeSWITCH Version 1.5.8b+git~20131202T224806Z~e127ca7d40~64bit (git >> e127ca7 2013-12-02 22:48:06Z 64bit) >> >> on centos 64-bit >> >> reply me what else i am missing >> >> regards >> >> Shahzad Bhatti >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Fwd: Recording in flac format ?
by julien terrasson 18 Jun '14

18 Jun '14

Hi, I'm using freeswitch version 1.4.5 I would like to record a session in .flac format. I assume this could be somehow possible since FS rely on libsndfile-1.0.26pre5 and libsndfile claims flac support since v1.0.18. I tried to compiled FS with instruction from description of https://jira.freeswitch.org/browse/FS-1197 : Apt-get install libflac-dev and "./configure --enable-flac".. However that didn't help, flac extension is still not usable on FS recording app. Does anyone have an idea how to have flac format enabled ? Regards, J.Terrasson

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 96, Issue 137
by Amin Dhamani Askhanuman 17 Jun '14

17 Jun '14

Yes its problem on inbound side. But where to set this parameter? in vars.xml? On Tue, Jun 17, 2014 at 1:40 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)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(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)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: Voice Mail issue (Brian West) > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 16 Jun 2014 13:40:15 -0500 > Subject: Re: [Freeswitch-users] Voice Mail issue > Answer, then do playback silence_stream://1500, that should clear it, > Sonus on the inbound side? > > -- > Brian West > brian(a)freeswitch.org > > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > > > On Jun 16, 2014, at 9:55 AM, Amin Dhamani Askhanuman <amin(a)askhanuman.com> > wrote: > > Hi, > > Currently we are facing issue with voicemail. Sound quality is not good. > Even caller can't hear the greeting properly. Can you guide how to resolve > it. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)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, Amin Dhamani Amin Dhamani | System Administrator | +66 911167159 | Krits Tower, 1032/1-5, 14 Rama 4, Bangkok 10120 *www.askhanuman.co.th <http://www.askhanuman.co.th/>* | www.facebook.com/AskHanuman | amin(a)askhanuman.com This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. ______________________________________________________________________________ -- This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. ______________________________________________________________________________

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DNS name as the public IP respond 403
by Shahzad Bhatti 17 Jun '14

17 Jun '14

Output of sofia status profile and sip trace log is here http://pastebin.freeswitch.com/22750 Regards Shahzad Bhatti ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 16 Jun 2014 13:41:44 -0500 > Subject: Re: [Freeswitch-users] DNS name as the public IP respond 403 > Please pastebin a sip trace of this. And the output of 'sofia status > profile internal' or what ever the profile name is. > -- > Brian West > brian(a)freeswitch.org > > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > On Jun 16, 2014, at 12:47 PM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> > wrote: > > Hi, i am trying to use DNS Name and for that i start my Freeswitch Using > -nonat and in sip profile create a new profile with the following changes > > <param name="sip-port" value="8080"/> > > and > > <param name="ext-rtp-ip" value="host:sip.mydomain-name.com"/> > > rest conf is same as internal and when i try to register i got 403 > response while on internal profile it works fine > i am using > FreeSWITCH Version 1.5.8b+git~20131202T224806Z~e127ca7d40~64bit (git > e127ca7 2013-12-02 22:48:06Z 64bit) > > on centos 64-bit > > reply me what else i am missing > > regards > > Shahzad Bhatti > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Freeswitch seems to be giving conflicting timestamps - what's the deal here?
by Phil Quesinberry 17 Jun '14

