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start_tone_detect, but no event is fired
by henry.houfeng 05 Apr '13

05 Apr '13

Hi, I want to detect call progress tone with start_tone_detect, but I can't get the event fired. My dialplan is: <extension name="skynet51"> <condition field="destination_number" expression="^(\d+)$"> <action application="start_tone_detect" data="1"/> <action application="bridge" data="sofia/gateway/skynet51/$1"/> </condition> </extension> The call senario is: Caller A is a user registered on local FS. Callee B is a user on remote gateway. A-------INVITE------------>B A<------100 Trying---------B A<-----183 with SDP-------B A<------RTP of Ring Tone--B The early media of ring tone or busy tone is transferred from B to A after 183. But I can't see the DETECTED_TONE is fired. I have set loglevel to 3 in spandsp.conf.xml: <descriptors debug-level="3"> I can only find some Tone segment at a frequencey less than 10: freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:12.618263 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = 4, f2 = 5, duration = 48 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:12.738255 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = -1, f2 = -1, duration = 128 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:12.858249 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = 4, f2 = 5, duration = 112 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:12.958243 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = -1, f2 = -1, duration = 112 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:13.018238 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = 4, f2 = 5, duration = 48 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:13.058236 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = -1, f2 = -1, duration = 48 Is my dialplan incorrect? I will very appreciate your suggestion. Thanks! Regards, Henry -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/start-tone-detect-but-no-even… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Mod Managed Conf Call
by Usama Zaidi 04 Apr '13

04 Apr '13

Hey, Any updates on when the conference call is going to be scheduled for? The thread kinda died. Regards. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of freeswitch-users-request(a)lists.freeswitch.org Sent: Tuesday, April 02, 2013 11:34 PM To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 82, Issue 19 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..."

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Call not originated through event socket
by Ashish gautam 04 Apr '13

04 Apr '13

Hi, I am generating calls through event socket using originate action to multiple numbers to PSTN network. ES is returning '+OK' for the call not generated. Still the call is not generated. Any help is appreciated. Thanks. -Ashish

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Regeneration of DTMF
by Michael Lutz 04 Apr '13

04 Apr '13

Hi, I have a problem, which I am trying to resolve, but can not exactly figure out where it is going wrong. I have an inbound call, this call comes in via SIP and uses inband dtmf, at the begining of the dialplan I enable dtmf detection using spandsp_start_dtmf. this works fine, and my Lua recognizes digits correctly. The tricky part is that I bridge this call in Lua using an api call "originate", this call is forwarded to the same switch, and is picked up by another Lua script. This script, is waiting for a custom event, to end the lua, and is bridged with a 3rd call. so the 1st and 3rd call can hear each other. This 3rd call is initiated asynchronosly by an esl server. (this al works fine, and is not 'the issue'..) The problem is the receiving end (3rd leg) is receiving the DTMF pressed by the 1st leg twice. When I eavesdrop the 2nd leg, i only hear the dtmf once, when i eavesdrop the 3rd leg, I can hear the dtmf twice. So it is somewhere generated along the way. I have tryed several different settings, using stop_dtmf_generate on different legs, but can not seem to diable this regeneration of this extra dtmf. Any help would be appreciated as this is really causing issues on my side, ps, I know this '3rd leg' principle might look a bit weird, but cannot be avoided, ps2. When my inbound call comes in using rfc2833, everything works perfectly. Best regards, Michael Lutz.

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Re: [Freeswitch-users] Can be modified switch_m_sdp?
by Adrian Fuentes 04 Apr '13

04 Apr '13

Ken Thanks for reply. I have this scenario IPPHONE --> FreeSwitch Bypass --> Other-SoftSwitch --> Gateway Internet --> Public IP --> Local IP --> IP Local & Public I want to change (to c = IN IP4 172.25.1.9) by (to c = IN IP4 200.56.93.77), to be able to bypass between IPPHONE and Gateway. This is because "Other-SoftSwitch" can not respond Gateway Public IP in SDP (to c = IN IP4 200.56.93.77). Adrian Fuentes El 04/04/2013, a las 02:03, freeswitch-users-request(a)lists.freeswitch.org escribió: > > De: Ken Rice <krice(a)freeswitch.org> > Asunto: Re: [Freeswitch-users] Can be modified switch_m_sdp? > Fecha: 3 de abril de 2013 19:07:29 CST > Para: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Responder a: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > what are you trying to do? you need to be specific... you can modify the sdp possibly but the end goal tells us how to advise this > > Ken > Sent from my iPad > > On Apr 3, 2013, at 19:18, Adrian Fuentes <adrian.fuentes(a)hovanetworks.com> wrote: > >> Hi all >> >> Is it possible to replace the sdp c = IN IP4 172.25.1.9 through 172.25.1.9 c = IN IP4? >> >> variable_switch_m_sdp: [v=0 >> o=- 13649160524104272291 13649160524104272291 IN IP4 10.5.86.99 >> s=it-ssw >> c=IN IP4 172.25.1.9 >> t=0 0 >> m=audio 6150 RTP/AVP 8 101 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> a=silenceSupp:off - - - - >> ] >> >> Regards >> Adrian Fuentes >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > >

