- FreeSWITCH-users - FreeSWITCH Mailing Lists

Add canditate to the SDP
by Alexander Haugg 10 Sep '15

10 Sep '15

Hi All, what can I do, that mod_sofia add a candidate to the SDP for an outbound leg? For an inbound leg with candidates in the SDP the mod_sofia answer correct with the favor candidate in the SDP. Thanks a lot!

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systemd won't enable for automatic start of FS
by Bote Man 09 Sep '15

09 Sep '15

I'm struggling to get systemd to start FreeSWITCH at boot time and also leave it running once it starts. Problem #1 - "systemd enable freeswitch" outputs the right steps, but ends with an error message. The error I get is "No such file or directory" but that is less than helpful. Problem #2 - When I start FreeSWITCH with "systemd start freeswitch" it starts alright, but systemd does not detect FS running properly so it kills it after the specified 45 seconds elapses and restarts it. I can see this in top when FS gets a new pid. Details -------- It's a new installation of Debian 8.1 and I chose to use systemd as the init process. I compiled FreeSWITCH from source using the latest Master as of late July. The only section of the unit file that is supposed to be used by the "enable" command is the [Install] section and that simply points to the multi-user.target as expected. I do not know what to do with the 2 other sample systemd files in /usr/src/freeswitch/debian, perhaps they are key. I have read up on systemd and the FS unit file looks good to my untrained eye. I did not use the sysvinit files that are also in that FS debian directory. I successfully used systemd to start, stop, and enable a few other system services with success so there must be a tweak to the FS unit file that I am missing. I can start FS manually and it runs normally. A Google search revealed a number of complaints that systemd would emit an error message with its "enable" command, yet it would successfully enable the service; this was with a few different services on Debian and Ubuntu, so it seems there might still be some weirdness with systemd on Debian. Has anybody gotten systemd to start FS reliably? Has anybody figured out how to make systemd detect FreeSWITCH running so that it doesn't restart it repeatedly? Once I figure out how to tame systemd I will document my findings in Confluence as these are the last steps that allow you to get FS up and running quickly. Thanks. --- Bote

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FreeSWITCH Week in Review (Master Branch) August 29th- September 4th
by Ken Rice 09 Sep '15

09 Sep '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1hYdrxA FreeSWITCH Week in Review (Master Branch) August 29th- September 4th Hello, again. This past week in the FreeSWITCH master branch we had 75 commits! The new features for this week include: improvements to the verto communicator, allowing Freeswitch to initiate late offer calls, the addition of CUSTOM esl events menu::enter and menu::exit when a call enter and exit an ivr menu, and more work toward the new module, mod_hiredis! 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. New features that were added: FS-8094 [verto_communicator] Added googEchoCancellation, googNoiseSuppression, and googHighpassFilter settings to the UI enabled by default FS-8107 [switch_ivr_menu] Adding CUSTOM esl events menu::enter and menu::exit when a call enter and exit an ivr menu FS-6833 Allow Freeswitch to initiate late offer calls FS-8075 [mod_hiredis] Add conf file to spec file too and updates for limit release case Improvements in build system, cross platform support, and packaging: FS-7966 [Windows] Build now uses visual studio 2015 and builds successfully, but is still missing video functionality FS-8124 [verto_communicator] Adding option –debug to grunt build, dist app will be generated without minified code. The following bugs were squashed: FS-8099 [mod_lua] Restored LUA dialplan ACTIONS functionality FS-7988 [filebug.pl] Moved -askall to -terse so old -askall behavior is default and old default is now -terse FS-8029 [jitterbuffer] Fixed robotic sound when using jitterbuffer FS-8103 [mod_rayo] Fixed handling of malformed requests FS-8108 [mod_lua] Removed legacy mod_lua, the regular mod_lua works with system lua now FS-8082 [mod_rayo] Fixed a segfault FS-8110 [mod_rayo] Prompt IQ error reply was being deleted after being sent for delivery. This is incorrect since message delivery thread will clean up the message FS-8116 [verto] Fixed device enumeration hanging on init FS-8118 [verto] Fixed calls not properly rejecting video when video is offered but only audio is accepted FS-7994 [verto_communicator] Using factory for group calls in history FS-8117 [verto_communicator] Calling verto.iceServers upon useSTUN changing on ModalSettings, correctly check for STUN setting in localStorage, and fixed ignoring useSTUN settings FS-8127 [mod_conference] Fixed an audio issue caused by the codec not updating often enough when detecting rate change FS-8088 [verto_communicator] Fixed members list being destroyed after switching conferences and ending the first conference And, this past week in the FreeSWITCH 1.4 branch we had 9 new commits merged in from master. And the FreeSWITCH 1.4.21 release is here! Go check it out! New features that were added: FS-7752 [mod_rayo] Increase maximum number of elements from 30 to 1024 to allow adhearsion to create large grammars to navigate IVR menus. Improvements in build system, cross platform support, and packaging: FS-8055 [build] Add confdir variable to freeswitch.pc The following bugs were fixed: FS-7135 [mod_sofia] Fixed response to re-invite with duplicate sdp (such as we get from session refresh) when soa is disabled to include an sdp. Fixed t.38 fax failure on session refresh FS-7903 [proxy_media] Fix Codec PROXY Exists but not at the desired implementation. 0hz 0ms 1ch error when using proxy media. FS-8056 [mod_voicemail] Fixed a segfault on vm_inject, regression from FS-7968 FS-7968 [mod_voicemail] Fixed verbose events FS-8110 [mod_rayo] Prompt IQ error reply was being deleted after being sent for delivery. This is incorrect since message delivery thread will clean up the message FS-8082 [mod_rayo] Do not remove items from hash while iterating FS-8103 [mod_rayo] Handle where finishes unexpectedly before start event is received

