October 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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CDR for non authenticated call
by Markus Bönke 20 Oct '16

20 Oct '16

Hello, I have a question if it is correct that freeswitch opens a channel and generates a CDR for the following call scenario. The call is initiated from registered SIP client: INVITE 100 Trying BYE 407 Proxy Authentication Required 200 OK The generated CDR has only the uuid and hangup_cause set, all other fields like destination_number, callerid, etc are null. Is this behavior as expected? Thanks and regards Markus

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LDAP authentication?
by Kevin Long 19 Oct '16

19 Oct '16

Is it possible to authenticate endpoints to freeswitch , and if so, does the LDAP module support doing so securely (with TLS connection between freeswitch and the LDAP server) I was checking this issue but whether STARTLES or LDAPS was ever implemented is unclear to me: https://freeswitch.org/jira/browse/FS-4870 <https://freeswitch.org/jira/browse/FS-4870> Thanks, Kevin Long

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Achieving streaming ASR
by Luca Pradovera 19 Oct '16

19 Oct '16

Hello, following a discussion on IRC, I thought I would move this here so it is more async. The need is to be able to use Nuance/IBM Watson/Google Speech APIs in streaming mode for real-time transcription (essentially, captioning a video conference and getting a transcript of the proceedings). My first idea was to simply get the frames from a recording as it is being written, though there might be some obstacles there. It was suggested the media bugs infrastructure would work, at the C level, though I need to check which formats it supports. Do you have any other suggestions? Thank you! Luca

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