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Set User-to-user header
by Peter Steinbach 04 Apr '13

04 Apr '13

In http://jira.freeswitch.org/browse/FS-4994 I have seen, that the User-to-User header can be passed into a channel variable. But what obout the other direction? Is there a way to set this header on an outbound call? Or do I just have to prefix with "sip_h_"? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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mod_callcenter + eavesdrop
by Tiago Sampaio 04 Apr '13

04 Apr '13

Hi ALL.. Im currently using eavesdroping on outgoing calls using: <action application="db" data="insert/spymap/${caller_id_number}/${uuid}" /> and is working fine! Now I want to hear calls from my agents, but i dun know how to insert in spymap what agent get the call from queue, any tips? im using options from callcenter.conf: <agents> <agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000@default" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" /> </agents> <tiers> <tier agent="1000@default" queue="support@default" level="1" position="1"/> </tiers> Another question is: How to put in cdr lost calls, example: queue support have agent1 and agent2. Incomming call, first ring to agent1 for 10sec and then ring and answer agent2. I wanna log the event of not answer from agent1 Im thinking about write an application to listen esl an caputure the events about callcenter and then log to cdr, am i write? Hugs!

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When doing re-INVITE procedure....
by Thomas Lee 04 Apr '13

04 Apr '13

Hi, It's SIP re-INVITE problem. Is it necessary for adding "a=sendrecv" on SDP field, when we're doing re-INVITE procedure? Thanks Regards, Thomas Lee -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/When-doing-re-INVITE-procedur… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Build Error
by info＠shishir.com.np 04 Apr '13

04 Apr '13

Hi I Came across the following error while performing make current this morning. Please suggest. quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -I/usr/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -Ilibs/sofia-sip/libsofia-sip-ua/sdp -Ilibs/sofia-sip/libsofia-sip-ua/su -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/src/freeswitch/libs/apr/include -I/usr/src/freeswitch/libs/apr-util/include -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/src/freeswitch/libs/libtpl-1.5/src -I/usr/src/freeswitch/libs/stfu -I/usr/src/freeswitch/libs/sqlite -I/usr/src/freeswitch/libs/pcre -I/usr/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/src/freeswitch/libs/srtp/include -I/usr/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -I/usr/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_rtp.lo -MD -MP -MF .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c -fPIC -DPIC -o .libs/libfreeswitch_la-switch_rtp.o cc1: warnings being treated as errors src/switch_rtp.c: In function ‘handle_ice’: src/switch_rtp.c:797: error: format ‘%ld’ expects type ‘long int’, but argument 8 has type ‘switch_size_t’ src/switch_rtp.c: In function ‘read_rtp_packet’: src/switch_rtp.c:3734: error: format ‘%ld’ expects type ‘long int’, but argument 9 has type ‘switch_size_t’ src/switch_rtp.c:3906: error: format ‘%ld’ expects type ‘long int’, but argument 11 has type ‘switch_size_t’ src/switch_rtp.c: In function ‘rtp_common_write’: src/switch_rtp.c:5289: error: format ‘%ld’ expects type ‘long int’, but argument 9 has type ‘switch_size_t’ src/switch_rtp.c:5293: error: format ‘%ld’ expects type ‘long int’, but argument 9 has type ‘switch_size_t’ make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 make[2]: Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 Thanks

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FreeTDM channel restart error
by Ashish gautam 04 Apr '13

04 Apr '13

Hi, Whenever I restart FreeSWITCH I get the following channel restart errors from mod_ftdm: 2013-04-04 10:35:43.534528 [NOTICE] ftmod_libpri.c:2159 [s1c4][1:4] -- Restart of channel completed 2013-04-04 10:35:43.534528 [WARNING] ftdm_io.c:3018 [s1c4][1:4] Channel not opened, proceeding anyway 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c5][1:5] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c6][1:6] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c7][1:7] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c8][1:8] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c9][1:9] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c10][1:10] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c11][1:11] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c12][1:12] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c13][1:13] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c14][1:14] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c15][1:15] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c17][1:17] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c18][1:18] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c19][1:19] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c20][1:20] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c21][1:21] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c22][1:22] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.494529 [ERR] ftmod_libpri.c:1950 [s1c23][1:23] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c24][1:24] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c25][1:25] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c26][1:26] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c27][1:27] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c28][1:28] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c29][1:29] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c30][1:30] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-04-04 10:36:08.514536 [ERR] ftmod_libpri.c:1950 [s1c31][1:31] -- T316 timed out, channel reached restart attempt limit '3' and is suspended It makes 3 attempts to restart all the 30 channels on span 1 of the PRI card and then show these errors. Please anybody help me getting out of this. Thanks in advance. Regards. -Ashish

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