September 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

get events data on freeswitch console
by Shahzad Bhatti 27 Sep '12

27 Sep '12

hi, i want to get the events data without ESL in lua and with the following: freeswitch.consoleLog("info",params:serialize()) but unable to any output but* ERROR* 2012-09-27 01:21:47.024894 [ERR] mod_lua.cpp:198 /usr/local/freeswitch/scripts/ESLInLua.lua:5: attempt to index global 'params' (a nil value) stack traceback: /usr/local/freeswitch/scripts/ESLInLua.lua:5: in main chunk reply me how i can get that. Regards Shahzad Bhatti

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Re: [Freeswitch-users] Sometimes missing a few seconds of audio when CN is offered in INVITE (and solution)
by Peter Olsson 27 Sep '12

27 Sep '12

Tony, your change fixed the problem! Thank you so much :) /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Anthony Minessale Skickat: den 26 september 2012 21:28 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Sometimes missing a few seconds of audio when CN is offered in INVITE (and solution) Looking at the code, I can see a solution assuming your theory is correct. I pushed a patch to HEAD if you care to test it. On Wed, Sep 26, 2012 at 3:26 AM, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: Hello everyone! I experience a strange issue for bridged calls from Lync -> FreeSWITCH -> Asterisk, it seems to be related to CN packets. I'm not sure if this a bug or not so I ask the question here first. The problem is that the audio back from the Asterisk server is not bridged back to the originating client for the first few seconds (how long is different on different calls), instead FS seems to send CN-packets back. I've looked into a wireshark dump, and I see the audio is coming from the Asterisk server (using the correct IP and port), but FS doesn't write the same packet to the other call leg (back to Lync), instead it sends a CN packet every second or so. My guess is that the timestamp is being handled wrong somehow, and FS believes that the real packet is too old (since it has already sent a CN packet with a higher timestamp?), and should not be written to the other leg, then after a while it sends a new CN packet and so on. The solution is to set "suppress_cng=true" before bridging the call to the Asterisk server, when this is done, the audio is always bridged correctly, and nothing is missing. I have pcaps of both working calls (with the variable set) and nonworking calls, so if you believe it might be something that FS should handle differently I'll submit this to Jira. /Peter _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 !DSPAM:5063553532761904613329!

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Freeswitch HA issue with failover to second node
by Виталий Давудов 27 Sep '12

27 Sep '12

Hi, list! I've installed Freeswitch HA cluster for failover to second node. It use corosync and pacemaker to failover IP-address and Mysql Master-Master replication. After rebooting the first node, the second node successfully recover calls, but appears problem with one-way media. Channel with PSTN-gateway (throught external profile) is established in 2-3 secs. Channel with registered user (throught internal profile) is established only after ~30 secs! Internal and external profiles listening to the same ip-address. I think sip-packets blocked by FS acl. When I do tcpdump, packets from FS arrive to the user ip-phone, but sip-packets from ip-phone didn't arrive to FS. This is from user directory: <user id="111" cidr="172.30.0.11/32"> <params> <param name="password" value="bla-bla-bla"/> </params> <variables> <variable name="user_context" value="private-vip-telecom-service"/> <variable name="effective_caller_id_number" value="4959894700"/> <variable name="outbound_caller_id_number" value="4959894700"/> <variable name="callgroup" value="support"/> </variables> </user> In sip-profile named "internal" enabled only one acl: <param name="apply-inbound-acl" value="domains"/> In acl.conf.xml: <list name="domains" default="deny">  <node type="allow" domain="172.30.0.1"/>   </list> And from FS console: freeswitch@internal> acl 172.30.0.11 domains true Here you can see FS console log from second node after sip recover: http://pastebin.freeswitch.org/19941 If anybody has ideas how to resolve this issue, I'll be very happy! -- Best regards, Vitaly.