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Re: [Freeswitch-users] No Audio with Skypopen
by Lyle Pratt 09 Sep '15

09 Sep '15

@giovanni Sure I can open a Jira issue, but at this point, I'm kind of just wondering if its working for anyone else. Basically we only have 1 Skype user configured on only 1 interface. Adding that user as a contact from any other Skype user just stays in the Pending state. Is it working for you? ---------- Forwarded message ---------- From: Giovanni Maruzzelli <gmaruzz(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Cc: Date: Wed, 9 Sep 2015 19:07:27 +0200 Subject: Re: [Freeswitch-users] No Audio with Skypopen No, there is no force on that. Can you please open a jira issue with all the information needed to replicate the problem? Thanks! sent from my mobile, Giovanni Maruzzelli cell: +39 347 266 56 18

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Re: [Freeswitch-users] No Audio with Skypopen (Oz Mortimer)
by Lyle Pratt 09 Sep '15

09 Sep '15

Adding the changes to /etc/pulse/system.pa that Oz suggested solved the problem for us! Thanks Oz! We are currently building a docker container with everything working now, hosted here: https://hub.docker.com/r/bettervoice/freeswitch-container/tags/ It should be finished building in a few minutes. Tag will be "1.6.0.skype". Many Thanks, Lyle ---------- Forwarded message ---------- From: Oz Mortimer <omortimer(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Cc: Date: Wed, 9 Sep 2015 16:55:06 +0100 Subject: Re: [Freeswitch-users] No Audio with Skypopen Hi Giovanni, I had it running on 2 quite old machines Fedora release 16 (Verne) / Linux 3.6.11-4.fc16.i686 i686 CentOS release 5.5 (Final) / Linux 2.6.18-194.el5 x86_64 Both were on v1.4.9 It was quite a while when I tested it but I did have exactly the same issue as described. — /etc/pulse/system.pa load-module module-null-sink load-module module-native-protocol-unix auth-anonymous=1 — /etc/pulse/daemon.conf daemonize = yes system-instance = yes resample-method = trivial flat-volumes = yes default-sample-rate = 48000 default-sample-channels = 1 default-fragment-size-msec = 20 and because I kept forgetting to start things, I added these 2 lines to the freeswitch start up script. /usr/bin/pulseaudio --system=true --daemonize=true --log-level=info --log-target=syslog & sh /usr/local/freeswitch/skypopen/skype-clients-startup-dir/start_skype_clients.sh I had 5 clients running with no issues what so ever. I can’t for the life of me remember where I found the auth-anonymous=1 command, but it certainly resolved the issue. Thanks Oz. On 9 Sep 2015, at 16:38, Giovanni Maruzzelli <gmaruzz(a)gmail.com> wrote: Oz, so nice from you to share your knowledge. Can you please paste here the entire file you modified, and give some more details (distro, version, FreeSWITCH version, etc). I will use all info to update Confluence (our documentation). Thanks again, -giovanni On Wed, Sep 9, 2015 at 5:33 PM, Oz Mortimer <omortimer(a)gmail.com> wrote: > for what its worth, I had to add auth-anonymous=1 to system.pa > load-module module-native-protocol-unix auth-anonymous=1 > which resolved the issue for me. > > > On 9 Sep 2015, at 16:26, Giovanni Maruzzelli <gmaruzz(a)gmail.com> wrote: > > Hello Lyle, > > for mod_skypopen, have you followed exactly, step by step, the > instructions in > https://freeswitch.org/confluence/display/FREESWITCH/mod_skypopen ? > > Also, please give us more details, as in: > https://freeswitch.org/confluence/display/FREESWITCH/mod_skypopen#mod_skypo… > > -giovanni > > On Wed, Sep 9, 2015 at 4:58 AM, Lyle Pratt<lylepratt(a)gmail.com> wrote: > >> Hi Guys, >> >> We've been trying to get Skypopen up and running. Everything is working >> well so far, except that we have no audio. :( >> >> It seems to be a Pulseaudio issue, but Pulseaudio is installed and the >> daemon is running. FS console reports tons of entries like this: >> >> 2015-09-09 02:47:35.541712 [DEBUG] mod_skypopen.c:900 >> [1.6.0+git~20150903T203652Z~6762f14140~64bit ] [DEBUG_SKYPE 900 >> ][skype101 ][UP,INPROGRS] READ BUFFER EMPTY,skypopen_audio_read >> Silence >> 2015-09-09 02:47:35.561715 [DEBUG] mod_skypopen.c:900 >> [1.6.0+git~20150903T203652Z~6762f14140~64bit ] [DEBUG_SKYPE 900 >> ][skype101 ][UP,INPROGRS] READ BUFFER EMPTY,skypopen_audio_read >> Silence >> 2015-09-09 02:47:35.581714 [DEBUG] mod_skypopen.c:900 >> [1.6.0+git~20150903T203652Z~6762f14140~64bit ] [DEBUG_SKYPE 900 >> ][skype101 ][UP,INPROGRS] READ BUFFER EMPTY,skypopen_audio_read >> Silence >> >> We are running Ubuntu 14.04. Has anyone encountered this? After quite a >> lot of searching I've not been able to find anyone with similar issues. >> >> Thanks, >> Lyle >> >> _________________________________________________________________________ >> 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 >> > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _________________________________________________________________________ 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 _______________________________________________ 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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No Audio with Skypopen
by Lyle Pratt 09 Sep '15

