June 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

transfer_after_bridge
by Andre 08 Jun '14

08 Jun '14

HI, can I call transfer_after_bridge from execute_on_answer? If so, what is the syntax to do that? Thanks

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mod_pocketsphinx: SpeechTools.jm error
by Mark Campbell-Smith 08 Jun '14

08 Jun '14

HI, I'm trying to test pocketsphinx and have been looking at the pizza example from the wiki. When I try to execute the test, I get the following error in the log: 2014-06-05 13:29:10.176189 [ERR] SpeechTools.jm:188 Exception: SyntaxError: Unexpected identifier (near: " console_log("debug", "----Hit score " + result.interpretation.@confidence") It doesn't appear that SpeechTools.jm has been changed for a while. Any ideas?

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Freeswitch FMS server as a endpoint
by info＠shishir.com.np 07 Jun '14

07 Jun '14

Hi all, I've been scratching my head with an issue - I am trying to pass (one way) audio from Freeswitch to FMS server directly via mod_rtmp. I'm able to connect to freeswitch from FMS server ; execute the dialplan ; able to receive audio on the FMS server. But the received audio from freeswitch is very choppy and lagging. I noticed I'm passing audio one way (from freeswitch to FMS) only and not other way around is this causing the issue ? I found similar behavior on FLEX client too, if I comment out the publish method. But if I publish the audio everything is normal. Is there a work around to receive one way audio only from freeswith to FMS without loosing the audio quality? Thanks Shishir

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Custom wording with SIP call rejections
by Victor Chukalovskiy 07 Jun '14

07 Jun '14

Good day, Is there a way to do custom rejection responses in FS like below? And in particular, to use them as rejection responses from limit application? "503 Mass Calling Event" "603 Don't do this" This would be helpful to let customer see specific CAC filter a call was rejected by. Thanks! -Victor

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Codecs in SDP
by Ričardas Stoma 06 Jun '14

06 Jun '14

I have created the following extension: <extension name="my_test"> <condition> <action application="bridge" data="{absolute_codec_string='PCMA,PCMU,GSM'}sofia/external/ 1001(a)192.168.1.78"/> <action application="hangup"/> </condition> </extension> then i call to this extension, freeswitch dials to 1001(a)192.168.1.78 and when i check the content of SDP, included in a SIP packet, i see something like this: v=0 o=FreeSWITCH 1401972565 1401972566 IN IP4 192.168.1.65 s=FreeSWITCH c=IN IP4 192.168.1.65 t=0 0 m=audio 17262 RTP/AVP 8 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 why my codecs from 'absolute_codec_string' are not included in SDP? I would expect to see these lines included: a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000

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Codecs in SDP
by Ričardas Stoma 06 Jun '14

06 Jun '14

I have created the following extension: <extension name="my_test"> <condition> <action application="bridge" data="{absolute_codec_string='PCMA,PCMU,GSM'}sofia/external/ 1001(a)192.168.1.78"/> <action application="hangup"/> </condition> </extension> then i call to this extension, freeswitch dials to 1001(a)192.168.1.78 and when i check the content of SDP, included in a SIP packet, i see something like this: v=0 o=FreeSWITCH 1401972565 1401972566 IN IP4 192.168.1.65 s=FreeSWITCH c=IN IP4 192.168.1.65 t=0 0 m=audio 17262 RTP/AVP 8 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 why my codecs from 'absolute_codec_string' are not included in SDP? I would expect to see these lines included: a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000

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capture/not sending DTMF to b-leg
by Dmitriy Shumaev 06 Jun '14

06 Jun '14

Hi. I use bind_meta_app and bind_digit_action: <action application="bind_meta_app" data="5 a b execute_extension::send_floor_notify_AT0 XML default"/> <action application="bind_meta_app" data="8 a b execute_extension::send_floor_notify_AT1 XML default"/> and <action application="bind_digit_action" data="default,1,exec:execute_extension,send_floor_notify_AT0 XML default"/> But I do not want to send *5 and *8 and 1 DTMF to b-leg. How can I do this? With best regards, Shumaev D.A., programmer @ KBR Ltd.

