January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Unloop extension
by Ержан Туле пов 11 Apr '13

11 Apr '13

What unloop extension is used for? BTW: vars.xml pre-(process)-sets unroll_loops, whereas DPs (public and default) are using channel variable syntax? Is this ok? Y.

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zrtp passthrough
by Kosztolányi Péter 09 Apr '13

09 Apr '13

I need to set up a passthrough zrtp in freeswitch. I followed this wiki: http://wiki.freeswitch.org/wiki/ZRTP and setup two Jitsi softphones. MiTM: works well but with different SAS in the softphones. It is ok because of a missing zrtp enrollment process. Passthrough: no SAS at all and Jitsi says the media is not encrypted I tried to achieve passthrough by uncommenting sip_profiles/internal.xml: <param name="inbound-proxy-media" value="true"/> What is the proper way to set up passthrough zrtp and having an end-to-end zrtp session with the same SAS on both softphones? Thank you, Peter

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FAX problems
by Steven Schoch 08 Apr '13

08 Apr '13

Thank you guys, I just sorted out the multiple sofia profiles and have FS accepting SIP registrations on our LAN. I plugged a phone into the LAN and it works. Now I'm trying to get this VoIP Plug-In for Microsoft Fax to work, that I downloaded from www.faxback.com. I configured it with a directory number I created, the dialplan seems to work, but it fails when sending the test FAX. The FAX number I'm calling is a POTS line connected to a Fax modem, which is answered by Hylafax. This works and is used everyday. The faxback software is installed on a Windows 7 machine, and configured by putting in a proxy address of 192.168.1.17, a username and password, and address of user@domain. The test call rings the fax modem, it answers, but then after some handshaking, it hangs up and the window says the call was hung up. Hylafax says: "No answer (T.30 T1 timeout)" -- Steve

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adding In-Reply-To in sip header not working
by Mario G 08 Apr '13

08 Apr '13

I never did this so I must be missing something, I tried both below but the bridge then fails. Can anyone shed some light on what I am doing wrong. My ITSP now supports in-reply-to so I can pass the caller ID to a forwarded call from FS. Mario G <action application="set" data="sip_rh_In-Reply-To=${caller_id_number}"/> <action application="set"><![CDATA[sip_h_In-Reply-To=<sip:${caller_id_number}>]]></action>

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Spandsp modems as non-root?
by Alex Crow 06 Apr '13

06 Apr '13

Hi, Has anyone figured out how to get the spandsp emulated modems running when Freeswitch is running as a non-root user yet? I don't even get the /dev/pts/pts* devices added either on debian or Ubuntu. Thanks Alex

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inbound-proxy-media with attended transfer
by Jeff Pyle 28 Mar '13

28 Mar '13

Hello, Is inbound-proxy-media compatible with attended transfers in any way, shape or form? Is there any way to avoid: [ERR] switch_core_io.c:1151 Codec PROXY PASS-THROUGH encoder error! Is there any way to emulate the behavior of a "proxy-media-after-att-xfer" option? I strongly prefer the codec transparency of inbound-proxy-media, but I do need attended transfers, even if the ringback is always with G711. SIP transfers in general are relatively new to me, so I very well may be asking for my cake, eating it, and then wanting the bakery to deliver daily to my office for free. Forgive me if that's the case. - Jeff

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video calls, full processing media mode
by royj 23 Mar '13

23 Mar '13

Is there ability to establish video call through FreeSWITCH with full processing media mode on profile. I see FreeSWITCH answers 183 Session Progress with 'm=video 0 RTP/AVP 19.', when video-phone offers for example 'm=video 4002 RTP/AVP 97.' -- Regards, royj

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ERROR! unable to open ZRTP cache file
by Vbvbrj 22 Mar '13

22 Mar '13

Hello. Regularly in FS log I see: 09:03:54.044779 [DEBUG] switch_rtp.c:915 [ zrtp cache]: Storing ZRTP Cache... 09:03:54.044779 [DEBUG] switch_rtp.c:915 [ zrtp cache]: Storing ZRTP cache to <... 09:03:54.044779 [DEBUG] switch_rtp.c:915 [ zrtp cache]: ERROR! unable to open ZRTP cache file <. 09:03:54.044779 [DEBUG] switch_rtp.c:915 [ zrtp cache]: Storing ZRTP Cache - DONE. 09:03:54.044779 [DEBUG] switch_rtp.c:776 Saving ZRTP cache: OK Why is this logged, and why an error? There is no file name to search for permission or access. Note the missing ">" where file name must be. -- Mimiko desu.

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Cannot Kill stale channels
by Marc de Corny 20 Mar '13

20 Mar '13

Hi I have calls that come into a queue and then get pulled out by a lua script in background and transfered to a destination. all works fine except that it looks like the sessions a not clearing from FS when all the parties clear the calls. When I do show channels, I get this for example: uuid direction created created_epoch name state cid_name cid_num ip_addr dest presence_id presence_data callstate callee_name callee_num callee_direction call_uuid hostname sent_callee_name sent_callee_num 5e8215e7-b036-4cce-9b96-e57db92f13ee outbound 03/12/2012 09:32 1.35E+09 sofia/external/02031950164 CS_HANGUP Outbound Call 2031950164 135.196.144.32 agent_to_queue_paymentsense_queue_service ACTIVE SEND 5e8215e7-b036-4cce-9b96-e57db92f13ee freeswitch2 4.4209E+11 2089623100 But when I try to get kill or query it, I get a message that the call is not anywhere in the FS. /usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_kill 5e8215e7-b036-4cce-9b96-e57db92f13ee" -ERR No Such Channel! /usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_exists 5e8215e7-b036-4cce-9b96-e57db92f13ee" false Does anyone have an idea how I can kill that call. If I restart the FS, they clear, but the problem is that they are hitting my limit on number of simultaneous calls. Any help much appreciated. Thansk marc

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Playing of mms stream by means of FS
by Yuriy Nasida 05 Mar '13

05 Mar '13

Hi guys What the best way for the playing of mms stream like this mms://live.rfn.ru/rcult_64 by means of FS ?I try to use mod_shout but without success. Do I have to use mplayer or vlc for this ?I saw this in wiki http://wiki.freeswitch.org/wiki/Mod_vlc But... probably there is the more better way ? Please adviseThanks.

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