October 2013 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] question regard sip-reregistrations/renewals in sofia_reg.c
by Michel Brabants 16 Oct '13

16 Oct '13

Hello, I'm looking through sofia_reg.c about how reregistrations are being handled. More specifically, at lines 1599 - 1663. Link to file: http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/endpointss/mod_… I would think that during a renewal of a registration, when multiple registrations aren't allowed, you would also get an update of the current registration. What happens now seems to be a removal of the current registration in the database and an insertion of the "renewal". Lines 1605-1607 and 1638-1648. Does this mean that when the user is being looked up from another profile at the moment that the deletion just happened and the "reinsertion" not yet, that the user can't be found and a "not available" would be generated or am I wrong? Wouldn't an update be better in this case (or even considered a fix) or would this break code that I don't know about currently? I'm willing to submit the patch ...., but I need to know whether this would break anything. Kind regards, Michel

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[Freeswitch-dev] FreeSWITCH news and Notes - FreeSWITCH 1.2.14 Released!
by Ken Rice 16 Oct '13

16 Oct '13

FreeSWITCHers! Today we are proud to announce FreeSWITCH 1.2.14! Available today via git, http://files.freeswitch.org/freeswitch-1.2.14.tar.bz2, and the deb and yum repos! This is a maintenance release to address several bugs that have been identified since the last release. Also dont forget ClueCon Weekly Conference Call! Every Wed at 1PM EST! For more information on how to join see: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instruct ions (What a coincidence! That¹s today!) -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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[Freeswitch-dev] PORTAUDIO issue ?
by Thomas Chen 14 Oct '13

14 Oct '13

the board i am using FREESWITCH has CS5535/CS5536 Audio from shell.... i can play a WAV sound file without problem # aplay xxx.wav however, in fs_cli / portaudio, i have problem playing the sound file fs_cli > pa play xxx.wav (kernel complains about dma problem).... any idea/ suggestion where to look ? is portaudio doing something different from alsa utility (aplay) ??? thanks

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[Freeswitch-dev] interdigit delays in RFC2833 DTMF
by Paul-Lists 14 Oct '13

14 Oct '13

Hi, Can anyone help me understand the dtmf inter digit delays using Freeswitch and rfc2833. Im calling switch_core_session_send_dtmf_string() to send a few digits to an endpoint for retransmission over a 2 wire connection. Im using a couple of different endpoints to regenerate the DTMF, nn OBI110 and a linksys ATA. In both cases I don't see much of an inter digit pause, I can change the length of the digits with dtmf duration settings on freeswitch, but not the inter digit gap (space). Is this a function of Freeswitch,, or the terminating hardware? Thanks in Advance. Paul C.

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[Freeswitch-dev] Friday Free For All Today, 2P Eastern until ???
by Ken Rice 11 Oct '13

11 Oct '13

FreeSWITCHers, Today, 2P Eastern, join us on 888 for the weekly FridayFreeForAll! Today¹s Topic? Who knows! Its the FreeForAll! Have a question? Join us and ask it, want to hang out and chat with the FreeSWITCH Devs or other FreeSWITCHers? Call in! Call sip:888@conference.freeswitch.org or visit http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions for more info on joining! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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[Freeswitch-dev] Fwd: SIP return code changed from 480 to 408
by Jeff Ullman 10 Oct '13

10 Oct '13

Hi All: I realized during originate command SIP return code 480 is set to 408 by FS. Wireshark capture shows 480 from carrier and capture is attached. Pastebin for FS log : http://pastebin.freeswitch.org/21485 I'm using stable version of FS 1.2.3 Can someone please take a look ? Thanks

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[Freeswitch-dev] Weekly News Notes Etc
by Ken Rice 09 Oct '13

09 Oct '13

Hey Guys! Don¹t forget to join us today for ClueCon weekly on 888 at 1P Eastern or 10A Pacific! If you haven¹t joined the Groups on Google+, Follow FreeSWITCH here https://plus.google.com/communities/110013892794022037793 Follow ClueCon here https://plus.google.com/communities/108955134544785139154 And don¹t forget to follow us on Twitter @FreeSWITCH_wire -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch @Ken4VoIP via Twitter

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[Freeswitch-dev] Bitcoin donations?
by Sam Russell 08 Oct '13

08 Oct '13

I just noticed the paypal donations text in the startup screen, has anyone (i.e. Anthony et al.) considered this? The dudes on /r/bitcoin on reddit are generally keen to help out anyone who starts accepting bitcoin donations (so you'd likely get a bit of cash there), and someone like BitPay or Coinbase could front-end it and do the conversion to USD. Good idea? Or a bit out-there or too much work?

