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Freeswitch logs
by Gonzalo Gasca Meza 23 Oct '16

23 Oct '16

Hi all, Is there a tool to process Freeswitch SIP logs?, I needed to create a Python script to parse SIP logs (freeswitch.log) and extract SIP BYE reasons (My provider was sending invalid BYE reasons and our CDR tool was processing call with Error versus Normal Call clearing). I already capture SIP logs with Network Module and Homer <https://github.com/sipcapture/homer/wiki/Examples%3A-FreeSwitch>, but wondering if there is such tool for Freeswitch already. Thank you -Gonzalo

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Some ringtones
by Emrah 22 Oct '16

22 Oct '16

Hi, I made some ringtones (first ones are ugly) and if you need them they are here. https://clearer.link/ringtones/ We should have an FS ringtone. Sort of something to use every time we have FS in the mix, maybe like a sound logo?

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Re: [Freeswitch-users] Distributed conference
by Alexandru Covalschi 22 Oct '16

22 Oct '16

Hello, Yeah, I got the idea, no more questions. Will try to implement it soon, thanks for your advice! Cheers, Alex -- Alexandru Covalschi VoIP engineer and system administrator tel: +37367398493

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does mod_http_cache support AWS Signature Version 4?
by Sun Harold 22 Oct '16

22 Oct '16

Hello, I’m working on a setup to use freeswitch as media server together with kamailio and rtpengine. We need to record the calls with freeswitch and upload the recording to S3. I found mod_http_cache is able to do exactly what we need. But when I configured this module and use it in the dailplan, the recordings are not uploaded in S3. I can see HTTP 403 errors in freeswitch logs. Since I’m using aws China region, AWS signature version 4 is only option here. I did a quick scan on the source code and found mod_http_cache seems only support aws signature version 2. Does anyone can confirm if mod_http_cache work with aws signature version 4? If not, I will try to implement that and contribute back. Harold Sun

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Re: [Freeswitch-users] Distributed conference
by Alexandru Covalschi 21 Oct '16

21 Oct '16

First of all thanks for your answers guys! I'm aiming to audio conf. Emrah, last few questions - which method do you use for conference merging? Conference bridge? I mean - I want to allow users to send commands over DTMF. In that case I suppose I must catch DTMF and execute conference commands on another layer using ESL. Maybe you know another concept? Also I don't really understand - do I really must use FS as SBC- due to some conference layer may be there - or can I use Kamailio (as i'm used to that very much)? Thanks again! -- Alexandru Covalschi VoIP engineer and system administrator tel: +37367398493

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Answer to SIP OPTIONS with RTP Port 0 in SDP
by Dominique Jeannerod 21 Oct '16

21 Oct '16

Hello, i'm facing a strange issue on a trunk (on Internet). Here is the summary of the call flow : 1. The call is established normally (RTP Ports, Codec). Everything is fine. 2. During the call, Freeswitch receives a SIP OPTIONS with the SAME callid as the ongoing call. 3. Freeswitch answers 200 OK, but with a SDP with a 0 RTP port. 4. Our facing SIP Proxy uses 0 for next RTP trafic and then the communication continues with no sound Is this a normal behaviour from Freeswitch ? Do I have a possible configuration item to tune for that ? Here is the SIP OPTIONS part of the SIP trace : ------------------------------------------------------------------------ recv 436 bytes from udp/[PUBLIC_SOURCE_IP]:5060 at 17:20:58.284455: ------------------------------------------------------------------------ OPTIONS sip:+3310129144152000@10.9.16.171:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP PUBLIC_SOURCE_IP:5060;branch=z9hG4bK72be.d4033057.0 To: <sip:+3310129144152000@domain.net;user=phone>;tag=2tHZ5Xgv3HB8c From: <sip:+33624431415@domain.net ;user=phone>;tag=SD66qua01-7e4e5aa0-0019-088d-0000-0000 CSeq: 2 OPTIONS Call-ID: SD66qua01-64e828d3c9030b4024d0e46628fb2949-0000000 Max-Forwards: 70 Content-Length: 0 User-Agent: Keyyo-S5-P ------------------------------------------------------------------------ send 905 bytes to udp/[PUBLIC_SOURCE_IP]:5060 at 17:20:58.289599: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP PUBLIC_SOURCE_IP:5060;branch=z9hG4bK72be.d4033057.0 From: <sip:+33624431415@domain.net ;user=phone>;tag=SD66qua01-7e4e5aa0-0019-088d-0000-0000 To: <sip:+3310129144152000@domain.net;user=phone>;tag=2tHZ5Xgv3HB8c Call-ID: SD66qua01-64e828d3c9030b4024d0e46628fb2949-0000000 CSeq: 2 OPTIONS Contact: <sip:+3310129144152000@FREESWITCH_PUBLIC_IP:5060;transport=udp> User-Agent: FreeSWITCH Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Length: 256 v=0 o=FreeSWITCH 1476957466 1476957467 IN IP4 FREESWITCH_PUBLIC_IP s=FreeSWITCH c=IN IP4 FREESWITCH_PUBLIC_IP t=0 0 m=audio* 0* RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp:19369 IN IP4 FREESWITCH_PUBLIC_IP ------------------------------------------------------------------------ *Dominique Jeannerod* *Interact-iv.com* Mobile: +33 698 778 763 Headquarter : 600 rue Felix Trombe – tecnosud – 66100 Perpignan - France France : 0826 10 23 21 (0,15€/mn) International +33 (0)4 20 10 45 45

