March 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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park_after_bridge does not work if far end does 180 Ringing
by Ali Pey 17 Mar '15

17 Mar '15

Hello, I see a different behaviour for 183 Ringing vs 180 Ringing and a 408 Request Timeout response message with regards to park_after_bridge: 1) I have "park_after_bridge=true". 2) Do a bridge to a PSTN number (SIP Trunk) 3) After sometime freeswitch receives 408 Request Timeout message: a) if carrier had sent 183 Ringing (in band ring back tone) the a-leg is parked after the 408 timeout message. b) if carrier had sent 180 Ringing, the aleg is hungup after timeout (408 message) Is this a bug? Is there a work around? Thanks, Ali Pey

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delay before ringback at 183 to 180 transition
by Jeff Pyle 17 Mar '15

17 Mar '15

Hello, FreeSWITCH Version 1.5.15b+git~20150313T215013Z~9872c52c57~64bit (git 9872c52 2015-03-13 21:50:13Z 64bit) I have a situation where a gateway sends a 183 with media, then a 180. Following Anthony's advice from FS-3859 I have the following: <extension name="sendit"> <condition field="destination_number" expression="(.*)"> <action application="set" data="ringback=/usr/local/freeswitch/sounds/ringtester.wav"/> <action application="bridge" data="{bridge_early_media=true}sofia/public/$1@${host2}"/> </condition> </extension> On the a-leg I hear media from the 183 until host2 stops it and sends the 180. Then there's a 4.8 second delay of no RTP from FS before I hear my ringtester.wav file. This occurs with a tone stream as well. The last event on the console is "Callstate Change EARLY -> RINGING" when the 180 comes in, then "nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering" when the 180 arrives. Then 4.8s of nothing. Then FS resumes the RTP with the ringback file. There is nothing new displayed when the RTP resumes with the ringback file. Is this a bug, or a misconfiguration? - Jeff

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Port changes in multiple 183s cause no audio after 200
by Jeff Pyle 17 Mar '15

17 Mar '15

Hello, The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit). The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with. The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example: - Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183). In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio. I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome! - Jeff

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