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Disable Re-INVITE on REFER ?
by Patrick Shea 19 Mar '15

19 Mar '15

I am seeing 2 problems that I can't seem to find the answers to. I've scoured the Freeswitch Wiki (old), Confluence, and the users list. I have 2 SIP calls bridge through Freeswitch - they are CONTROL channels, so no media is negotiated. A Leg is SIP over Websockets, B Leg is SIP. SIPjs (WS) <---- A Leg ---> Freeswitch <---- B Leg ---> EP The problem I have is in some cases I get a REFER from the EP to transfer the call somewhere else, and this results in a Re-INVITE from Freeswitch to the SIPjs client. (Ideally, I'd like to "skip" this re-Invite and just bridge the A-leg to the new AP). Problem 1. Freeswitch sends media in the SDP of the Re-INVITE. This doesn't look right to me, but I assume using some local codecs specified somewhere. m=audio 28692 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 Can I disable these and just send the original m= line received for the initial call? m=application 28118 udp as Problem 2. SIPjs doesn't support Re-INVITE, yet, so it responds with 488 Unacceptable Here. At this point, Freeswitch, drops the A leg and the transfer fails. I can't find a solution to work around this - any ideas? Like I said, Ideally I'd like to skip the re-INVITE altogether. Currently debugging SOFIA to see how this can be done - is there a parameter already somewhere that I am overlooking? Thanks Patrick

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Keep A Leg alive after bridge fails
by Grant Bagdasarian 19 Mar '15

19 Mar '15

Hello, I'm using outbound ESL socket to control calls and one of the actions is to bridge the A leg to another destination. This works great when the remote party answers, but when the remote party does not answer or is busy the bridge fails and disconnects the A leg. I wish to keep the A leg alive. I've tried setting different kinds of channel variables on the a leg and even the b leg, but without success. There is a thread from a few years ago discussing this very same issue (although they were talking about inbound socket), so I was wondering if anyone has a solution for this? Continue_on_fail, hangup_after_bridge, park_after_bridge, ignore_early_media, all don't work. Any ideas?

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Simple PBX tutorial
by Stanislav Sinyagin 19 Mar '15

19 Mar '15

hi, Here's a short article that I wrote to help people start using FreeSWITCH: https://github.com/voxserv/freeswitch_conf_minimal/blob/master/docs/tutoria… your feedback will be appreciated.

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mod_callcenter custom headers
by Ali Jibran 18 Mar '15

18 Mar '15

Hi all. I wanted to know if it was possible to pass/export variables or custom headers into mod callcenter. Like if I pass a variable in A-Leg I can retrieve it in the B-Leg whenever an agent picks up. Is this possible?

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sofia recovery, A leg is being hung up with INCOMPATIBLE_DESTINATION
by Joseph Dickson 18 Mar '15

18 Mar '15

Greetings! I'm working on putting together an HA FreeSwitch setup, and I'm having trouble getting sofia recovery to work. I've got to the point where call state is shared successfully (master-master MySQL at the moment), and where the new node attempts to recover the calls. The problem seems to be in the A leg re-invite. I've tried a few different A-leg user agents, and get the same issue with all... FS hangs up the leg with INCOMPATIBLE_DESTINATION after sending an ACK to the A leg's OK. The SDP looks fine to me, and I don't see any error messages about codec selection etc.. I turned up logging as far as I knew how (I'm still pretty new to FS, so if there's a better log method, let me know!).. The output is on pastebin here: http://pastebin.com/WK8A7Kx7 This log snippet is of the log messages generated after a sofia recover on my secondary node (being issued by pacemaker). There was a single call connected at the time of the failover.. Any ideas? I've seen a couple JIRA entries and mailing list entries for things that are similar, but usually the resolution was to update to the newest FS.. I'm running the latest 1.4 from the freeswitch debian repo.. The A leg client from this log trace is Zoiper Free. version info: # freeswitch -version FreeSWITCH version: 1.4.15-1~64bit (-1 64bit) Any pointers would be appreciated! Joe