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fusionpbx
by Valter Nogueira 21 Oct '16

21 Oct '16

Sorry if it is OT. I am trying to figure out how fusionpbx stores freeswitch configuration. freeswitch global_getvar conf_dir shows /etc/freeswitch But when I look at /etc/freeswitch/sip_profiles/external, I don't see any gateway declared, altough when I do: sofia status I get: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external profile sip:mod_sofia@192.168.2.13:5080 RUNNING (0) external::d7369e36-c636-483c-b50b-177cef15caf9 gateway sip:50000@192.168.2.108 REGED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@192.168.2.13:5060 RUNNING (0) ================================================================================================= So, where is external::d7369e36-c636-483c-b50b-177cef15caf9 gateway declared? I thought it could be via mod_xml_curl, but there is no config for it in autoload_configs. Regards,

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Distributed conference
by Alexandru Covalschi 21 Oct '16

21 Oct '16

Hello list, Wanted to ask for hints. Let's imagine we have a very big conference, so it doesn't fit in one Freeswitch box. I need to somehow populate users on other box and merge the conferences. For now the only way I see is conference call out, but that's really complicated as soon as we need to synchronize a lot of stuff (conference master commands, members list and so on) using event socket and a completely new API layer. Is there any other way to get a bigger distributed conference? Maybe there is already an existing solution? Regards, Alex -- Alexandru Covalschi VoIP engineer and system administrator tel: +37367398493

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Conference Use-Case Question
by Cody Rosenbloom 21 Oct '16

21 Oct '16

Hi, I'm quite new to mod_conference and I've had good success implementing Stanislav Sinyagin's conferencing solution <https://txlab.wordpress.com/2012/09/17/setting-up-a-conference-bridge-with-…> on his blog for our freeswitch setup. This is fantastic for planned conference calls where everyone can dial into the conference line. I'm now looking to address the use case where 1 person is talking to another and suddenly needs to pull in one or two more people into the call to discuss (sort of a dynamic conferencing solution). Is this possible and what's the best way to go about achieving it? Appreciate any advice you can give me. Thanks Cody

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FP FILE ERR on Verto call
by Cj B 20 Oct '16

20 Oct '16

Hi All, I’m struggling a bit getting verto to work. I’m able log into verto but when I go to make a call there’s this error (FP File err): 2016-10-20 11:41:45.979170 [DEBUG] switch_core_media.c:3425 Save audio Candidate cid: 1 proto: udp type: srflx addr: 209.237.88.75:59551 2016-10-20 11:41:45.979170 [DEBUG] switch_core_media.c:3425 Save audio Candidate cid: 2 proto: udp type: srflx addr: 209.237.88.75:50413 2016-10-20 11:41:45.979170 [DEBUG] switch_core_media.c:3425 Save audio Candidate cid: 1 proto: udp type: srflx addr: 209.237.88.75:52747 2016-10-20 11:41:45.979170 [DEBUG] switch_core_media.c:3425 Save audio Candidate cid: 2 proto: udp type: srflx addr: 209.237.88.75:65293 2016-10-20 11:41:45.979170 [ERR] switch_core_cert.c:209 FP FILE ERR! 2016-10-20 11:41:45.979170 [DEBUG] switch_core_media.c:3467 Searching for rtp candidate. 2016-10-20 11:41:45.979170 [DEBUG] switch_core_media.c:3472 Choose rtp candidate, index 2, 209.237.88.75:59551 2016-10-20 11:41:45.979170 [DEBUG] switch_core_media.c:3250 verto.rtc/3000 choosing family v4 Because it’s switch_core_cert I’m assuming it’s a certificate error, but I can’t figure out what file exactly. I’ve got wss.pem and dtls-srtp.crt in the proper directories and if I do openssl s_client -connect my_ip:5061 and openssl s_client -connect my_ip:8082 I get the proper certificate information back. Any suggestions? Thanks for any help! Cj B

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