- FreeSWITCH-users - FreeSWITCH Mailing Lists

Nibblebill
by David Villasmil 15 Feb '13

15 Feb '13

Hello guys, Is it possible to use different billing tables for different applications? Say i have 2 different context on which i want to use different tables for balance. Thanks! David

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which sofia gateway received the call
by Jeff Pyle 15 Feb '13

15 Feb '13

Hello, Is there a variable name available in the dialplan that will contain the name of the gateway that received the call? I see "variable_sip_gateway_name" and "variable_sip_gateway" on the Channel Variables page under Info App Variable Names. In my case both (without variable_) are empty. - Jeff

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Door intercom/gate controller
by Mr Nathan Downes 15 Feb '13

15 Feb '13

Hi list, I have a gate controller that only understands RFC 2833 but it calls endpoints that can only provide inband DTMF, I can't seem to get the digits recognised by the gate controller to trigger the relay using start_dtmf_generate, as they are coming back to the a-leg? Rather than out the b-leg.. It works fine when I call a SPA502g as it will provide RFC 2833.. is there some trickery I can achieve this? Or am I just doing it wrong?? J

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Hey Guys, Don¹t Forget today is the Friday Free For All on 888 this afternoon....
by Ken Rice 15 Feb '13

15 Feb '13

Joins us for some Fun at sip:888@conference.freeswitch.org or use one of the normal PSTN access numbers Todays Topic: Killer Asteroids! (or whatever else we feel like talking about) When? This afternoon... Targeted start time is 3P EST, but join us whenever you feel like it...lets get it started and see if we can keep it going all afternoon Who should join? EVERYONE... Get in here... -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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In-band and out band call issues
by Archana Venugopan 15 Feb '13

15 Feb '13

Hi, Am facing with a typical issue. In my phone settings, if i enable 'Force RFC2833 Out-of-Band DTMF' then my voicemail is not working properly( If I record any greeting or VM, only part of the message is being recorded) but offnet calls for banks or shops is working. If 'Force RFC2833 Out-of-Band DTMF' is disabled my voicemail is working properly but When I call offnet numbers like Banks or shops, then my key press is not being recognised i.e no DTMF detection. My internal .xml and external settings for DTMF is <param name="dtmf-type" value="rfc2833"/> <param name="rfc2833-pt" value="101"/> <param name="dtmf-duration" value="2000"/> Please let me know what needs to be changed so that both voicemail and offnet numbers work properly. Regards, Archana

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post record processing for mod_callcenter module
by Jayesh Nambiar 15 Feb '13

15 Feb '13

Hi All, I have been using this variable record_post_process_exec_app in my dialplan for post-processing of recordings where I actually convert the WAV files into an MP3 using Lame. I earlier used to record it in mp3 directly using the mod_shout module, but it was very CPU extensive. I wish to do the same for calls that are recorded using the mod_callcenter module. In callcenter groups, I can define record_template, but I dont find a place where I can do post record processing. One way I figured to do this, was in the dialplan do this: <action application="set" data="record_post_process_exec_app=system:lame file.wav file.mp3"/> <action application="set" data="cc_export_vars=record_post_process_exec_app "/> <action application="callcenter" data="group1"/> The above works but the challenge is when you want the agent name in the recording filename; because there is no way the agent name is known while we are in the dialplan. Basically the record template in groups is defined as follows: <param name="record-template" value="/recordings/${strftime(%Y%m%d%H%M%S)}_${caller_id_number}_${cc_agent}.mp3"/> Can we also have a record-post-parameter config parameter for callcenter module calls. Thanks for any suggestions in getting this done using a different approach !! --- Jayesh

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Event status
by Archana Venugopan 15 Feb '13