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g729 installer 404
by Peter Hartmann 18 Mar '15

18 Mar '15

Hey, There seems to be a hosting problem with g729 installer. It's located at: http://files.freeswitch.org/g729/fs-201501231218-installer However, when one attempts to build FS from source (1.5.15b+git~20150317T232708Z~42961b7f2e~64bit and also the 1.4 version) the scripts are attempting to retrieve it from: http://files.freeswitch.org/g729/fsg729-201501231218-installer resulting in a 404. Note the different filename prefix. ----- cd /usr/src/freeswitch/libs && /bin/bash /usr/src/freeswitch/build/getg729.sh fsg729-201501231218-installer --2015-03-18 12:31:47-- http://files.freeswitch.org/g729/fsg729-201501231218-installer Resolving files.freeswitch.org (files.freeswitch.org) 209.105.235.7, 2607:f348:1021::7 Connecting to files.freeswitch.org (files.freeswitch.org)|209.105.235.7|:80... connected. HTTP request sent, awaiting response... 404 Not Found 2015-03-18 12:31:47 ERROR 404: Not Found. ------- I'd file a JIRA but I don't think there's anything wrong with the code. 'fsg729' seems to be a more descriptive filename prefix as compared to 'fs'. Sed got me through the build, but it seems something that should be fixed, no? And if 'fs' is the preferred filename prefix I'd be happy to file a report. Thanks much, Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com (212)203-8870 If I can't explain it to you in plain language, that means I don't understand it.

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ASR and PocketSphinx
by Guillermo Ruiz Camauer 18 Mar '15

18 Mar '15

I am looking to experiment with some simple ASR. Small vocabulary (yes, No, one, two.. zero, repeat). I found mod_pocketsphinx in the old WIKI. I am wondering if this module is being maintained or if it has fallen into disuse. If the latter, can anyone share their experiences with other FREE ASRs? Are there any free still out there or must we go the commercial route? By the way, I am looking for both English and Spanish ASR. Thanks, -- Guillermo Ruiz Camauer

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Custom Headers with Sip.js
by Ali Jibran 18 Mar '15

18 Mar '15

Hi all. I make a call to FS callcenter using webRTC. Now I would like to add some additional data to the header. That I did by using extraheaders. Now all that is good. I can see the custom header in the sip trace on freeswitch. For e.g in sip.js I add: var options = { extraHeaders: [ 'a: foo' ] }; on FS i get: INVITE sip:900@anonymous.invalid SIP/2.0 Via: SIP/2.0/WS 9jscv0q24s4m.invalid;branch=z9hG4bK1002472 Max-Forwards: 70 To: <sip:900@anonymous.invalid> From: <sip:anonymous.hfsqa4@anonymous.invalid>;tag=depmif1fo5 Call-ID: l5pieavpjguoset7ugvd CSeq: 1458 INVITE * a: foo* Contact: <sip:8rgi365g@9jscv0q24s4m.invalid;transport=ws;ob> Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Contact: <sip:8rgi365g@9jscv0q24s4m.invalid;transport=ws;ob> Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: outbound User-Agent: SIP.js/0.5.0 Content-Length: 3636 How do I access it in the dialplan? I tried ${sip_h_X-a} but it didnt't work. Help please?

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FS default behaiour when no ACK is received
by Abdul Hakeem 18 Mar '15

18 Mar '15

Hello, Is there any way to override FS behaviour when no ACK is received by FS from registered clients? I'm hoping there might be a way for FS to continue processing rather than end the call. Cheers, Abdul Hakeem

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Channel Variable: variable_pre_transfer_caller_id_number
by Paul Atreides 18 Mar '15

18 Mar '15

Hi, <action application="info"/> when I call the info app in my dial plan then I can see the following vars in my log variable_pre_transfer_caller_id_name: [] variable_pre_transfer_caller_id_number: [18] But when I try to access one single channel variable its always empty. Why? ${variable_pre_transfer_caller_id_number} Thank you for your help.

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