September 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Audio files are not bein played
by Vikas Kumar 24 Sep '15

24 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: Dialplan: sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> Action playback(/usr/local/videocallhandler/media/InQue1.wav) Dialplan: sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> Action playback(/usr/local/videocallhandler/media/id.wav) Dialplan: sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> Action playback(/usr/local/videocallhandler/media/InQue2.wav) Dialplan: sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> Action endless_playback(/tmp/times.wav) 2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183>) State Change CS_ROUTING -> CS_EXECUTE 2015-09-22 11:38:43.542214 [DEBUG] switch_core_session.c:1357 Send signal sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> [BREAK] 2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:474 (sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183>) State ROUTING going to sleep 2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183>) Running State Change CS_EXECUTE 2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183>) State EXECUTE 2015-09-22 11:38:43.542214 [DEBUG] mod_sofia.c:243 sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> SOFIA EXECUTE 2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:209 sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> Standard EXECUTE EXECUTE sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> answer() 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[G729:18:8000:20:8000] 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200] 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3216 Set Codec sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> GSM/8000 20 ms 160 samples 13200 bits 2015-09-22 11:38:43.542214 [DEBUG] switch_core_codec.c:111 sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> Original read codec set to GSM:3 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5477 Set 2833 dtmf send/recv payload to 101 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3475 AUDIO RTP [sofia/internal/1016(a)119.9.73.183] 119.9.73.183 port 25094 -> 10.150.178.79 port 60204 codec: 3 ms: 20 2015-09-22 11:38:43.542214 [DEBUG] switch_rtp.c:2041 Starting timer [soft] 160 bytes per 20ms 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3742 Set 2833 dtmf send payload to 101 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3748 Set 2833 dtmf receive payload to 101 2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3775 sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183> Set rtp dtmf delay to 40 2015-09-22 11:38:43.542214 [NOTICE] sofia_glue.c:4386 Pre-Answer sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183>! 2015-09-22 11:38:43.542214 [DEBUG] switch_channel.c:3370 (sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183>) Callstate Change RINGING -> EARLY 2015-09-22 11:38:43.542214 [DEBUG] mod_sofia.c:866 Local SDP sofia/internal/1016(a)119.9.73.183<mailto:sofia/internal/1016@119.9.73.183>: v=0 o=FreeSWITCH 1442877029 1442877030 IN IP4 119.9.73.183 s=FreeSWITCH c=IN IP4 119.9.73.183 t=0 0 m=audio 25094 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Warning given is "/usr/local/videocallhandler/media/Hold.wav sample rate 11025 doesn't match requested rate 8000" Should I ignore this warning, ?? 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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channel variable access issue
by John Nash 24 Sep '15

24 Sep '15

Here is my call flow and issue Step1 -- Call comes to freeswitch default dialplan Step2 -- I set some custom channel variables )sip_h_my_var Step3 -- I fork the call to multiple destinations using bridge application with coma separated list. Step4 - As one of the destination is my own freeswitch itself, call again comes to same profile with different destination prefix Step5 - I catch the call in different extension and try to access channel variable sip_h_my_var but get nil value I tried export application and Bridge application arguments to pass variables but I do not see my variables when i dump variables using Info application. I only see the variables as per the Initial invite (It converts some X- variables to sip_h_X variables) What else can I try?

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Timeout counted from receiving 180
by Joseph Waite 23 Sep '15

23 Sep '15

Hi Everyone Im wondering if there is a way to set a timeout ina similar way to originate_timeout when placing a call but that only starts the timeout counter after freeswitch recieves the 180 Ringing from the carrier. We have a carrier which on an invalid number does not send a 404 straight away, instead they send early media with a recorded message for over 30 seconds saying number is invalid. This causes the call to timeout and be classed in CDR's as a NOANSWER instead of INVALID I have run a sip trace and both a valid and invalid number recieve a 183 eraly media message but only a valid number recieves a 180. I have tried ignore_early_media but this does not seem to work. Regards

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[mod_hiredis installation error]
by Bilal Abbasi 22 Sep '15

22 Sep '15

Hi Users, I am getting the below error while building the mod_hiredis, Makefile:815: *** You must install libhiredis-dev to build this module. Stop. I can see libhiredis in /usr/lib64/libhiredis.so i have installed redis,hiredis and hiredis-devel OS: CentOS release 6.6 (Final) Regards Bilal Abbasi

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Announcing the OpenSIPS Summit in Austin, TX on November 9th & 10th 2015 at OpenSIPS Week
by Alex Goulis 22 Sep '15

