September 2015 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] FreeSWITCH and Twilio
by Michael Nielsen 13 Oct '15

13 Oct '15

Hello everyone, I'm trying to setup FreeSWITCH with Twilio - to place outbound calls. I've created a SIP trunk in Twilio and are now trying to terminate calls via Twilio - all 10 digit calls should go this way (2 digit country code and 8 digit number - Danish numbers). This is my twilio.xml in sip_profiles/external: <include> <gateway name="Twilio-outbound"> <param name="username" value="MY-USER"/> <param name="password" value="MY-PASSWORD" /> <param name="proxy" value="MY-SIP.pstn.twilio.com"/> <param name="register" value="false"/> </gateway> </include> This is my 02_twilio.xml in dialplan/default/: <include> <extension name="international.twilio"> <condition field="${toll_allow}" expression="international"/> <condition field="destination_number" expression="^(\d{10})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/Twilio-outbound/$1"/> </condition> </extension> </include> When I try and make a call for a Danish number (for example 4512345678) I get a "Bad Request 400". This is my FS log: freeswitch@MY-SIP> 2015-07-23 07:19:45.745254 [NOTICE] switch_channel.c:1075 New Channel sofia/internal/ 4561912233(a)MY-SIP.MY-DOMAIN.com [4af1a50e-5145-4ad8-8b9a-499a27e9e3ec] 2015-07-23 07:19:45.905238 [INFO] mod_dialplan_xml.c:635 Processing 4561912233 <4561912233>->4512345678 in context default 2015-07-23 07:19:45.905238 [NOTICE] switch_channel.c:1075 New Channel sofia/external/4512345678 [54267cad-615c-478a-9ecc-61950a995068] 2015-07-23 07:19:45.905238 [NOTICE] sofia.c:7543 Hangup sofia/external/4512345678 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2015-07-23 07:19:45.925245 [INFO] mod_dptools.c:3276 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2015-07-23 07:19:45.925245 [NOTICE] switch_channel.c:4747 Hangup sofia/internal/4561912233(a)MY-SIP.MY-DOMAIN.com [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2015-07-23 07:19:45.925245 [NOTICE] switch_core_session.c:1641 Session 47 (sofia/internal/4561912233(a)MY-SIP.MY-DOMAIN.com) Ended 2015-07-23 07:19:45.925245 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/4561912233(a)MY-SIP.MY-DOMAIN.com [CS_DESTROY] 2015-07-23 07:19:45.925245 [NOTICE] switch_core_session.c:1641 Session 48 (sofia/external/4512345678) Ended 2015-07-23 07:19:45.925245 [NOTICE] switch_core_session.c:1645 Close Channel sofia/external/4512345678 [CS_DESTROY]

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[Freeswitch-dev] Strange mod_portaudio permissions issue
by Mark Haun 13 Oct '15

13 Oct '15

Hi, I posted this to freeswitch-users but did not receive any replies. Probably no one but the developers have a chance to know what is happening, so I'm posting it here in the hope that it jogs someone's memory. It would be nice to understand it properly. --- I built freeswitch version 1.7.0+git~20150730T192909Z~ab7f83c654~32bit for my ARM-based home server (an NVIDIA Jetson board), with the intention of using mod_portaudio with a wireless USB headset for my main home phone. It works well when running freeswitch directly as root, but when I configured freeswitch to run at system startup, mod_portaudio could no longer find any audio devices. No problem, I thought, it's a simple permissions issue. I am using the example debian init script from the wiki (this is a Jessie install with SysV init rather than systemd). It runs freeswitch as user freeswitch, group daemon using the -u and -g command-line options. So I added "freeswitch" to the audio group, then used sudo to verify that I could play sound files with "aplay" as the freeswitch user. That worked, but it did not solve the mod_portaudio failure. Thinking there must be an issue with portaudio separate from ALSA, I fetched and built the "pa_devs" utility which enumerates and prints the available audio devices. (This is basically what mod_portaudio is doing on startup.) Running as the freeswitch user, that worked too. Hmmmm. I tried running freeswitch from the command line using sudo rather than the -u and -g options, i.e. $ sudo -u freeswitch /usr/local/freeswitch/bin/freeswitch -nc and that worked as well! Then I started experimenting with -u and -g and discovered that $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch allows mod_portaudio to load correctly, whereas $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch -g daemon causes mod_portaudio to fail. Can anyone explain what's happening here? In both cases the process UID, GID, EUID, EGID, FUID, and FGID are the same (freeswitch:daemon) as displayed with "ps xao pid,uid,gid,euid,egid,fuid,fgid". Could this be a bug in freeswitch? There are a couple of threads from 2-3 years ago in the archives, where mod_portaudio permissions issues were never resolved and the solution remained a mystery. For the time being I guess this (not using -g) is a workaround, but I'd sure like to understand what is going on. Mark