17 Jun '14

Looking into an intermittent problem that we're having with one of our DIDs, I just noticed that some of the logged timestamps don't seem to make sense, it's like there are two different clocks at work here. What's interesting is that I'm seeing this wrong time in received contact header info as well but I'm guessing that's coming back to us from our own NTP timestamp in the SDP we send back to them after the initial invite. The system time is set correctly. Here is a short excerpt of the console log with siptrace enabled on the external profile demonstrating this behavior: recv 939 bytes from udp/[209.216.2.202]:5060 at 16:10:58.987966: ------------------------------------------------------------------------ INVITE sip:1410xxx7612@173.59.xxx.xxx:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 209.216.2.202:5060;branch=z9hG4bK68239f89;rport From: "+1410xxx3048" <sip:+1410xxx3048@209.216.2.202>;tag=as0e92c0c82014-06-17 12:07:58.767052 [DEBUG] switch_core_session.c:1034 Send signal sofia/external/+1410xxx3048(a)209.216.2.202 [BREAK] To: <sip:1410xxx7612@voip.ourdomain.com:5080>;tag=Hj722eXX0yj8S Contact: <sip:+1410xxx3048@209.216.2.202>2014-06-17 12:07:58.767052 [DEBUG] switch_core_session.c:1034 Send signal sofia/external/+1410xxx3048(a)209.216.2.202 [BREAK] Call-ID: 63cd75ae5caa77be67d5a6944e86f5fe(a)209.216.2.202 CSeq: 103 INVITE User-Agent: DidForSale-GW Max-Forwards: 70 Remote-Party-ID: "+1410xxx3048" <sip:+1410xxx3048@209.216.2.202>;privacy=off;screen=no Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 289 v=0 o=root 2802 2803 IN IP4 207.223.69.137 s=session c=IN IP4 207.223.69.137 t=0 0 m=audio 51480 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - ------------------------------------------------------------------------ send 405 bytes to udp/[209.216.2.202]:5060 at 16:10:58.988281: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.216.2.202:5060;branch=z9hG4bK68239f89;rport=5060 From: "+1410xxx3048" <sip:+1410xxx3048@209.216.2.202>;tag=as0e92c0c8 To: <sip:1410xxx7612@voip.ourdomain.com:5080>;tag=Hj722eXX0yj8S Call-ID: 63cd75ae5caa77be67d5a6944e86f5fe(a)209.216.2.202 CSeq: 103 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.6b+git~20131022T212715Z~20a55d3cc6 Content-Length: 0 ------------------------------------------------------------------------ 2014-06-17 12:07:58.787035 [DEBUG] sofia.c:5863 Channel sofia/external/+1410xxx3048(a)209.216.2.202 entering state [ready][200] 2014-06-17 12:07:58.787035 [DEBUG] sofia.c:5863 Channel sofia/external/+1410xxx3048(a)209.216.2.202 entering state [received][100] 2014-06-17 12:07:58.787035 [DEBUG] sofia.c:5873 Remote SDP: v=0 o=root 2802 2803 IN IP4 207.223.69.137 s=session c=IN IP4 207.223.69.137 t=0 0 m=audio 51480 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 2014-06-17 12:07:58.787035 [DEBUG] switch_core_media.c:2990 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2014-06-17 12:07:58.787035 [DEBUG] switch_core_media.c:1879 Already using PCMU 2014-06-17 12:07:58.787035 [DEBUG] switch_core_media.c:3161 Set 2833 dtmf send/recv payload to 101 Is this expected behavior or is this a bug in the version we're using? Thanks. Phil Quesinberry Q Systems Engineering, Inc. Embedded Hardware and Software Development (410) 969-8002 http://www.qsystemsengineering.com <http://www.qsystemsengineering.com/>

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FreeSWITCH Developer OpenVZ Container Template
by Brian West 17 Jun '14

17 Jun '14

The template can be download from http://files.freeswitch.org/openvz/ The root password is 'freeswitch', It already has a checkout of the latest FS git repo in /usr/src/freeswitch.git, The FS Team SSH Keys are installed, you can report any feedback or bugs on the VM/CT ONLY to https://jira.freeswitch.org/browse/FSVIRT, Any bugs related to FreeSWITCH please report to https://jira.freeswitch.org/browse/FS Thanks, -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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Knowing in dialplan XMl when a bridge was succesful/failed
by Grant Bagdasarian 17 Jun '14

17 Jun '14

Hello, Let me explain my scenario first. I'm using ESL inbound to originate a call which will go into a certain context when the call is answered. After this the dialplan XML takes over. ESL is out of the scope after this. Once answered, I give the user the option to be bridged to another destination. >From what I understood, when bridging a call, there is no way to execute dialplan application anymore. Is there a way to know if a bridge was successful or not, regardless of who hung up the call? If the B leg hangs up, I can use the hangup_after_bridg(false), and perhaps access some variable which will tell me if a bridge was successful (if available). But if the A Leg hangs up during the bridge I wouldn't be able to do anything anymore. Any ideas? Regards, Grant

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No hangup from Bria on Mac
by Oscar Franzen 17 Jun '14

17 Jun '14

I register Bria with freeswitch through the external profile (but on the internal network) and a user in the directory. This setup works fine. I originate a call to the bria (sofia/domain/userID and play moh (extension 9664). When I hang up in Bria the call ends there but no event is sent to FS. Any ideas on what I might be doing wrong? -- Regards *Oscar Franzén**Developer* *Advisa* oscar.franzen(a)advisa.se Mobil: 070-720 45 81 Besöksadress: Kungsgatan 30, Stockholm www.advisa.se ––––––––––––––––––––––––––– Informationen i detta meddelande är konfidentiell och endast avsedd för den/de angivna mottagarna. Om Du inte är en avsedd mottagare, ber vi Dig att inte läsa, kopiera eller använda informationen i meddelandet samt att inte heller vidaresända informationen eller delge den till annan. Vi ber Dig även att omedelbart meddela avsändaren att Du erhållit meddelandet felaktigt samt att därefter genast radera det från Ditt system. Advisa ansvarar inte för eventuella personliga åsikter eller värderingar i detta meddelande.

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