- FreeSWITCH-users - FreeSWITCH Mailing Lists

INCOMPATIBLE_DESTINATION
by Shlomi Agiv 10 Jun '14

10 Jun '14

Hi, I'm running freswitch version 1.4.4 and getting a strange codec error when i call from a phone running CSIPSIMPLE(registered as user 1000) to 1005, the call hangs up and I get: *2014-06-05 08:31:59.168784 [INFO] switch_ivr_originate.c:1191 Sending early media* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [SILK:99:24000:20:0]/[G722:9:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [SILK:99:24000:20:0]/[PCMU:0:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [SILK:99:24000:20:0]/[PCMA:8:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [SILK:99:24000:20:0]/[GSM:3:8000:20:13200]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:30:32000]/[G722:9:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:30:32000]/[PCMU:0:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:30:32000]/[PCMA:8:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:30:32000]/[GSM:3:8000:20:13200]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMA:8:8000:30:32000]/[G722:9:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMA:8:8000:30:32000]/[PCMU:0:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMA:8:8000:30:32000]/[PCMA:8:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMA:8:8000:30:32000]/[GSM:3:8000:20:13200]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3309 Set telephone-event payload to 101* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3626 Set 2833 dtmf send/recv payload to 101* *2014-06-05 08:31:59.168784 [ERR] mod_sofia.c:2201 CODEC NEGOTIATION ERROR. SDP:* *v=0* *o=- 3610945918 3610945918 IN IP4 10.0.1.46* *s=pjmedia* *c=IN IP4 10.0.1.46* *t=0 0* *m=audio 4020 RTP/AVP 99 0 8 101* *c=IN IP4 10.0.1.46* *a=rtpmap:99 SILK/24000* *a=fmtp:99 useinbandfec=0* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:101 telephone-event/8000* *a=fmtp:101 0-15* *a=rtcp:4021 IN IP4 10.0.1.46* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_session.c:907 Send signal sofia/internal/1000(a)10.0.1.233 <1000(a)10.0.1.233> [BREAK]* *2014-06-05 08:31:59.168784 [NOTICE] switch_channel.c:3432 Hangup sofia/internal/1000(a)10.0.1.233 <1000(a)10.0.1.233> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]* when i call using cmd 'originate user/1005 &bridge(user/1000)' The call goes through and I get: *2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]* *2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3437 Audio Codec Compare [PCMU:0:8000:20:64000] ++++ is saved as a match* *2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]* *2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000]* *2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:20:64000]/[GSM:3:8000:20:13200]* *2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3309 Set telephone-event payload to 101* *2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:2343 Set Codec sofia/internal/sip:1000@10.0.1.46:51113 <http://sip:1000@10.0.1.46:51113> PCMU/8000 20 ms 160 samples 64000 bits* *2014-06-05 08:25:18.073562 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:1000@10.0.1.46:51113 <http://sip:1000@10.0.1.46:51113> Original read codec set to PCMU:0* *2014-06-05 08:25:18.093570 [DEBUG] switch_core_media.c:3617 Set 2833 dtmf send payload to 101* *2014-06-05 08:25:18.093570 [DEBUG] switch_core_media.c:4841 AUDIO RTP [sofia/internal/sip:1000@10.0.1.46:51113 <http://sip:1000@10.0.1.46:51113>] 10.0.1.233 port 32636 -> 10.0.1.46 port 4016 codec: 0 ms: 20* *2014-06-05 08:25:18.093570 [DEBUG] switch_rtp.c:3349 Starting timer [soft] 160 bytes per 20ms* *2014-06-05 08:25:18.113587 [INFO] switch_core_media.c:5053 Activating RTCP PORT 4017* Even stranger, when I call using the same configuration under 1.2.17, the call goes through: *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [SILK:99:24000:20:0]/[G722:9:8000:20:64000]* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [SILK:99:24000:20:0]/[PCMU:0:8000:20:64000]* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [SILK:99:24000:20:0]/[PCMA:8:8000:20:64000]* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [SILK:99:24000:20:0]/[GSM:3:8000:20:13200]* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:3186 Set Codec sofia/internal/1000(a)10.0.1.233 <1000(a)10.0.1.233> PCMU/8000 20 ms 160 samples 64000 bits* *2014-06-05 08:43:39.400774 [DEBUG] switch_core_codec.c:111 sofia/internal/1000(a)10.0.1.233 <1000(a)10.0.1.233> Original read codec set to PCMU:0* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5446 Set 2833 dtmf send/recv payload to 101* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:3445 AUDIO RTP [sofia/internal/1000(a)10.0.1.233 <1000(a)10.0.1.233>] 10.0.1.233 port 23172 -> 10.0.1.46 port 4004 codec: 0 ms: 20* what could possibly be the reason for that? I have little understanding of codecs handling. Thanks, Shlomi Agiv

