March 2015 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Silence in conference recording
by miguelo sana 06 May '15

06 May '15

Hi, I use the recording functionality of mod_conference to record conference calls. The recording is save to a .wav file. Now for the second time in a few months I experienced something strange. The recording goes silent after 15 minutes of recording. But the recording is not stopped. There is nothing strange found in logging. And other recordings are not having this issue. Is there anyone who knows anything about this and how to fix or prevent this. Thanks. Miguelo

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[Freeswitch-dev] FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 27 Mar '15

27 Mar '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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[Freeswitch-dev] FreeSwitch INVITE Issue with absolute_codec_string for G723.
by Dhrupalkumar Tilava 21 Mar '15

21 Mar '15

Hi All, I am doing the call from the A Endpoint to B Endpoint using FreeSwitch In Between When FreeSwitch Get the INVITE Request from the A Endpoint, I am using the following absolute_codec_string. absolute_codec_string = "PCMU,PCMA,G726-40,G726-32,G726-24,G726-16,G729,GSM,L16,ILBC,G723" FreeSwitch is also sending the INVITE Request to B Leg with the codecs defining in the absolute_codec_string. Which is as below. 592f8a31-4978-1233-90b2-d4ae52cde95f] send 1291 bytes to udp/[192.168.14.68]:5060 at 07:48:32.929086: ------------------------------------------------------------------------ [592f8a31-4978-1233-90b2-d4ae52cde95f] INVITE sip:12191121212@192.168.14.68:5060 SIP/2.0 [592f8a31-4978-1233-90b2-d4ae52cde95f] Via: SIP/2.0/UDP 192.168.2.165;rport;branch=z9hG4bK4mtN7XDy2Xg7B [592f8a31-4978-1233-90b2-d4ae52cde95f] Max-Forwards: 68 [592f8a31-4978-1233-90b2-d4ae52cde95f] From: "dhrupla" <sip:14191121212@192.168.2.165>;tag=4cHc3S7N80tpm [592f8a31-4978-1233-90b2-d4ae52cde95f] To: <sip:12191121212@192.168.14.68:5060> [592f8a31-4978-1233-90b2-d4ae52cde95f] Call-ID: 592f8a31-4978-1233-90b2-d4ae52cde95f [592f8a31-4978-1233-90b2-d4ae52cde95f] CSeq: 73082792 INVITE [592f8a31-4978-1233-90b2-d4ae52cde95f] Contact: <sip:iMAX_SS@192.168.2.165:5060> [592f8a31-4978-1233-90b2-d4ae52cde95f] User-Agent: iMax_SS [592f8a31-4978-1233-90b2-d4ae52cde95f] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY [592f8a31-4978-1233-90b2-d4ae52cde95f] Supported: precondition, path, replaces [592f8a31-4978-1233-90b2-d4ae52cde95f] Allow-Events: talk, hold, conference, refer [592f8a31-4978-1233-90b2-d4ae52cde95f] Content-Type: application/sdp [592f8a31-4978-1233-90b2-d4ae52cde95f] Content-Disposition: session [592f8a31-4978-1233-90b2-d4ae52cde95f] Content-Length: 653 [592f8a31-4978-1233-90b2-d4ae52cde95f] v=0 [592f8a31-4978-1233-90b2-d4ae52cde95f] o=iMax_SS 1426824519 1426824521 IN IP4 192.168.2.165 [592f8a31-4978-1233-90b2-d4ae52cde95f] s=iMax_SS [592f8a31-4978-1233-90b2-d4ae52cde95f] c=IN IP4 192.168.2.212 [592f8a31-4978-1233-90b2-d4ae52cde95f] t=0 0 [592f8a31-4978-1233-90b2-d4ae52cde95f] m=audio 13193 RTP/AVP 0 8 98 99 100 102 18 3 103 104 101 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:0 PCMU/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:8 PCMA/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:98 G726-40/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:99 G726-32/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:100 G726-24/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:102 G726-16/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:18 G729/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:3 GSM/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:103 L16/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:104 iLBC/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=fmtp:104 mode=30 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:101 telephone-event/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=fmtp:101 0-16 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=silenceSupp:off - - - - [592f8a31-4978-1233-90b2-d4ae52cde95f] a=ptime:20 [592f8a31-4978-1233-90b2-d4ae52cde95f] m=audio 13193 RTP/AVP 4 101 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:4 G723/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:101 telephone-event/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=fmtp:101 0-16 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=silenceSupp:off - - - - [592f8a31-4978-1233-90b2-d4ae52cde95f] a=ptime:30 But Here Strange Behaviour is with G723 Codec. As you can see in SDP there is a separate "m=" Field for the G723 Codec.This SDP is wrong. But when I am using the following absolute_codec_string, I am getting the proper INVITE with proper SDP. (Removing G723 Codec) absolute_codec_string = "PCMU,PCMA,G726-40,G726-32,G726-24,G726-16,G729,GSM,L16,ILBC" 592f8a31-4978-1233-90b2-d4ae52cde95f] send 1291 bytes to udp/[192.168.14.68]:5060 at 07:48:32.929086: ------------------------------------------------------------------------ [592f8a31-4978-1233-90b2-d4ae52cde95f] INVITE sip:12191121212@192.168.14.68:5060 SIP/2.0 [592f8a31-4978-1233-90b2-d4ae52cde95f] Via: SIP/2.0/UDP 192.168.2.165;rport;branch=z9hG4bK4mtN7XDy2Xg7B [592f8a31-4978-1233-90b2-d4ae52cde95f] Max-Forwards: 68 [592f8a31-4978-1233-90b2-d4ae52cde95f] From: "dhrupla" <sip:14191121212@192.168.2.165>;tag=4cHc3S7N80tpm [592f8a31-4978-1233-90b2-d4ae52cde95f] To: <sip:12191121212@192.168.14.68:5060> [592f8a31-4978-1233-90b2-d4ae52cde95f] Call-ID: 592f8a31-4978-1233-90b2-d4ae52cde95f [592f8a31-4978-1233-90b2-d4ae52cde95f] CSeq: 73082792 INVITE [592f8a31-4978-1233-90b2-d4ae52cde95f] Contact: <sip:iMAX_SS@192.168.2.165:5060> [592f8a31-4978-1233-90b2-d4ae52cde95f] User-Agent: iMax_SS [592f8a31-4978-1233-90b2-d4ae52cde95f] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY [592f8a31-4978-1233-90b2-d4ae52cde95f] Supported: precondition, path, replaces [592f8a31-4978-1233-90b2-d4ae52cde95f] Allow-Events: talk, hold, conference, refer [592f8a31-4978-1233-90b2-d4ae52cde95f] Content-Type: application/sdp [592f8a31-4978-1233-90b2-d4ae52cde95f] Content-Disposition: session [592f8a31-4978-1233-90b2-d4ae52cde95f] Content-Length: 653 [592f8a31-4978-1233-90b2-d4ae52cde95f] v=0 [592f8a31-4978-1233-90b2-d4ae52cde95f] o=iMax_SS 1426824519 1426824521 IN IP4 192.168.2.165 [592f8a31-4978-1233-90b2-d4ae52cde95f] s=iMax_SS [592f8a31-4978-1233-90b2-d4ae52cde95f] c=IN IP4 192.168.2.212 [592f8a31-4978-1233-90b2-d4ae52cde95f] t=0 0 [592f8a31-4978-1233-90b2-d4ae52cde95f] m=audio 13193 RTP/AVP 0 8 98 99 100 102 18 3 103 104 101 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:0 PCMU/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:8 PCMA/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:98 G726-40/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:99 G726-32/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:100 G726-24/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:102 G726-16/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:18 G729/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:3 GSM/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:103 L16/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:104 iLBC/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=fmtp:104 mode=30 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=rtpmap:101 telephone-event/8000 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=fmtp:101 0-16 [592f8a31-4978-1233-90b2-d4ae52cde95f] a=silenceSupp:off - - - - [592f8a31-4978-1233-90b2-d4ae52cde95f] a=ptime:20 So, I am facing the problem when I am using the G723 Codec in the absolute_codec_string. Please help me to resolve this issue. I am using the freeswitch in the Proxy Mode. Thank you for your kind help Regards. Dhrupal Tilava