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[Freeswitch-dev] Freeswitch rejecting streams when a=rtpmap:96 /0
by sangdrax8 08 Oct '13

08 Oct '13

It seems that the current head has actually started producing this "rtpmap: 96 /0", when my client devices do not include it in the sdp. This causes problems when I am trying to upgrade one switch at a time in my production environment. I assume (will test here shortly) that if all switches are on the newer version, this wouldn't be an issue as the indicated patch should handle it. I would really like to be able to upgrade my boxes individually, but if an upgraded box add's this to the SDP, my non-upgraded boxes will kill the call. The following is an example SDP trace from my latest head (10.01.2013) showing the SDP from the client device is normal, and then what is sent to the other freeswitch (my currently non upgraded one) includes this problem causing rtpmap. I should note that this seems to only be the case when I activate zrtp on the clients, which causes my dial plan to set proxy_media=true. I haven't opened a bug in jira, I wanted to confirm if this rtpmap is in fact a bug. If there is a good reason it is there now and isn't in my older freeswitch versions, then I may have to attempt to upgrade all at one time. recv 1072 bytes from tls/[1.1.1.1]:50272 at 13:42:16.509540: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK5N858D7BX46je Contact: <sip:15715550001@192.168.10.136:60068;transport=tls> From: "Test 1" <sip:17035550001@securevoice>;tag=jSmQZ67etF8pa Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a CSeq: 50067799 INVITE To: <sip:15715550001@192.168.10.136:60068 ;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY, MESSAGE Supported: replaces, path Content-Type: application/sdp Content-Length: 534 v=0 o=- 98627 19401 IN IP4 192.168.10.136 s=menduco c=IN IP4 192.168.10.136 t=0 0 m=audio 58620 RTP/SAVP 9 101 a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi a=ptime:30 a=fmtp:101 0-15 a=zrtp-hash:1.10 99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0 a=sendrecv m=audio 0 RTP/AVP 9 0 101 a=rtpmap:101 TELEPHONE-EVENT/8000 a=ptime:30 a=fmtp:101 0-15 a=zrtp-hash:1.10 18489BD8B5526F4C92AB6EB2A04022C25E8CB5D7E58D4E633FC4545783447F8C ------------------------------------------------------------------------ send 472 bytes to tls/[1.1.1.1]:50272 at 13:42:16.510594: ------------------------------------------------------------------------ ACK sip:15715550001@192.168.10.136:60068;transport=tls SIP/2.0 Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK6y1ya9QFtDX5S Max-Forwards: 70 From: "Test 1" <sip:17035550001@securevoice>;tag=jSmQZ67etF8pa To: <sip:15715550001@192.168.10.136:60068 ;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a CSeq: 50067799 ACK Contact: <sip:mod_sofia@2.2.2.2:5061;transport=tls> Content-Length: 0 ------------------------------------------------------------------------ send 1516 bytes to tls/[10.110.11.31]:47793 at 13:42:16.516187: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 10.110.11.31:5071 ;branch=z9hG4bKD97QXp3XZgX4a;rport=47793 From: "Test 1" <sip:17035550001@10.110.11.31>;tag=DU0S42e40FNpg To: <sip:15715550001@dcprotopop1.private.sec :5071;transport=tls>;tag=tcXjHQ6gme91S Call-ID: 6fd5bd89-a6d4-1231-8291-005056ac002c CSeq: 50067799 INVITE Contact: <sip:15715550001@10.110.11.32:5071;transport=tls> User-Agent: FreeSWITCH-mod_sofia/1.5.6b+git~20131001T160636Z~6d2280df08 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 408 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:15715550001@dcprotopop1.private.sec X-FS-Support: update_display,send_info Remote-Party-ID: "Outbound Call" <sip:15715550001@dcprotopop1.private.sec >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1167543025 1167543026 IN IP4 10.110.11.32 s=FreeSWITCH c=IN IP4 10.110.11.32 t=0 0 m=audio 27868 RTP/SAVP 9 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi a=ptime:30 a=zrtp-hash:1.10 99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0 m=audio 0 RTP/AVP 9 0 96 a=rtpmap:96 /0

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[Freeswitch-dev] sleep()
by Paul-Lists 06 Oct '13

06 Oct '13

Hi everyone, I'm developing a bespoke app for freeswitch that communicates via modem tones and DTMF with remote devices. All is going well and Freeswitch seems an ideal development platform. I am struggling with some of the bigger concepts, like threads and scheduling. My dial plan, answers the call and calls my app where I am attaching a media bug from my main app entry point. the media bug gets the rtp packets and happily decodes them until it has received a full block of data. I have implemented my own event queue and timer processes, and my main loop reads from its queue, does the work, executes a switch_sleep() then goes round the loop until the incoming phone call ends My problem is that when the sleep ends, there is some kind of a glitch and I lose an RTP packet, containing a few bytes of modem data. If I make the sleep last for 60 seconds, I get stacks of modem data and it all decodes fine without a missing bit, just the act of waking, then going back to sleep causes the problem. Is there a better way to suspend my app for a second or less while I'm decoding data in the background, I don't want to hog the processor. Any pointers would be much appreciated. Best regards, Paul Collister

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