January 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Quick Question
by Magnus Kelly 13 Jan '17

13 Jan '17

Hello All, would be very helpful if someone could help me with quick reminder on how to change SIP header on bridge as in - incoming call on LegA Bridged to Leg B while changing the "ip:FreeSWITCH" to sip: +44xxxxxxxxxx now this format appears as default - From: "+44xxxxxxxxxx" <sip:FreeSWITCH@10.11X.X.XX:5068;transport=tcp>;tag= UB7tNX0mmBt7j And I would like help to change it to be in this format - From: "+44xxxxxxxxxx " <sip: +44xxxxxxxxxx@10.11X.X.X:5068; transport=tcp>;tag=UB7tNX0mmBt7j Thanks

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multiple profiles and user not registered
by David Villasmil 12 Jan '17

12 Jan '17

hello guys, i have 2 profiles for two different ip addresses (IP1 and IP2), both profiles have force-register-domain and force-register-db-domain to the IP1, i can register on IP2 but when calling this user like user@IP1, fs returns USER_NOT_REGISTERED... i thought by forcing the register domain i could call the user to any of the 2 ips... any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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B-leg for each user
by Murugan Pandian 12 Jan '17

12 Jan '17

I try get each user cdr from goup call.its work fine if user are sip when i make call user pstn using loopbackm i am getting only one leg event all user busy

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Audio cut off at the begin of the verto call to sip external voicemail
by José Lopes 12 Jan '17

12 Jan '17

Hello Guys, I have audio cut off at the begin of the verto call to FreeSwitch that redirect to sip external voicemail (Access voicemail mailbox) . This happen when I use PCMU at verto codecs and sip codecs (if i use opus at verto codecs, there is no issue, but this causes audio transcoding) . At dialplan i used the example "Bridging from WebRTC (mod_verto) to PSTN/ITSPs" from https://freeswitch.org/confluence/display/FREESWITCH/mod_verto. I notice if i remove the playback action, there is no issue. But I need the playback action to send rtp packets to verto client. I simulate this using another FreeSwitch as external voicemail server and I only listen "id followed by pound" from the initial message of voicemail ("Please enter your id followed by pound"). The log of this call is at https://pastebin.freeswitch.org/view/507fa115 What I can do to use PCMU at verto codecs and sip codecs on type of call? Should i open a issue on FreeSwitch JIRA ? Best regards, Jose Lopes

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Merging two calls
by Richard A. Branham, Jr. 12 Jan '17

12 Jan '17

I am trying to merge 2 separate calls together into a conference or 3-way call while maintaining the ability to put each one on/off hold or hang up without affecting the other. Using Verto in a web app, the usual call flows to FS extensions and the PSTN are working. The user can put an existing call on hold and make another call. If a call comes in and the user answers it, the first call is placed on hold and the second one becomes active. I have implemented various switch commands using ESL and the UUIDs of calls and channels can be retrieved. >From here we'd like to place all 3 into a conference or 3-way call while enabling the "main" user to place the other calls on/off hold or hang them up individually. Is it possible to merge 2 separate calls into 1 while maintaining the same control over the calls as when they're just 1-to-1? Thanks! Richard

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Postgresql in fs core: recommendations
by Antonio Silva 11 Jan '17

11 Jan '17

Hi, On debian jessie, i've switch from using sqlite in the core with db located at /dev/shm to postgresql server running on the same box, but now for the first time, i see SIP messages in FS "503 Server Busy" and losing endpoint registration with expire, i think the delay added by pg to process the queries affects fs main process to handle huge number of sip messages per second. In terms of fs i have: - call average 6 cps - avg 100 registers / sec (the expire value configure in the accounts is of 300 seconds) - avg 150 options / sec (max of 400 pkt/sec) - avg 50 notify / sec (max of 100 pkt/sec) In fs configurations i've: max-db-handles = 100 max-sessions = 2000 sessions-per-second = 50 My server i think is quite power to handle this traffic: 2 x Intel(R) Xeon(R) CPU E5-2620 v4 @ 2.10GHz 32GB of ram sata disk 7200RPM I was wonder if any of you have this issues and if you could share your pg configuration? For now i i've try for pg: max_connections = 100 ssl = false shared_buffers = 5GB dynamic_shared_memory_type = posix Thanks, -- Saludos / Regards / Cumprimentos, António silva