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bind_digit_action targeting/responding on opposite leg
by Dmitriy Shumaev 10 Jun '14

10 Jun '14

Hi. I want to use bind_digit_action instead of bind_meta_app. There is one problem - I need to do actions on opposite leg. Using bind_meta_app I can do something like this: <action application="bind_meta_app" data="2 ab o execute_extension::INFO_calloverreq XML default"/>  , but how can I do the same using bind_digit_action - ?. With best regards, Shumaev D.A., programmer @ KBR Ltd.

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Unable to call extension
by Richard Mace 09 Jun '14

09 Jun '14

Hi All, I have a vanilla debian 7 FreeSWITCH installation with a file called 201.xml in /etc/freeswitch/dialplan/default/ When I dial 201 (from extension 203) it won't ring the extension. I have included a SIP trace on: www.rocksolidpbx.co.uk/download/sofia_sip_trace.txt Any ideas what I'm doing wrong? Thanks, as always, in advance Richard

4 23

Getting logs of registrations/deregistrations
by Alex Lake 09 Jun '14

09 Jun '14

Is there a way to get the console to show end-user registrations and de-registrations as they happen? I note that registrations are OK, you get something like this: 2014-06-09 13:37:44.091960 [WARNING] sofia_reg.c:1541 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1129301(a)004-1129.sb12.dmclub.org] from ip 84.92.90.149 But de-registrations are very quiet. Also, is there an api command to send a "SIP OPTIONS" message to a particular registered session?

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Re: [Freeswitch-users] how can i get the server ip/hostname where freeswirch is installed?
by Robert Hadley 09 Jun '14

09 Jun '14

Hi, There are global variables that FS core sets at startup: IP: $${local_ip_v4} or $${local_ip_v6} HOSTNAME: $${hostname} Regards, Robert From: Snabel Kabiya [mailto:snabel@lexifone.com] Sent: Sunday, June 08, 2014 8:56 AM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] how can i get the server ip/hostname where freeswirch is installed? Hi, how can i get the server ip/hostname where freeswirch is installed? i want to get the ip/hostname and use it in lua script. Thanks, Snabel

1 0

Do the official ClueCon presentations start on August 5?
by D D 09 Jun '14

09 Jun '14

Hi, I saw that OpenSips is having a summit on August 4. Do the official ClueCon presentations start on the 5th? Thanks, David

1 0

registration-thread-frequency
by Alex Lake 09 Jun '14

09 Jun '14

Struggling to understand how this value relates to how often OPTIONS packets are sent.... I've got this in my internal.xml: <param name="all-options-ping" value="true"/> <param name="nat-options-ping" value="true"/> <param name="unregister-on-options-fail" value="true"/> <param name="registration-thread-frequency" value="5"/> But I don't see any OPTIONS messages going to my registered SIP handset: freeswitch@internal> show registrations reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata 1129301,176.9.31.72,fbbb6ffb2e8e3c9d@192.168.1.203,sofia/internal/sip:1129301@84.92.90.149:33888;transport=udp,1402068621,84.92.90.149,33888,udp,hzhmay14a.dmclub.net, And i've done this: sofia profile internal siptrace on Feel I'm banging my head against a brick wall on this!

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how can i get the server ip/hostname where freeswirch is installed?
by Snabel Kabiya 09 Jun '14

09 Jun '14

Hi, how can i get the server ip/hostname where freeswirch is installed? i want to get the ip/hostname and use it in lua script. Thanks, Snabel

3 2

Fwd: Voice recognition service
by Илья Арефьев 09 Jun '14

09 Jun '14

Hi! I've made a quick research and think that scheme softphone->audio->RTP->FS->pocketsphinx->text->RTP->softphone may take a place. Could you please acknowledge/refuse this scheme and point me how to implement or point me direction of further research? Thank you in advance! -- Ilya Arefiev

3 4

reINVITE with a=sendonly
by Dmitriy Shumaev 09 Jun '14

09 Jun '14

I'm using FS for PTT (Push-To-Talk) services. To start talking FS sends reINVITE (I use "uuid_media_reneg <_uuid>" command for this). I need to add "a=sendonly" to this reINVITE. How it can be done? With best regards, Shumaev D.A., programmer @ KBR Ltd.

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