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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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Concurrent RINGING calls limitation
by Kevin Mathy 17 Jun '14

17 Jun '14

Hi List, I'm trying to find something on freeswitch to make a "per-user" limitation of concurrent "RINGING" calls (or more generally, calls in PROCEEDING state). I've found the "max-proceeding" value I can set in switch.conf.xml, but it's a system's wide parameter, and I'd like more granularity... So, after some time spent on the mailing list and with my friend Google, it seems that someone asked for quite the same thing 2 years ago, but didn't get any answer : http://marc.info/?l=freeswitch-users&m=134375187528797 So, today, is there anything to achieve this ? Maybe something based on the "limit" module ? Thanks a lot, Kevin *Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP --

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 96, Issue 138
by Amin Dhamani Askhanuman 17 Jun '14

17 Jun '14

Thanks alot for your prompt reply. I will do myself. what configuration I need to change? Because provider is same for both inbound and outbound. On Tue, Jun 17, 2014 at 1:41 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: Need to deploy 5 inbound and 5 outbound (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 13:40:53 -0500 > Subject: Re: [Freeswitch-users] Need to deploy 5 inbound and 5 outbound > should be no different than when you added the first one, Did you do that > or did you hire someone? Are they all different sip accounts or one sip > account with multiple DID's? Also one question at a time please. > -- > 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 9:54 AM, Amin Dhamani Askhanuman <amin(a)askhanuman.com> > wrote: > > Hi, > > Currently we are using freeswitch with 1 inbound and 1 outbound number now > we want to add 5 more inbound and outbound numbers to existing > configuration and call from 5 inbound and outbound will route to certain > extensions. Can you guide me how I can do that. > > > > _______________________________________________ > 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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Core Dump when Calling Extension from WebRTC/SIP.js
by Lyle Pratt 17 Jun '14

17 Jun '14

I'm trying to do some testing with WebRTC support using OnSIP's SIP.js library. However, whenever I try to call a FreeSwitch extension, I get a core dump on 1.4.beta. I'm able to connect the website AND register, but the moment I do an INVITE, it crashes. Has anyone else encountered this? Console Log: 2014-06-17 01:48:13.856557 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1111(a)sip3.xyz.com [71d88070-f5c1-11e3-b884-638e8bd00f1c] 2014-06-17 01:48:13.856557 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/1111(a)sip3.xyz.com [BREAK] 2014-06-17 01:48:13.856557 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/1111(a)sip3.xyz.com [BREAK] 2014-06-17 01:48:13.856557 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1111(a)sip3.xyz.com) Running State Change CS_NEW 2014-06-17 01:48:13.856557 [DEBUG] sofia.c:8334 sofia/internal/1111(a)sip3.xyz.com receiving invite from 70.113.42.156:57858 version: 1.4.3 git 4904746 2014-05-19 17:38:31Z 64bit 2014-06-17 01:48:13.876638 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/1111(a)sip3.xyz.com [BREAK] 2014-06-17 01:48:13.876638 [DEBUG] sofia.c:2033 detaching session 71d88070-f5c1-11e3-b884-638e8bd00f1c 2014-06-17 01:48:13.876638 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/1111(a)sip3.xyz.com) State NEW 2014-06-17 01:48:13.996579 [DEBUG] sofia.c:2138 Re-attaching to session 71d88070-f5c1-11e3-b884-638e8bd00f1c 2014-06-17 01:48:13.996579 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/1111(a)sip3.xyz.com [BREAK] 2014-06-17 01:48:13.996579 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/1111(a)sip3.xyz.com [BREAK] 2014-06-17 01:48:14.016556 [DEBUG] sofia.c:8334 sofia/internal/1111(a)sip3.xyz.com receiving invite from 70.113.42.156:57858 version: 1.4.3 git 4904746 2014-05-19 17:38:31Z 64bit Floating point exception (core dumped)

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IPv6 Audit
by Brian West 16 Jun '14

16 Jun '14

We need the community to assist in modules that do not support IPv6, I've filed FS-6600 for everyone to comment on, Please do some research and help eradicate the last lingering modules that don't support IPv6. https://jira.freeswitch.org/browse/FS-6600 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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Missing modules message - FS Install with mod_freetdm
by bedgar＠vseinc.com 16 Jun '14

16 Jun '14

All, I am in the process of updating to the newest 1.2.stable build, commit 8e975d8fed7bc9ee39f5ddc9406ef36ec8f186e1, and have encountered a few messages at the end of "make install". I have not encountered these before. I am using RBS/CAS T1's and am sure that these will require updating. WARNING: installed module: ftmod_analog_em.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: ftmod_analog.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: ftmod_skel.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: ftmod_wanpipe.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: ftmod_zt.so was not installed by this build. It is not present in modules.conf. Can someone shed some light on this for me. Thank you, Brian

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mod_fifo documentation addition
by Alvaro Mari-Thompson 16 Jun '14

16 Jun '14

Hi all, Some of you may have some various posts by me requesting information on mod_fifo and mod_callcenter as we've been looking at introducing a very simple ring-all call queueing to our hunt groups. After delving through source code for mod_fifo I came across the outbound_per_cycle variable which determines how many concurrent members ring when a call comes in. This is something I've spent days and days trying to get working but couldn't figure out what the behaviour was. I can now calculate on the fly how many members a hunt group has, and set this variable on each fifo. I'd like to contribute to the documentation of mod_fifo by adding a small paragraph based on this variable. I tried registering on wiki.freeswitch.org but it tells me new accounts are locked as the documentation is now moving over to confluence.freeswitch.org, but that site doesn't have anything on mod_fifo yet. Will the documentation be moved across, or is it waiting for someone to write it all up? Regards, Alvaro --- Voicebooth Limited t: +44 (0)20 3475 3999 e: <mailto:info@voicebooth.com> info(a)voicebooth.com w: <http://www.voicebooth.com/> www.voicebooth.com Voicebooth specialises in Hosted Telephony Services and SIP Trunks. Voicebooth Limited is Registered in England and Wales with Registration Number 07758368.

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