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[Freeswitch-dev] Upper or Pass thru Registration
by Abraham Celik 12 Oct '15

12 Oct '15

Hi All, Ive googled and searched rough the forum, and i can’t see that anyone has successfully implemented Upper registration on Freeswitch. I see that Sangoma has implemented the feature and mentioned that they will upload it back in a change window, but that was a few years ago. Has anyone been successful? Regards, Abraham

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[Freeswitch-dev] Freeswitch IAX2 Support
by Bharat Lalcheta 12 Oct '15

12 Oct '15

Hello All, I have one DIDWW DID configured with IAX2 protocol and routed to this server. When i am dialling this DID, i am getting calls on server on 4569 port which i checked in ngrep. However, there is no event on freeswitch. Just for your information, I configured mod_opal on both latest 1.2 and 1.4 version and found same result for both. Can anyone please guide me what i need to do additionally to get IAX2 calls on freeswitch? Or can anyone guide me how to troubleshoot this issue ? Do i need to configure anything for IAX2 ? Is there any thi ng i can test to check it is configured and working properly ? Thanks & Regards, Bharat Lalcheta

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[Freeswitch-dev] About mod_java Console output (stdout/stderr)
by Pong Lam 12 Oct '15

12 Oct '15

Hi all, By default , System.out.println will output to stdout and throwing Exception will output to stderr. But I cannot find any output in the freeswitch.log, please advice? Regards Pong

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[Freeswitch-dev] Safely Traversing call lists
by Ronnie Beck 12 Oct '15

12 Oct '15

Hi All, I have just a small question about the thread safety of the following code: switch_console_callback_match_t *callList = switch_core_session_findall(); for( node = callList->head; node; node = node->next ) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "(%s) Checking session '%s'\n", __FUNCTION__, node->val); } I just want to check some specific channel variables on all the active calls on a Freeswitch server. However it seems to me that while traversing the list, it is possible that "node" could be deleted by freeswitch while my code is still using it. Is there some sort of protection I should be doing in order to safely traverse the list and investigate each session? I thought I could lock each session as I interrogate the session however this doesn't protect against the situation where the "next" node is deleted as my code switches to that node. In C++, if you traverse a list with an iterator, changes to the list invalidate the iterator. So one must obtain a new iterator each time the list is changed. But I don't see how to do this with the list provided by switch_core_session_findall(). Any tips? Cheers, Ronnie Beck

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[Freeswitch-dev] libesl.a is no longer being built ???
by Thomas Chen 12 Oct '15

12 Oct '15

in previous version (at least 1.5.6b, when building event socket, a copy of libesl.a is built in libs/esl/ directory.... i just loaded the 1.5.14, it does not seem to build libesl any more... is there a reason for not building it ??? is there a replacement somewhere else ? thanks

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Re: [Freeswitch-dev] [Freeswitch-users] FreeSWITCH Unstable Repo
by Michael Jerris 12 Oct '15

12 Oct '15

For core freeswitch, likely yes. For the deps, someone will have to do the work and find out. I'd prefer to do this under cowbuilder along side how we build for all other platfroms, but someone will need to do the work to figure out what all exactly is necessary to make it happen. > On Jun 18, 2015, at 11:23 AM, Stanislav Sinyagin <ssinyagin(a)gmail.com> wrote: > > Does cross-compiling work? Because building arm packages on an arm server takes about 6 hours. > > On Jun 18, 2015 4:45 PM, "Brian West" <brian(a)freeswitch.org <mailto:brian@freeswitch.org>> wrote: > If the community steps up and assists on this front we can expand our target platforms to non-x86, ubuntu, and various others... We'll have the RPM based unstable repo soon too. Ken is wrapping that up soon. > > So anyone wanting to get involved please email us. > > On Thu, Jun 18, 2015 at 9:34 AM, jungle Boogie <jungleboogie0(a)gmail.com <mailto:jungleboogie0@gmail.com>> wrote: > On 18 June 2015 at 07:18, Brian West <brian(a)freeswitch.org <mailto:brian@freeswitch.org>> wrote: > > > > To help speed the testing and release cycle up, Team FreeSWITCH has created a FreeSWITCH Unstable Debian repo, This repo is updated after every commit to master and the debs lag behind about an hour from each commit. > > > Wow, this is really wonderful. > > Are you considering also packages for non x86 platforms, too? Specifically arm. > > > > -- > ------- > inum: 883510009027723 > sip: jungleboogie(a)sip2sip.info <mailto:jungleboogie@sip2sip.info> > xmpp: jungle-boogie(a)jit.si <mailto:jungle-boogie@jit.si> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/> > > > > -- > Brian West > brian(a)freeswitch.org <mailto:brian@freeswitch.org> > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com <http://www.freeswitchbook.com/> > http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/> > ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> | Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> > T:+19184209001 <tel:%2B19184209001> | F:+19184209002 <tel:%2B19184209002> | M:+1918424WEST (9378) > iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.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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Re: [Freeswitch-dev] Audio files are not bein played
by Vikas Kumar 22 Sep '15