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FreeSwitch as SIP route server
by Boris Kovalenko 06 Jun '14

06 Jun '14

Hello! Is it possible to use freeswitch as SIP route server? I have commercial SIP server which does not has LCR with it. Is uses external routing instead. It sends INVITE packet to external SIP server and waits for 3XX response. I tried to get FS working as RS. My DP is very simple, I just want to see I get a call. <extension name="del-group"> <condition field="destination_number" expression="^.*$"> <action application="log" data="ALERT Here I am"/> </condition> </extension> Console output: 2014-06-07 00:41:18.475135 [NOTICE] switch_channel.c:1053 New Channel sofia/lcr_standard/230069@127.0.0.1:5062 [bc7fe4e8-7496-48b6-9381-a5debd7c2423] 2014-06-07 00:41:18.475135 [NOTICE] sofia.c:7112 Hangup sofia/lcr_standard/230069@127.0.0.1:5062 [CS_HIBERNATE] [MANDATORY_IE_MISSING] 2014-06-07 00:41:18.495131 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/lcr_standard/230069@127.0.0.1:5062) Ended 2014-06-07 00:41:18.495131 [NOTICE] switch_core_session.c:1636 Close Channel sofia/lcr_standard/230069@127.0.0.1:5062 [CS_DESTROY] So, if the FS can not be used as SIP route server may be You can suggest another software? -- Regards, Boris

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CPU Affinity
by Andre 06 Jun '14

06 Jun '14

Hi, Can anyone give me some best performance advise how I should set my CPU Affinity for using Freeswitch in a high performance Dialer environment? Thanks

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Error in codec negotiation: What are all the places that specify accepted codecs for outgoing calls?
by Mark Boots 06 Jun '14

06 Jun '14

We're using a major VoIP terminator and finding that on many outgoing calls, Freeswitch hangs up with INCOMPATIBLE_DESTINATION after trying to negotiate codecs. The carrier offers PCMA, and it looks like Freeswitch refuses to accept anything but PCMU: 2014-06-06 13:18:54.115197 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0011104555493618652! 2014-06-06 13:18:54.115197 [DEBUG] switch_channel.c:3277 (sofia/external/0011104555493618652) Callstate Change DOWN -> EARLY 2014-06-06 13:18:54.175210 [DEBUG] switch_core_session.c:1005 Send signal sofia/external/0011104555391494140 [BREAK] 2014-06-06 13:18:54.175210 [DEBUG] switch_core_session.c:1005 Send signal sofia/external/0011104555391494140 [BREAK] 2014-06-06 13:18:54.175210 [DEBUG] sofia.c:5741 Channel sofia/external/0011104555391494140 entering state [proceeding][183] 2014-06-06 13:18:54.175210 [DEBUG] sofia.c:5750 Remote SDP: v=0 o=- 313783328916230866 1 IN IP4 81.52.176.165 s=- c=IN IP4 81.52.176.165 t=0 0 a=sendrecv m=audio 25080 RTP/AVP 8 13 101 a=rtpmap:8 PCMA/8000 a=rtpmap:13 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2585 Set comfort noise payload to 13 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2662 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2662 Audio Codec Compare [CN:13:8000:20:0]/[PCMU:0:8000:20:64000] 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2662 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2824 Set 2833 dtmf send payload to 101 2014-06-06 13:18:54.175210 [DEBUG] switch_core_session.c:1005 Send signal sofia/external/0011104555391494140 [BREAK] 2014-06-06 13:18:54.175210 [NOTICE] sofia.c:5868 Hangup sofia/external/0011104555391494140 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] However, our codec settings in vars.xml specify: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> The sip profile uses these settings. Any idea how to debug what codecs Freeswitch is willing to accept? Thanks! +Mark

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