March 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ClueCon is Coming. Start Getting Ready!
by Ken Rice 20 Mar '15

20 Mar '15

New Post on freeswitch.org from anthm check it out at http://ift.tt/1x6SG9o ClueCon is Coming. Start Getting Ready! ClueCon was a big hit last year but don’t take it from us. Here are some testimonials from 2014. Don’t forget the call for speakers is now live. Visit http://ift.tt/1FBM7gN and reserve a slot!

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start_dtmf when no RFC 2833
by Phillip Jones 19 Mar '15

19 Mar '15

Hi All, The following dial plan detects RFC 2833 and if it is not present, runs the "start_dtmf" application. <condition field="${switch_r_sdp}" expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never"> <action application="set" data="rtp_payload_number=$1"/> <anti-action application="start_dtmf"/> </condition> I need to do this on the outbound leg - using Lua. Can anyone point me to some code example that would achieve this or get me started? Any help appreciated. Thanks Phil

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Problem with make current
by Guillermo Ruiz Camauer 19 Mar '15

19 Mar '15

I am having problems trying to update my Freeswitch instances. MAKE CURRENT ends with: make[1]: Entering directory `/usr/src/freeswitch' Pulling updates... error: Failed connect to freeswitch.org:443; Connection timed out while accessing https://freeswitch.org/stash/scm/fs/freeswitch.git/info/refs fatal: HTTP request failed make[1]: *** [update] Error 1 make[1]: Leaving directory `/usr/src/freeswitch' Are the stash servers down? -- Guillermo Ruiz Camauer

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variable_sip_to_tag not set in Leg A for bridged call
by Luis Azedo 19 Mar '15

19 Mar '15

Hi, making a bridge call and listening to all events in both legs with esl. i was expecting that CHANNEL_PROGRESS_MEDIA / CHANNEL_ANSWER (on Leg A) would include variable_sip_to_tag since there is already this info (seen in message sent) but we never get this variable to be set in any event. we only get the to_tag on leg A when the call is disconnected or if we put onhold/unhold to force the creation of event. during the call (before forcing hold/unhold) i do a uuid_dump of leg A and i can see the to_tag variable. the channel variable seems to be set only on nua_i_ack and there seems to be no event fired after that. thoughts ? Best

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Automated testing
by Stanislav Sinyagin 19 Mar '15

19 Mar '15

I need to build a test suite for a service provider's SIP service infrastructure. The box will initiate and accept calls and verify that SIP messages are valid, audio media is two-way, and call control features are also working. My plan for the moment is to use FreeSWITCH and to trigger the outbound calls via ESL, and then use packet capture and Perl modules for parsing and verifying the SIP messages. What are other options? I see that SIPp doesn't look too bad, but I never worked with it. This page lists mostly outdated and closed-source packages, so not very helpful: http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing I'm planning to publish as much of my work as possible as open source. Any input will be appreciated. thanks, stan

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Any whay to reload vars.xml?
by Victor Medina 19 Mar '15

19 Mar '15

Hi guys! Is there any way to reload vars.xml? -- Víctor E. Medina M. Software [image: Zoiper Click2Dial]+58424 291 4561 <zoiper://+584242914561>[image: ve] BB #79A8AFA2 /@VMCibersys

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bug with events?
by Денис 19 Mar '15

19 Mar '15

Hi. Possible, we found a bug. If we call to freeswitch with redirect to mobile phone and hangup before state "ringing" - ESL events show leg B without Call-ID (i.e. CHANNEL_HANGUP, CHANNEL_HANGUP_COMPLETE...) example (json view) https://pastebin.freeswitch.org/24022

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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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