22 Sep '15

HI , Can anybody answer the queries, its urgent. From: Vikas Kumar Sent: 21 September 2015 17:26 To: 'freeswitch-dev(a)lists.freeswitch.org' Subject: Audio files are not bein played HI All, We are using Linphone and zoiper on other hand. We have some limit in dial plan that only one call is admissible at one time. If I call from other softphone while call is in progress than this limit is hit. See logs: EXECUTE sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> db(insert/realm/key/value) EXECUTE sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> limit(hash 119.9.73.183 1019 1 handle_over_limit XML over_limit_actions) 2015-09-21 17:24:43.362206 [INFO] switch_limit.c:126 incr called: 119.9.73.183_1019 max:1, interval:0 2015-09-21 17:24:43.362206 [INFO] mod_hash.c:193 This is the problem 1 2015-09-21 17:24:43.362206 [INFO] mod_hash.c:213 Usage for 119.9.73.183_1019 is already at max value (1) , max = 1, increament= 1 , remote_usage.total_usage= 0 2015-09-21 17:24:43.362206 [DEBUG] switch_ivr.c:1839 (sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183>) State Change CS_EXECUTE -> CS_ROUTING 2015-09-21 17:24:43.362206 [DEBUG] switch_core_session.c:1357 Send signal sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> [BREAK] 2015-09-21 17:24:43.362206 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> [BREAK] 2015-09-21 17:24:43.362206 [NOTICE] switch_ivr.c:1846 Transfer sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> to XML[handle_over_limit@over_limit_actions] 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183>) State EXECUTE going to sleep 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183>) Running State Change CS_ROUTING 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:474 (sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183>) State ROUTING 2015-09-21 17:24:43.362206 [DEBUG] mod_sofia.c:150 sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> SOFIA ROUTING 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:117 sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> Standard ROUTING 2015-09-21 17:24:43.362206 [INFO] mod_dialplan_xml.c:558 Processing 1010 <1010>->handle_over_limit in context over_limit_actions Dialplan: sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> parsing [over_limit_actions->oops, too many calls for this one] continue=false Dialplan: sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> Regex (PASS) [oops, too many calls for this one] destination_number(handle_over_limit) =~ /handle_over_limit/ break=on-false Dialplan: sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> Action answer() Dialplan: sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> Action playback(/usr/local/videocallhandler/media/InQue1.wav) Dialplan: sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> Action playback(/usr/local/videocallhandler/media/id.wav) Dialplan: sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> Action playback(/usr/local/videocallhandler/media/InQue2.wav) Dialplan: sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> Action endless_playback(/tmp/times.wav) 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183>) State Change CS_ROUTING -> CS_EXECUTE 2015-09-21 17:24:43.362206 [DEBUG] switch_core_session.c:1357 Send signal sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> [BREAK] 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:474 (sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183>) State ROUTING going to sleep 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183>) Running State Change CS_EXECUTE 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183>) State EXECUTE 2015-09-21 17:24:43.362206 [DEBUG] mod_sofia.c:243 sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> SOFIA EXECUTE 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:209 sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> Standard EXECUTE EXECUTE sofia/internal/1010(a)119.9.73.183<mailto:sofia/internal/1010@119.9.73.183> answer() 2015-09-21 17:24:43.362206 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[G729:18:8000:20:8000] 2015-09-21 17:24:43.362206 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2015-09-21 17:24:43.362206 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2015-09-21 17:24:43.362206 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200] .wav file are not being played . I am stuck at this point , Please help Thanks Vikas Disclaimer :- The information in this e-mail and any attachments is confidential and may be legally privileged. It is intended solely for the addressee or addressees. If you are not an intended recipient, please delete the message and any attachments and notify the sender of misdelivery. Any use or disclosure of the contents of either is unauthorised and may be unlawful. All liability for viruses is excluded to the fullest extent permitted by law. Any views expressed in this message are those of the individual sender, except where the sender states them, with requisite authority, to be those of the specific TIMES GROUP company.