09 Sep '15

Hi Guys, We've been trying to get Skypopen up and running. Everything is working well so far, except that we have no audio. :( It seems to be a Pulseaudio issue, but Pulseaudio is installed and the daemon is running. FS console reports tons of entries like this: 2015-09-09 02:47:35.541712 [DEBUG] mod_skypopen.c:900 [1.6.0+git~20150903T203652Z~6762f14140~64bit ] [DEBUG_SKYPE 900 ][skype101 ][UP,INPROGRS] READ BUFFER EMPTY,skypopen_audio_read Silence 2015-09-09 02:47:35.561715 [DEBUG] mod_skypopen.c:900 [1.6.0+git~20150903T203652Z~6762f14140~64bit ] [DEBUG_SKYPE 900 ][skype101 ][UP,INPROGRS] READ BUFFER EMPTY,skypopen_audio_read Silence 2015-09-09 02:47:35.581714 [DEBUG] mod_skypopen.c:900 [1.6.0+git~20150903T203652Z~6762f14140~64bit ] [DEBUG_SKYPE 900 ][skype101 ][UP,INPROGRS] READ BUFFER EMPTY,skypopen_audio_read Silence We are running Ubuntu 14.04. Has anyone encountered this? After quite a lot of searching I've not been able to find anyone with similar issues. Thanks, Lyle

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Re: [Freeswitch-users] No Audio with Skypopen
by Lyle Pratt 09 Sep '15

09 Sep '15

@giovanni I am actually having one other issue. The documentation states that the skype client should automatically accept any contact requests. However, when I add the contact from another account, it always shows as "Pending contact request". Is there some way to force that to accept? ---------- Forwarded message ---------- From: Giovanni Maruzzelli <gmaruzz(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Cc: Date: Wed, 9 Sep 2015 18:33:08 +0200 Subject: Re: [Freeswitch-users] No Audio with Skypopen (Oz Mortimer) super! Power of collaboration ! -giovanni

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Cleanup destination_number with country code
by Michael Nielsen 09 Sep '15

09 Sep '15

I would like to check if the called number contains a country code, if not then one should be added. I've got a variable in my user's XML file called country_code containing +44. This is my code so far, however, I think the last to conditions overlap: <condition field="destination_number" expression="^(\+\d{3,20})$"> <action application="export" data="dialed_extension=$1"/> </condition> <condition field="destination_number" expression="^00(\d{3,20})$"> <action application="export" data="dialed_extension=+$1"/> </condition> <condition field="destination_number" expression="^(\d{3,20})$"> <action application="export" data="dialed_extension=${country_code}$1"/> </condition> I would like to change 00 to +, and if no + is on the number then use country_code from the user. If it does contain a +, then just use the dialled number.

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Add country code if it aren't dialled.
by Michael Nielsen 09 Sep '15

09 Sep '15

I would like my dial plan in FS to automatically add a country code (from the users variable "country_code") if no country code has been dialled. This is part of my dial plan: <condition field="destination_number" expression="^\+(\d{3,20})$"> <action application="export" data="dialed_extension=$1"/> Is it possible to automatically check if either + or 00 is dialled, if not then add + <user country_code>. If 00 is dialled then this should be replaced with a +.

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Using dialplan routing from ESL?
by René Weiss 09 Sep '15

09 Sep '15

Hi, I have set up some simple dialplan rules like: - Numbers with 4 digits or shorter: internal calls - 5 digits or longer: use external sip gateway This works perfectly fine with our (soft-)phones. Now I would also like to use the same rules when I make calls from our software over the ESL, but it seems that I always have to tell the "originate" command which gateway to use. Is there a syntax for "originate" (or another command) which uses the dialplan rules to decide which gateway to use, or do I have to implement the dialplan rules a 2. time in the software? Thanks, René

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