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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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Freeswitch install from yum
by John Nash 09 Jan '17

09 Jan '17

I installed freeswitch using yum commands as per given https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 All seems to be OK but I have two questions .. 1- I do not find ulimit settings in any startup file as recommend . I see files /etc/sysconfig/freeswitch, /etc/systemd/system/multi-user.target.wants/freeswitch.service am I supposed to create some kind of init script and mention ulimit settings there? 2- In my box I see lua version as Lua 5.1.4 Copyright (C) 1994-2008 Lua.org, PUC-Rio . Is higher version of lua recommended? Any memory related issue with 5.1.4 version?

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tls with letsencrypt
by ITwrx.org 06 Jan '17

06 Jan '17

hi, i'm trying to use a letsencrypt generated cert with freeswitch but am not sure how to proceed. I've read the old and new wiki posts concerning tls but they don't seem to cover my exact scenario. It seems to me that freeswitch is looking into the configured "tls-cert-dir" for the hardcoded filename tls.pem and is expecting that a self generated ca has signed it. i have placed the fullchain.pem in that directory (generated with certbot) and have renamed it tls.pem but i guess it's not finding the CA sig it expects(?) as i'm getting: tport_tls.c:1044 tls_connect() tls_connect(0x373c000e8d0): TLS setup failed (error:00000005:lib(0):func(0):DH lib) when trying to connect with csipsimple from phone. I would like to avoid generating client certs signed by a custom CA where users have to copy the client cert and ca cert to their device as it adds complexity and problems. Is there a workaround or suggested method for using a letsencrypt cert with freeswitch so that clients like csipsimple can just validate against their built-in CA store? thanks in advance, ITwrx -- Information Technology Works https://ITwrx.org @ITwrxorg

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Re: [Freeswitch-users] tls with letsencrypt
by Kim Culhan 06 Jan '17

06 Jan '17

On Thu, Jan 5, 2017 at 2:55 PM, ITwrx.org <info(a)itwrx.org > <https://webmail.kim.net/src/compose.php?send_to=info@itwrx.org>> wrote: > >> hi, >> >> i'm trying to use a letsencrypt generated cert with freeswitch but am >> not sure how to proceed. > > Suggest you proceed to take a look at the docs here: https://wiki.freeswitch.org/wiki/SIP_TLS Steps 1 & 2 involve making your own certificate; you don't have to use letsencrypt certs. Start with Step 3 and proceed from there, as Bryan mentioned there is a lot more to it than '..just copied the pem formatted cert that certbot generated to /etc/freeswitch/tls and named it tls.pem.' regards -kim

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