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[Freeswitch-dev] Audio files are not bein played
by Vikas Kumar 21 Sep '15

21 Sep '15

HI All, We are using Linphone and zoiper on other hand. We have some limit in dial plan that only one call is admissible at one time. If I call from other softphone while call is in progress than this limit is hit. See logs: EXECUTE sofia/internal/1010(a)119.9.73.183 db(insert/realm/key/value) EXECUTE sofia/internal/1010(a)119.9.73.183 limit(hash 119.9.73.183 1019 1 handle_over_limit XML over_limit_actions) 2015-09-21 17:24:43.362206 [INFO] switch_limit.c:126 incr called: 119.9.73.183_1019 max:1, interval:0 2015-09-21 17:24:43.362206 [INFO] mod_hash.c:193 This is the problem 1 2015-09-21 17:24:43.362206 [INFO] mod_hash.c:213 Usage for 119.9.73.183_1019 is already at max value (1) , max = 1, increament= 1 , remote_usage.total_usage= 0 2015-09-21 17:24:43.362206 [DEBUG] switch_ivr.c:1839 (sofia/internal/1010(a)119.9.73.183) State Change CS_EXECUTE -> CS_ROUTING 2015-09-21 17:24:43.362206 [DEBUG] switch_core_session.c:1357 Send signal sofia/internal/1010(a)119.9.73.183 [BREAK] 2015-09-21 17:24:43.362206 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1010(a)119.9.73.183 [BREAK] 2015-09-21 17:24:43.362206 [NOTICE] switch_ivr.c:1846 Transfer sofia/internal/1010(a)119.9.73.183 to XML[handle_over_limit@over_limit_actions] 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/1010(a)119.9.73.183) State EXECUTE going to sleep 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/1010(a)119.9.73.183) Running State Change CS_ROUTING 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:474 (sofia/internal/1010(a)119.9.73.183) State ROUTING 2015-09-21 17:24:43.362206 [DEBUG] mod_sofia.c:150 sofia/internal/1010(a)119.9.73.183 SOFIA ROUTING 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:117 sofia/internal/1010(a)119.9.73.183 Standard ROUTING 2015-09-21 17:24:43.362206 [INFO] mod_dialplan_xml.c:558 Processing 1010 <1010>->handle_over_limit in context over_limit_actions Dialplan: sofia/internal/1010(a)119.9.73.183 parsing [over_limit_actions->oops, too many calls for this one] continue=false Dialplan: sofia/internal/1010(a)119.9.73.183 Regex (PASS) [oops, too many calls for this one] destination_number(handle_over_limit) =~ /handle_over_limit/ break=on-false Dialplan: sofia/internal/1010(a)119.9.73.183 Action answer() Dialplan: sofia/internal/1010(a)119.9.73.183 Action playback(/usr/local/videocallhandler/media/InQue1.wav) Dialplan: sofia/internal/1010(a)119.9.73.183 Action playback(/usr/local/videocallhandler/media/id.wav) Dialplan: sofia/internal/1010(a)119.9.73.183 Action playback(/usr/local/videocallhandler/media/InQue2.wav) Dialplan: sofia/internal/1010(a)119.9.73.183 Action endless_playback(/tmp/times.wav) 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/1010(a)119.9.73.183) State Change CS_ROUTING -> CS_EXECUTE 2015-09-21 17:24:43.362206 [DEBUG] switch_core_session.c:1357 Send signal sofia/internal/1010(a)119.9.73.183 [BREAK] 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:474 (sofia/internal/1010(a)119.9.73.183) State ROUTING going to sleep 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/1010(a)119.9.73.183) Running State Change CS_EXECUTE 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/1010(a)119.9.73.183) State EXECUTE 2015-09-21 17:24:43.362206 [DEBUG] mod_sofia.c:243 sofia/internal/1010(a)119.9.73.183 SOFIA EXECUTE 2015-09-21 17:24:43.362206 [DEBUG] switch_core_state_machine.c:209 sofia/internal/1010(a)119.9.73.183 Standard EXECUTE EXECUTE sofia/internal/1010(a)119.9.73.183 answer() 2015-09-21 17:24:43.362206 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[G729:18:8000:20:8000] 2015-09-21 17:24:43.362206 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2015-09-21 17:24:43.362206 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2015-09-21 17:24:43.362206 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200] .wav file are not being played . I am stuck at this point , Please help Thanks Vikas Disclaimer :- The information in this e-mail and any attachments is confidential and may be legally privileged. It is intended solely for the addressee or addressees. If you are not an intended recipient, please delete the message and any attachments and notify the sender of misdelivery. Any use or disclosure of the contents of either is unauthorised and may be unlawful. All liability for viruses is excluded to the fullest extent permitted by law. Any views expressed in this message are those of the individual sender, except where the sender states them, with requisite authority, to be those of the specific TIMES GROUP company.

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