22 Sep '15

The OpenSIPS Project is happy to announce that this year's US summit will be held in Austin, TX on November 9th and 10th 2015! Come and help us "Keep Austin Wierd" at our eclectic venue, The Contemporary Austin Jones Center. "The Contemporary Austin reflects the spectrum of contemporary art"... and for 2 days in November it will reflect a spectrum of SIP. This will be a 100% OpenSIPS event showcasing the star of this year's summit... OpenSIPS 2.1! We'll be presenting, via presentations and workshops, all the great additions we have in 2.1 - webRTC, async queries, SIP compression, fraud detection and many others. Of course, the Summit is not only about 2.1, but also about tools, platforms or engines which are OpenSIPS interconnected (like billing, interfaces and others). We'll also have presentations from companies using OpenSIPS in their production environments, giving us insight into it's real world applications. You can get more information about the summit at: http://www.opensips.org/Community/Summit-2015Austin For beginners, get answers to questions like: * why OpenSIPS is the right choice for you network * what are the OpenSIPS capabilities, features and limits * how to extend, expand and enhance your SIP service with OpenSIPS Seasoned professionals can: * Learn about new modules and features * Discuss advanced topics * Discuss the development roadmap for OpenSIPS We'll be announcing a full list of Speakers soon, but people you can look forward to include: * Bogdan-Andrei Iancu - Founder of the OpenSIPS Project * Flavio Gonclaves - SipPulse * Alex Goulis - RateTel Become an OpenSIPS Certified Professional! Don't miss this rare opportunity to train live with the founder of the OpenSIPS Project, Bogdan-Andrei Iancu! As part of OpenSIPS Week in Austin, we will be holding a live training and examination to become an OpenSIPS Certified Professional. To register and get more information go to: http://ebootcamp.opensips.org Our hotel for the event will be The OMNI Austin Downtown. It is across the street from the summit Venue and located in the heart of downtown Austin. Hotel information can be found at: http://www.opensips.org/Community/Summit-2015Austin#toc5 More information on our venue can be found at: http://www.thecontemporaryaustin.org Interested in coming early or staying later to visit Austin? What to do while you're here: http://www.opensips.org/Community/Summit-2015Austin#toc16 We look forward to seeing you in Austin! -The OpenSIPS Project Team

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fork dialing
by Anand Kumar 22 Sep '15

22 Sep '15

Hi Guys, I am looking help on fork dialing. Using below XML i am able to dial the call on both destination, but once sound file has started to playing on lookback PSTN gateway channel is stopped. I don't want cancel or stop ringing any channel until one channel has answered or timeout and freeswitch don't send the cancel to destination. <?xml version = "1.0" encoding="UTF-8" standalone="no"?> <document type="Switch/xml"> <section name="dialplan" description="RE Dial Plan For Switch"> <context name="default"> <extension name="outbound_international"> <condition field="destination_number" expression="^(\+?)(\d+)$"> <action application="set" data="hangup_after_bridge=True"/> <action application="set" data="continue_on_fail=TRUE"/> <action application="pre_answer"/> <action application="set" data="call_timeout=60"/> <action application="set" data="execute_on_answer=sched_hangup +3120 alloted_timeout "/> <action application="set" data="bypass_media=false"/> <action application="bridge" data="{enable_heartbeat_events=30,nibble_rate=0.010000,nibble_account=AS_288, absolute_codec_string=${ep_codec_string}}sofia/internal/91XXXXXXX(a)X.X.X.X ,loopback/915454/default1"/> </condition> </extension> </context> <context name="default1"> <extension name="star" continue="true"> <condition field="destination_number" expression="^(\+?)(\d+)$"> <action application="playback" data="EN/thanks.wav" /> </condition> </extension> </context> </section> </document>

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deny_refer_requests doesn't work
by Alex Kostenko 22 Sep '15

22 Sep '15

Hello, I need to deny refers per session so I found this chan variable and it worked for me in ver. 1.2.17. But after upgrade I can't force to work this variable. Where can be the issue? I use in my dialplan: <action application="set" data="deny_refer_requests=true"/> But I still see REFER in allowed methods. Tanks. Alex

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[Condition AND Problem]
by Bilal Abbasi 22 Sep '15

22 Sep '15

Hi All, I have the following as my dial plan, <extension name="ASDF" continue="true"> <condition> <action application="set" data="effective_caller_id_number=2"/> <action application="log" data="CRIT This is Caller_ID_NUM ${effective_caller_id_number}"/> </condition> <condition field="destination_number" expression="(^\d{11}$|^\d{10}$)"/> <condition field="${effective_caller_id_number}" expression="2"> <action application="log" data="CRIT This is Caller_ID_NUM ${effective_caller_id_number}"/> <action application="hangup"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="set" data="bypass_media=true"/> <anti-action application="log" data="CRIT Caller_ID_NUM ${effective_caller_id_number}"/> <anti-action application="bridge" data="sofia/external/$ 1(a)10.100.0.60"/> </condition> </extension> I have changed the effective called id to 2, now what I expect is that action should be executed, but dial plan goes to the anti-action Don't know what I am missing out. Help is appropriated, Regards Bilal Abbasi

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FreeSWITCH Week in Review (Master Branch) September 12th-18th
by Ken Rice 21 Sep '15