15 Feb '13

Hi, I am trying to get event status in freeswitch. For which I have passed the below line in my lua script. But I am getting error. element = e:getHeader("Orig-status") Error: 2013-02-15 10:41:14.177168 [ERR] mod_lua.cpp:198 /usr/local/freeswitch/scripts/directory.lua:201: attempt to index global 'e' (a nil value) stack traceback: /usr/local/freeswitch/scripts/directory.lua:201: in main chunk 2013-02-15 10:41:14.177168 [ERR] mod_lua.cpp:264 LUA script parse/execute error! I have installed mod_event_multicast too. Can anyone please point me out why I am not able to fetch element value. Many thanks. Regards, Archana

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hangup error missing remote port
by Kirill Sysoev 15 Feb '13

15 Feb '13

Hi Allan! Did you ever manage to solve this issue? I saw the same behavior yesterday while was trying to accept incoming calls from CS2K FreeSWITCH Version 1.3.13b+git~20130214T200725Z~b14fd4a5a4 (git b14fd4a 2013-02-14 20:07:25Z) Best regards, Kirill ######################################################################### *Allan Piske*zyryznet at gmail.com<mailto:freeswitch-users%40lists.freeswitch.org?Subject=%5BFreeswitch-users%5D%20hangup%20error%20missing%20remote%20port&In-Reply-To=> /Sun Sep 4 08:44:25 MSD 2011/ Hi, I'm having some trouble discovering the source and solution to one problem. I setup a FS box for receiving faxes, and a already have another FS box as SBC. SPA3102 ---> opensips ---> FS_SBC --> opensips --> FS_FAX this works ok, call gets answered, i hear the fax tone, the SPA3102 changes to T38 and everyone is happy. Nortel CS2000 ------> FS_SBC -------> opensips(regitrar,rtpproxy,etc) ---> FS_FAX but his doen't. This little devil CS2000 sends every invite with m=image header along with m=audio. With aparently FS doesn't like. like that: freeswitch at internal <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> recv 1215 bytes from udp/[10.143.82.250]:5060 at 08:26:08.226858: ------------------------------------------------------------------------ INVITE sip:4730365302 at 10.143.82.253 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:5060;transport=UDP;user=phone SIP/2.0 Record-Route: <sip:10.143.82.250;lr=on;ftag=-45026-10d9cdb-2668a317-10d9cdb> f: <sip:4784064435 at 10.150.65.16 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:5060 ;user=phone>;tag=-45026-10d9cdb-2668a317-10d9cdb t: <sip:4730365302 at 10.143.82.250 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:5060;user=phone> i: 416cf3e01041960a13c410d9cdb1d2cba42b82409c10f14bb8-0322-4481 CSeq: 1 INVITE User-agent: CS2000_NGSS/9.0 P-Asserted-Identity: <sip:4784064435 at 10.150.65.16 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>;user=phone> Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK,UPDATE Via: SIP/2.0/UDP 10.143.82.250;branch=z9hG4bK1d78.5262ccd1.0 v: SIP/2.0/UDP PAE1CS2K:5060;maddr=10.150.65.16;branch=z9hG4bK-10d9cdb-1d2cba43-57ebb21d Max-Forwards: 69 m: <sip:10.150.65.16:5060;transport=UDP> k: 100rel c: application/sdp l: 420 v=0 o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202 s=- p=+1 6135555555 c=IN IP4 10.152.204.202 t=0 0 m=audio 55920 RTP/AVP 18 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=fmtp:18 annexb=no m=image 64112 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy But then we have a problem ... My beloved FS_SBC changes the SDP before sending it to the next hop .. as it should becase they are on diferent subnets and set to proxy_media 2011-09-04 01:26:08.231102 [DEBUG] sofia_glue.c:1759 sofia/tpa/4730365302 Patched SDP --- v=0 o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202 s=- p=+1 6135555555 c=IN IP4 10.152.204.202 t=0 0 m=audio 55920 RTP/AVP 18 8 101 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 64112 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy +++ v=0 o=FreeSWITCH 2718891449 2718891450 IN IP4 A.B.C.D s=FreeSWITCH p=+1 6135555555 c=IN IP4 A.B.C.D t=0 0 m=audio 19262 RTP/AVP 18 8 101 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 19262 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy so we have now 2 m= fields with the same port !!!!! then when it gets answered on the FS_FAX box .. we got disconnection on the FS_SBC box 2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4761 Channel sofia/tpa/4730365302 entering state [completing][200] 2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4772 Remote SDP: v=0 o=FreeSWITCH 1315079202 1315079203 IN IP4 189.45.192.19 s=FreeSWITCH c=IN IP4 189.45.192.19 t=0 0 m=audio 31436 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 m=image 0 udptl 19 2011-09-04 01:26:08.258409 [DEBUG] sofia.c:4761 Channel sofia/tpa/4730365302 entering state [ready][200] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2830 (sofia/tpa/4730365302) Callstate Change RINGING -> ACTIVE 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2842 Send signal sofia/voxip/4784064435 at 10.150.65.16 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:5060 [BREAK] 2011-09-04 01:26:08.258409 [NOTICE] sofia.c:5318 Channel [sofia/tpa/4730365302] has been answered 2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:2774 Set Codec sofia/tpa/4730365302 PROXY/8000 20 ms 160 samples 0 bits 2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:3079 PROXY AUDIO RTP [sofia/tpa/4730365302] A.B.C.D:19262->W.X.Y.Z:0 codec: 0 ms: 20 2011-09-04 01:26:08.258409 [ERR] sofia_glue.c:3512 AUDIO RTP REPORTS ERROR: [Missing remote port] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2563 (sofia/tpa/4730365302) Callstate Change ACTIVE -> HANGUP 2011-09-04 01:26:08.258409 [NOTICE] sofia_glue.c:3513 Hangup sofia/tpa/4730365302 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2579 Send signal sofia/tpa/4730365302 [KILL] 2011-09-04 01:26:08.258409 [DEBUG] switch_core_session.c:1116 Send signal sofia/tpa/4730365302 [BREAK] 2011-09-04 01:26:08.259605 [DEBUG] switch_ivr_originate.c:3299 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2011-09-04 01:26:08.259605 [INFO] mod_dptools.c:2647 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER port = 0 !!! is that maybe a fault in opensips in the middle because the both m= fields with the same port ? , or one of the FS boxes ? something is broken here, sadly :( Appreciate any help. Allan

