October 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

rtp issues with mod_dingaling
by Ubuntu Rocks 07 Oct '11

07 Oct '11

Hi All, I am trying to get mod_dingaling working with gtalk. I have configured extention 1004 in jingle_profiles/client.xml. I make a call from another gtalk user,the call lands up fine at the extention I have configured.But there is no audio.I searched a lot and have tried everything I found on the net including using different STUN servers, using different values for candidate-acl,disable-rtp-auto-adjust. etc. I enabled debugging in mod_dingaling and found in the xmpp messages that along with a local RTP ,and a stun RTP a relay RTP port is being advertized by the google server,I ran wireshark on my freeswitch server and looks like the audio is going to the stun RTP port not to the realy RTP port. Can this be a problem? Does gtalk want the audio to be sent directly to the concerned gtalk client or does it have to go through the google server? The FreeSwitch vesrion I am using is ' FreeSWITCH version: 1.0.7 (svn-exported)'. Please find below my jingle_profiles/client.xml. <include>   <profile type="client"> <param name="name" value="jclient"/> <param name="login" value="XXXXXXXX(a)gmail.com/gtalk"/> <param name="password" value="XXXXX"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="auto-login" value="true"/>  <param name="sasl" value="plain"/>  <param name="server" value="talk.google.com"/>  <param name="tls" value="true"/>  <param name="use-rtp-timer" value="true"/> <param name="disable-rtp-auto-adjust" value="true"/>  <param name="exten" value="1004"/>    <param name="vad" value="both"/>  <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include> I will appreciate any help or pointers on this. -- Thanks Saurabh

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mod_native_file and speex
by Cliff Wells 07 Oct '11

07 Oct '11

I'm using Speex to bridge calls between an out-of-state Asterisk box and a local FreeSWITCH IVR. This works great, but I'm having to do needless transcoding on the FS system to play the IVR .wav files. I tried using speexenc to generate speex versions of the files but the audio is garbled (sounds like a screech). No errors or warnings are generated on the console when playing the file. I'm only using 8000Hz, mono files and generate the speex file with -n (narrowband) argument. I believe speexenc generates the files in an ogg container, so perhaps this is the problem, but I'm not sure how to get around it if it is. In any case, if anyone knows of a recipe for creating speex files that mod_native_file can use, I'd appreciate a pointer. Regards, Cliff

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playing files through a remote URL
by Holger Esser 06 Oct '11

06 Oct '11

Hello, I am trying to write a simple ivr script that answers the call, plays a greeting and collects a few digits. Is it possible to fetch the audio through a remote URL (i.e. http://messageServer/en.us/greeting.wav<http://messageserver/en.us/greeting.wav> )? Or does it have to reside on the local file system or a mapped drive? Kind regards, Holger

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Sofia ODBC causes multiple b-legs on a single call
by R H 06 Oct '11

06 Oct '11

Hey Everyone, I am preparing to set up HA Failover to my production freeswitch environment. In preparation for this I have been doing some testing in a development environment and I encountered the following problem: I currently have two sip-profiles set up that I am using full time, when I enable odbc in the switch.conf.xml everything works correctly. When I enable odbc in ONE of the sip-profiles everything works correctly. But when I enable odbc in the second profile I begin to see two b-legs for ever call that I initiate. Both b-legs are directed at the same client. Does anyone have any idea why enabling odbc on two sofia profiles would cause every inbound call to generate two outbound calls? I also decided to test this problem by adding two more profiles for a total of 4. With 4 profiles I get 4 b-legs. What am I doing wrong? Ryan

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callcenter: getting the number of users waiting
by Yehavi Bourvine 06 Oct '11

06 Oct '11

Hello, I am implementing a callcenter using the mod_callcenter. I would like the user to hear some text/music while waiting and from time to time to announce his/her place in the queue. Is it possible by some simple mean, or do I have to use an external application with uuid_broadcast to do that? Is there a table that holds the list of currently waiting incoming calls? Thanks, __Yehavi:

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Playing with mod_sms
by John Rose 06 Oct '11

06 Oct '11

It's awesome that we have this module now. Kudos to all those that worked on it. I'm going to start playing with this a bit. Currently I'm using some of the SIP MESSAGE functionally with a tweaked Freeswitch 1.0.6+ sofia_presence.c to receive SIP MESSAGE ESL events from unsolicited addresses and have to figure out somehow on confirming that 200 OK's were received on MESSAGE send and not failures, a 404 or whatever. Hopefully this new module will help with that. Any pointers would be appreciated. John

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Adding new MOH stream - isn't reloadxml enough to make it effective?
by Victor Chukalovskiy 06 Oct '11

06 Oct '11

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Re: [Freeswitch-users] RTCP packets not written
by Madovsky 06 Oct '11

06 Oct '11

if I understand the RFC, RTCP is only for RTP application statistics so it's not really necessary ? Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, October 04, 2011 6:26 PM Subject: RTCP packets not written Hi, I noticed 2011-10-04 14:24:28.512366 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:24:33.512388 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:24:38.512378 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:24:43.512390 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:24:48.512362 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:24:53.512375 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:24:58.512364 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:25:03.512369 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:25:08.512386 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:25:13.512390 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:25:18.512364 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:25:23.512376 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:25:28.512366 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:25:33.512365 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:25:38.512368 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:25:43.512372 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:25:48.512366 [DEBUG] switch_rtp.c:3910 RTCP packet not written 2011-10-04 14:25:53.512367 [DEBUG] switch_rtp.c:3910 RTCP packet not written is it come from a permission file somewhere ? Thanks

1 0

sofia auth challenge loglevel warning to debug10 & fail2ban
by Leon de Rooij 06 Oct '11

06 Oct '11

Hi all, I noticed that the loglevel for sofia auth challenge was changed from warning to debug10 about two weeks ago - which I really like btw, it shouldn't be a warning http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/endpoints/mod_s… But doesn't this change break fail2ban for the ppl who use that ? The regex should at least be changed on the wiki: http://wiki.freeswitch.org/wiki/Fail2ban#Edit_Config_Files But I think most people don't log debug level to files, so fail2ban won't be triggered even if the regex is changed, right ? Don't know if it's a problem, maybe I'm overlooking something, just wanted to let everyone know.. regards, Leon

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3pcc
by mazilo 06 Oct '11

06 Oct '11

On Line #66 of the /conf/sip_profiles/external.xml/ file, the 3pcc global option is commented out by default. I don't know if this 3pcc is a /fortunate/ or /unfortunate/ option to have, but it will make http://whistlephone.com WhistlePhone to work properly. Without the 3pcc option, there will be no incoming calls from http://whistlephone.com WhistlePhone (FS CLI will dump some messages as shown below) unless FS registers to all http://whistlephone.com WhistlePhone proxy servers as shown in http://wiki.freeswitch.org/wiki/Provider_Configuration:_WhistlePhone here . The question I have is if it is safe to enable the 3pcc option. If it is, then I don't mind to go ahead to update the wiki page. ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/3pcc-tp6865965p6865965.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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