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Recording problem speed and noise
by Tiago N. Sampaio 27 Sep '12

27 Sep '12

Hi All! Im facing a strange issue.. My recordings (in mp3, wav or gsm) are too fast.. now recordings are completely inaudible, only with noise and voice metallized... But sometimes it not happen, just speed issue... im running this version: FreeSWITCH Version 1.2.0-rc2 (1.2.0-rc2) today i tryed to change to 1.3.0, but faced another problem, noise on calls. i checked load on my machine, and load was 0.2, almost 100% cpu FREE.. i have a khomp board with R2 signaling.. Can you help me to debug what is happening? Im quite new to FS!!! Hugs

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Sometimes missing a few seconds of audio when CN is offered in INVITE (and solution)
by Peter Olsson 26 Sep '12

26 Sep '12

Hello everyone! I experience a strange issue for bridged calls from Lync -> FreeSWITCH -> Asterisk, it seems to be related to CN packets. I'm not sure if this a bug or not so I ask the question here first. The problem is that the audio back from the Asterisk server is not bridged back to the originating client for the first few seconds (how long is different on different calls), instead FS seems to send CN-packets back. I've looked into a wireshark dump, and I see the audio is coming from the Asterisk server (using the correct IP and port), but FS doesn't write the same packet to the other call leg (back to Lync), instead it sends a CN packet every second or so. My guess is that the timestamp is being handled wrong somehow, and FS believes that the real packet is too old (since it has already sent a CN packet with a higher timestamp?), and should not be written to the other leg, then after a while it sends a new CN packet and so on. The solution is to set "suppress_cng=true" before bridging the call to the Asterisk server, when this is done, the audio is always bridged correctly, and nothing is missing. I have pcaps of both working calls (with the variable set) and nonworking calls, so if you believe it might be something that FS should handle differently I'll submit this to Jira. /Peter

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FreeSWITCH Conf Call Today
by Michael Collins 26 Sep '12

26 Sep '12

Don't forget about today's call! http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_26 We get to have a deeper technical conversation with Scott and Daniel regarding the repro SIP proxy. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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intercom=true on originate bleg
by Gerald Weber 26 Sep '12

26 Sep '12

Hi, is there a reason why intercom=true is sent to the bleg (2001) with originate [sip_auto_answer=true,execute_on_media='set sip_auto_answer=']user/2000 2001 (from http://jira.freeswitch.org/browse/FS-3223) or originate [sip_auto_answer=true]user/2000 &bridge([sip_auto_answer=false]user/2001) (from http://lists.freeswitch.org/pipermail/freeswitch-users/2012-January/079524.…) ? 2000 is a snom phone which auto answers correctly, 2001 is phonerlite. I thought its possible to turn that off with sip_auto_answer=false Wireshark trace is here http://pastebin.freeswitch.com/19938 Sofia loglevel all 9 and tracelevel debug is here http://pastebin.freeswitch.com/19939 I'm on latest git FreeSWITCH Version 1.3.0+git~20120925T211258Z~817439d76a (git 817439d 2012-09-25 21:12:58Z) Thx gw

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Video Calls over Freeswitch
by Tahir Almas 26 Sep '12

26 Sep '12

I will appreciate any help regarding video call support in Freeswitch and how it can be implemented using Andriod phones ? *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT

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CLI Debug request?
by David Villasmil 26 Sep '12

26 Sep '12

Hello All, It would be VERY nice to be able to set a "fsctl loglevel debug" on a specific "called number", maybe by looking at the cldnum and filtering all output but the uniqueID? And by this I mean Sofia as much as ANY log information happening ONLY for that specific call.. When one has high loads it is extremely difficult to debug a specific situation... Maybe this is already possible, but I don't know how? Thanks! David

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simultaneous calls
by Zaid Tillma 26 Sep '12

26 Sep '12

My goal is to have two phone announce a recording (in a wav file). I have accomplished goal in the following manner: I have an extension configured like this: <extension name="wavs"> <condition field="destination_number" expression="^2009$"> <action application="playback" data="/path/to/your.wav"/> </condition> </extension> When you call, you get an audio playback. I am trying to originate a call simultaneously to two separate phone (i.e want to two phone to play audio) at the same time, and I am doing it like this (and it works): /usr/local/freeswitch/bin/fs_cli -x 'originate {ignore_early_media=true,sip_auto_answer=true}user/1000 2009' /usr/local/freeswitch/bin/fs_cli -x 'originate {ignore_early_media=true,sip_auto_answer=true}user/1001 2009' There's got to be a better way to do this way without having two seperate originates.. I did try the Enterprise originate, but only one phone answered... Not sure if there was a mistake in my Enterprise originate implementation. Please let me know if there is a more elegant way to do this..

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