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[Freeswitch-dev] FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 20 Mar '15

20 Mar '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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[Freeswitch-dev] Using Freeswitch with TLS
by Yücel ALTUNAY 13 Mar '15

13 Mar '15

Hi, I want to use Freeswitch with TLS and i want to learn which modules i need mandatory. i want to remove modules the i dont need. can you list modules here or write any web page link? Thank you.

2 1

[Freeswitch-dev] FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 13 Mar '15

13 Mar '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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[Freeswitch-dev] Bare Minimum Freeswitch
by Jacob Hodges 09 Mar '15

09 Mar '15

Hi all, I'm quite new to freeswitch and have used it in the past to establish a SIP Trunk between Lync and another SIP provider. I'm now looking at Freeswitch as a potential replacement for a legacy PBX at my local church. I'm extremely wary of the default Freeswitch configuration as it seem to include a lot of functionality that I don't need and a lot configuration I don't understand. All I need is the following: - 5 extensions that can dial each other and the PSTN - Ability to do group pickup, paging and voicemail to email - Ability to set day/night mode - Ability to use SPA3102 as a gateway to the PSTN. Can anyone tell me the minimum number of modules / configuration that is needed to accomplish this? Thanks, Jacob.

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[Freeswitch-dev] mod_rayo patching
by ik 08 Mar '15

08 Mar '15

Hello, There is a bug in mod_rayo that I wish to patch myself, however I have never touched any sources of FS before, not to mention rayo itself (and it's been many years since I touched C). Before I'll start to figure out the proper source code to touch, is there a good guide to read about FS development that can help in the matter ? Thanks Ido

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[Freeswitch-dev] FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 06 Mar '15

06 Mar '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

1 0