March 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Call drop on transfer (REFER) after SIP 302 MOVED
by Piccinin Francesco 23 Mar '17

23 Mar '17

Hi all, we are experiencing a strange issue during call transfer (REFER) after a SIP 302 MOVED. Please find below a brief description of our test environment. Extensions involved: - 8911 (original caller) - 8910 (forwarded to 5536 through phone ui) - 5536 (originator of call transfer) - 5532 (destination of call transfer) Step to reproduce the issue: - 8911 calls 8910 which is forwarded on 5536 (call fwd is configured directly on phone gui) - 5536 correctly answer the call from 8911 - 5536 begins a consult transfer to 5532: Hold step is ok but call drops when 5536 completes the transfer. >From the SIP signaling point of view: - 8910 answers to 8911 with a SIP MOVED. - Freeswitch send an invite to 5536 as 8911. - Call transfer from 5536 to 5532 is managed with a refer. - When 5536 complete the transfer, Freeswitch send a bye first to 8911 (original caller) and then to 5536. On Freeswitch dialplan we manage refer on a different context, setting "sip_redirect_context" variables. Without MOVED involved, all call transfer are correctly managed. It seems that we need to keep alive a-leg after the bridge and then perform the transfer. We tried all the following settings on redirect context: - set + export of variables: force_transfer_context + force_transfer_dialplan - set + export park_after_bridge=true - set continue_on_fail=true - unset of sip_redirect_context But all without success! Please find attached a pcap trace and freeswitch cli log. Thanks in advance for help. Regards Francesco

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Codec late-negotiation change between 1.6.6 and 1.6.15
by David Ponzone 22 Mar '17

22 Mar '17

Does anyone know if there was a change with late-negotiation between 1.6.6 and 1.6.15 ? I am not sure yet about the issue, I am still trying to figure it out. According to the release notes, there were no change in this area. David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr <mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr <blocked::http://www.ipeva.fr/> - www.ipeva-studio.com <blocked::http://www.ipeva-studio.com/> Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

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Luis Daniel Lucio Quiroz
by Mario G 22 Mar '17

22 Mar '17

What a scummy thing to do putting people here on linkedin which shares our email addresses. Hope no one uses this guy! Mario

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Dialplan/default.xml
by Yücel ALTUNAY 22 Mar '17

22 Mar '17

Hi, I installed FS 1.6 on Debian and using it on local lan. Do I need extensions in dialplan/defaut.xml except Local Extension? Thanks.

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G.729 patent expired. How to do now?
by José Lopes 21 Mar '17

21 Mar '17

Hello guys, I notice that the patent of G.729 is expired (since January 1, 2017) as noticed at http://www.sipro.com/G729.html . So it is possible now to use G729 at FreeSwitch without license ? Best Regards, Jose Lopes

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(no subject)
by Babak Yakhchali 21 Mar '17

21 Mar '17

Hi I want to play a phrase on aleg before bridge using this xml chunk <action application="set" data="bridge_pre_execute_aleg_app=playback"/> <action application="set" data="bridge_pre_execute_aleg_data=phrase:queue_agent:*${cc_agent}*"/> but because cc_agent is expanded before it is set, the phrase input is always empty when bridging. is there anyway to have it expanded just before bridge?

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Playing a phrase before bridge with variable expansion
by Babak Yakhchali 21 Mar '17

21 Mar '17

Hi I want to play a phrase on aleg before bridge using this xml chunk <action application="set" data="bridge_pre_execute_aleg_app=playback"/> <action application="set" data="bridge_pre_execute_aleg_ data=phrase:queue_agent:*${cc_agent}*"/> but because cc_agent is expanded before it is set, the phrase input is always empty when bridging. is there anyway to have it expanded just before bridge?

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Group Call did not release
by Shawn Wheeler 21 Mar '17

21 Mar '17

Hello, Back Ground. Recently I did a group call on my system. After hanging up the initiating phone, FS did not release the call on the receiving end points. I looked through the confluence documentation but I could not find a CLI command to kill the call. I ended up restarting the FS service. Once I did that all the calls were release. Question Is there a CLI command I and run to release all current calls? If yes, would one of you on this list share please. Thank you in advance. Shawn

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No Voice on Proxy Media
by Yücel ALTUNAY 21 Mar '17

21 Mar '17

Hi, First, i am sorry for my English. I use FS 1.4.26 on ubuntu and configured it to use proxy media mode and late negotitation with TLS. My phones are behind nats. So i must configure FS with proxy media. I can call phones but I don't hear any voice. When i capture the traffic with wireshark i see udp packets sent from phone to server and reverse. but no media. If I configure it to bypass media, it works with phone on the same lan, but not for phones behind nats. How can i solve this problem. Thanks

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mod_callcenter shared db development
by Rovshan Baghirov 21 Mar '17

21 Mar '17

Hello, As I understand mod_callcenter does not support shared database setup. However in 2 of our projects we need this functionality and would like to sponsor development of this feature. How do we proceed with finding correct person to develop this functionality? Best Regards, Rovshan

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