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Speech recognition SpeechTools.jm Error
by Gonzalo Gasca Meza 11 Jan '17

11 Jan '17

Hi all, I'm trying to install Speech Recognition engine. Version: FreeSWITCH (Version 1.6.13 -21-e755b43 64bit) Following this example: https://wiki.freeswitch.org/wiki/Mod_pocketsphinx <include> <extension name="pizza_demo"> <condition field="destination_number" expression="^(pizza|74992)$"/> <condition field="${module_exists(mod_v8)}" expression="true"/> <condition field="${module_exists(mod_pocketsphinx)}" expression="true"> <action application="javascript" data="ps_pizza.js"/> </condition> </extension> </include> I'm replacing mod_spidermonkey with mod_v8 based on this document and when I call this number I get: EXECUTE sofia/external/14088053951(a)104.236.190.206 javascript(ps_pizza.js) 2017-01-10 03:04:39.099748 [ERR] SpeechTools.jm:191 Exception: SyntaxError: Unexpected identifier (near: " if (result.interpretation.@confidence >= grammar_object.min_score) {") If I change: result.interpretation.@confidence to result.interpretation.['@confidence'] EXECUTE sofia/external/14088053951(a)104.236.190.206 export(RFC2822_DATE=Tue, 10 Jan 2017 03:13:10 -0500) 2017-01-10 03:13:10.139816 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [RFC2822_DATE]=[Tue, 10 Jan 2017 03:13:10 -0500] EXECUTE sofia/external/14088053951(a)104.236.190.206 javascript(ps_pizza.js) 2017-01-10 03:13:10.159763 [ERR] SpeechTools.jm:191 Exception: SyntaxError: Unexpected string (near: " if (result.interpretation.['@confidence'] >= grammar_object.min_score) {") SpeechTools.jm from: https://raw.githubusercontent.com/lordnull/FreeSWITCH/master/scripts/javasc… Similar discussion: http://lists.freeswitch.org/pipermail/freeswitch-users/2014-June/105818.html Any pointers? I get the mod_v8 module compiled from Freeswitch using this: https://freeswitch.org/confluence/display/FREESWITCH/mod_v8 Thanks

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Freeswitch and JSSip, getting 488
by Tayeb Meftah 10 Jan '17

10 Jan '17

Hey guys it have been a long time;) i am having an issue with JSSip and Freeswitch (WSS) lovely no signed certificate... ;) the issue is i am getting 488 sip traces: http://paste.debian.net/908067/ thanks!

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Gateway outbound configuration
by Murugan Pandian 10 Jan '17

10 Jan '17

Hai, I try to use reliance sip trunking directly VOIP client(zoiper) its work fine ,but when i try to use with freeswitch gateway i am getting always fails bellow is my siptrunking details. Details: Username:+914146380100 password:12345 proxy: 10.237.246.225 domain: bangalore.relianceims.in Auth username:+914146380100@bangalore.relianceims.in *My Gateway XML* <include> <gateway name="reliance"> <param name="username" value="+914146380100"/> <param name="password" value="123456"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="realm" value="bangalore.relianceims.in"/> <param name="proxy" value="10.21.254.231"/> <param name="ping" value="25"/> </gateway> </include>

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Regex Nothing to repeat Dialplan issue
by Murugan Pandian 10 Jan '17

10 Jan '17

HI, When i receive inboung call from , iam getting dialplan condition error , Compile error: 1 [nothing to repeat] inbound number contain + like +91414.

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