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Freeswitch + freetdm dumps core on Ubuntu 12.10
by Johan Helsingius 15 Feb '13

15 Feb '13

Hi! Tried compiling freeswitch with freetdm on Ubuntu 12.10 using both v1.2.stable and Master from git. Both dump core. Here is what gdb says: Program received signal SIGSEGV, Segmentation fault. 0xb7e12596 in switch_loadable_module_process (key=key@entry=0x8120ea8 "mod_file_string", new_module=new_module@entry=0x82a4200) at src/switch_loadable_module.c:321 321 for (i = 0; ptr->extens[i]; i++) { What is the best way forward? Report it as a bug? Julf

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FreeSWITCH HA BoF Meetings
by Michael Collins 14 Feb '13

14 Feb '13

Hello! Yesterday we had a nice discussion <http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_13>about all things HA. Eliot Gable gave us a lot of very good information about the state of mod_ha_development and all the work that he's done over the past 4+ years in the world of HA and FreeSWITCH. It's now time to let those who have a vested interest in HA to get together with Eliot and continuing the discussion. To that end we propose a FreeSWITCH HA BoF (Birds of a Feather) conference. Eliot says that he can be available after 8PM EST. I would like to propose that the meetup be Tuesday evenings at 8:00PM EST in the main FreeSWITCH public conference. If you would like to meetup then please email me off-list and CC Eliot ( egable+freeswitch(a)gmail.com) so that we can keep track of who is most interested in this topic. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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