21 Sep '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1YwCwRd FreeSWITCH Week in Review (Master Branch) September 12th-18th Hello, again. This past week in the FreeSWITCH master branch we had 57 commits. The features for this week include: added support for timestamp based counting for jitter buffer in audio mode, added support for X-headers in this 3p mode in mod_sofia, and fine-tuning FEC with repacketization and improved jitter buffer debugging with FEC in mod_opus. And, a security issue was fixed by properly handling malformed json when parsing json! Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. Security Issues: FS-8160 Fixed head overflow in json parser, properly handle malformed json when parsing json with \u at the end of a json string. Thank you to Marcello Duarte who discovered and reported this issue. New features that were added: FS-8053 Additional work toward handling a=sendonly, a=sendrecv, a=recvonly to change who is sending video during a call FS-8130 Added support for timestamp based counting for jitter buffer in audio mode FS-6833 FS-6834 [mod_sofia] Added support for X-headers in this 3p mode FS-8175 Added continue_on_answer_timeout variable to allow channel to proceed from a tripped answer timeout FS-8080 [mod_opus] Fine-tune FEC with repacketization (ptimes: 80 ms,100 ms,120 ms) FS-8179 [mod_opus] Improved jitter buffer debugging with FEC Improvements in build system, cross platform support, and packaging: FS-8165 [Debian] Fixed build depends for mod_hiredis FS-5660 [Debian] Added freeswitch.py to the freeswitch-mod-python Debian package FS-6972 [Debian] Fixed a whitespace error in bootstrap.sh The following bugs were squashed: FS-6833 FS-6834 Fixed double re-invite on media establishment in new late offer invite feature FS-8053 [mod_conference] Fixed some regressions from original merge and added auto mute-unmute when toggling video send/receive FS-8114 Fixed opus and telephone event payload types colliding on REFER FS-8169 Fixed uuid_displace on stereo channels can lead to memory corruption causing a crash FS-8167 [mod_lua] Fixed a segfault caused by using api:execute or session:execute and not quoting the first argument like api:execute(log, “Second argument”) instead of api:execute(“log”, “Second argument”) FS-8172 [mod_conference] Fixed broken admin controls in verto demo app caused by adding verto_dvar_ prefixed variables to the json status even when we have json status disabled FS-8173 [core] Properly respond RTP/AVPF to an AVPF offer instead of assuming its secure and responding with SAVPF FS-8178 [verto_communicator] Define first share device selected by default in settings modal FS-8078 [verto_communicator] Fixed display flex in safari FS-8180 [verto] Fixed an issue when disabling video on the creation of a dialog, video can still mistakenly be enabled causing some issues with the SDP still offering video FS-8155 [verto_communicator] Fixed issue with not detecting hangup when using uuid_kill or fsctl hupall calls FS-8146 [verto_communicator] Fixed display of long names in members list FS-8181 [verto] Fixed failed init if a camera isn’t available FS-8184 [mod_conference] Fixed possible memory leak when hanging up on a video call FS-8185 [core] Allow xml preprocessor to expand variables where the resulting value is much longer than the original size And, this past week in the FreeSWITCH 1.4 branch we had 7 new commits merged in from master. And the FreeSWITCH 1.4.21 release is here! Go check it out! Security Issues: FS-8160 Fixed head overflow in json parser, properly handle malformed json when parsing json with \u at the end of a json string. Thank you to Marcello Duarte who discovered and reported this issue. New features that were added: FS-8175 Added continue_on_answer_timeout variable to allow channel to proceed from a tripped answer timeout The following bugs were fixed: FS-8149 [mod_xml_cdr] Fixed curl dependency in makefile FS-8147 [mod_erlang_event] Fixed the process spawning segfault FS-8140 [mod_sofia] Fixed a user_name typo in sofia_handle_sip_i_invite FS-8131 [mod_voicemail] Fixed issues with allowing an empty password change and then locking out the user FS-1772 [mod_voicemail] Fixed the reset of voicemail greeting to default to allow entering 0 to restore the default greeting FS-8143 [mod_rayo] Fixed a crash caused by client disconnecting from mod_rayo while a message is being delivered to that client. This is caused by the XMPP context’s JID -> XMPP stream mapping not being cleaned up on XMPP stream destruction.

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Call a remote extension on specific domain
by Tanguy 21 Sep '15

21 Sep '15

Hello I have two multi-domains freeswitch servers 1 and 2 and i want to interconnect them for internal calls I read https://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes but it did not fully fit to my need because I want to call from freeswitch1 a specific extension of freeswitch2 on a specific domain ( and not just default ). In short i want to tell to freeswitch2: Connect me with extension 1000@domain-name and not just connect me with extension 1000 My guess is send custom sip headers from freeswitch1 and parse X-domainname on freeswitch2 dialplan <action application="set" data="sip_h_X-domainname=somedomain"/> <action application="bridge" data="sofia/internal/${destination_number}(a)freeswitch2.company.com "/> This will probably work, but is there a clean way to do it without using non standard headers ? Best Regards

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