February 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Choice of codecs for incoming call
by Александр Мустафин 02 Feb '13

02 Feb '13

Hello. My FS receive INVITE with 2 codecs in SDP: PCMA and PCMU. After hunting dialplan, this call forwarding to IVR, and I see next message in log: 2013-01-31 01:23:22.388780 [DEBUG] sofia_glue.c:5094 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-01-31 01:23:22.388780 [DEBUG] sofia_glue.c:3077 Set Codec sofia/internal/9089080908@PSTN PCMU/8000 20 ms 160 samples 64000 bits 2013-01-31 01:23:22.388780 [DEBUG] switch_core_codec.c:111 sofia/internal/9089080908@PSTN Original read codec set to PCMU:0 And FS send 200 OK to remote side with one codec in SDP: PCMU. My parameters: In internal.xml (profile) <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> In vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,G729,G722,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729,GSM"/> <X-PRE-PROCESS cmd="set" data="inbound_codec_prefs=PCMU,PCMA,G729,GSM"/> This is normal behavior?How can I manage set of codec in inbound calls? Best regards, Alexander.

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FreeSWITCH Mirror on Github
by Ken Rice 02 Feb '13

02 Feb '13

Hey Guy, As many of you know there is a Github mirror of the GIT repo at https://github.com/FreeSWITCH/FreeSWITCH . This is intended as a READ ONLY mirror of our master repos. I have noticed that some people are actually doing push request and opening issues there. Please do not do that. The Core dev team rarely (if ever) reviews the other functions that Github provides. If you have an issue or a possible patch, please open a jira and attach the patch. If you have already patched your local copy, using the git diff¹ command to get the patch will work, just pipe its output to a .txt file and attach to your Jira. Thanks Ken -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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[CS_NEW] [WRONG_CALL_STATE] with incoming calls
by Frederick Pruneau 01 Feb '13

01 Feb '13

Hi guys! I have a problem with my freeswitch server and I need some help. All incoming calls are not working since yesterday. All I did is to change the IP address. The only error message I have is this one: [WARNING] switch_core_state_machine.c:514 7b24f8fa-6ae6-11e2-a6be-25229cd2cfbb sofia/internal/XXX@XXX.XXX.XXX.XXX:5060 Abandoned [NOTICE] switch_core_state_machine.c:517 Hangup sofia/internal/XXX@XXX.XXX.XXX.XXX:5060 [CS_NEW] [WRONG_CALL_STATE] What does Wrong call state mean and what can I do to fix this issue? Everything else is working. I can make outgoing calls and call other extensions. Thanks in advance. Fred

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ESL and Event Socket with secure connection
by Eduardo Nunes Pereira 01 Feb '13

01 Feb '13

How is the best way to implement connection between ESL and Event Socket with secure connection, something like HTTPS.

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Friday Free For All on Today?
by curriegrad2004 01 Feb '13

01 Feb '13

Is the Friday Free For All still going on today?

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Site Specific Errors (Polycom? Comcast Router? FS?)
by Sean Devoy 01 Feb '13

01 Feb '13

Hi, I am having intermittent problems with one of my new customers. These problems have not been reported by other customers using the same FS server, but that does not mean there may not be a FS solution. I appreciate any suggestions you folks have. I just did a GIT update yesterday from head. The site is unique in that it uses a Comcast Business Router/Modem/Gateway and is our only site with Polycom 330 phones. We were forced to set the local (phone) SIP port to a unique number across the board. That us each line on each phone has a unique port on the NAT'ed Router. As I mentioned this is our first attempt at using Polycom phones. The periodic problems are: Cross-Talk: Users report hearing other conversations in the office on their call. Googling produced a proposed suggestion of putting each phone's connection on a SWITCH to isolate rtp traffic. We removed their hub and put in a layer 2 switch. Several Users have reported that as they are dialing if another call comes in to their phone from FS they immediately get a busy signal. One user reported a call dropped when another call came in to his phone. I mentioned cross-talk only because it may mean something to you. I am hoping that issue is resolved. These are all intermittent, and our staff has not been able to reproduce at the customer site! Thanks again. Sean

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Re: [Freeswitch-users] Execute app in sync mode when using event sockets
by Peter Olsson 01 Feb '13

01 Feb '13

My best suggestion is to "think different". If you're using ESL in inbound mode, you probably don't want to block the socket for anything, since you might receive events for other channels etc. The best way is to use it asynchronous - as intended. Send the speak command, and then trigger on the event that is sent when the playback is finished. In my use case I use it for an IVR system, I have the ESL lib, and then I create one thread per new channel (could be done differently, but it was the easiest approach right now). Then I queue the events for the separate channels back into their threads (separated per channel uuid). In the thread I do the things I want asynchronously, then I wait for events to be queued to my thread, and I handle these events. When I get the reply that the playback is finished, I do the next step etc. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Hector Geraldino Skickat: den 31 januari 2013 20:19 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Execute app in sync mode when using event sockets Hi, I have a java app that uses mod_event_socket (inbound mode) to control FreeSWITCH. I can send api commands which, by default, are synchronous, so the thread locks until the command is successfully executed. The problem I'm facing is when I have to execute commands that are in other modules (like playback, or speak). As they are not api commands, I must send an execute appName + arguments, which by nature is executed in async mode. What I want to achieve is to execute an speak command and wait until the TTS operation ends (blocking the thread). This sounds like a trivial use case: play some audio (TTS, file) and then execute the rest of the logic. It is easy to accomplish on the dialplan, but I just can't figure out how to do it on an ESL application. Here's what I've tried so far: + send execute speak command + api sleep and wait for the CHANNEL_EXECUTE_COMPLETE (to simulate a lock [sleep] and continue [break after event is received]), but the event arrives after the sleep command completes. + use uuid_broadcast uuid speak::args, doesn't work Any hints, ideas are welcomed. Thank you! Hector !DSPAM:510ac0b632761097822086!

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Messaging & video calls using IMSDROID SIP client
by Sanath Prasanna 01 Feb '13

01 Feb '13

Hi, Recently I configured customized IMSDROID SIP client with Freeswitch Server. (sms-mod enabled) I can get calls between 2 IMSDROID clients through Freeswitch. I want to extend that to messaging & video calls. So if some one doing that. pls help to configure Freeswitch Server & IMSDROID client sides. Br, Sanath

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mod_httapi problem when trying to play an audio hosted in S3 amazon cloud
by Leonardo N. S. Pereira 01 Feb '13

01 Feb '13

Hi. I'm having problem when trying to play an audio hosted in S3 Amazon cloud The MP3 URL is: / / /h//ttp://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2Fblind_willie.mp3?Expires=1349381157&AWSAccessKeyId=AKIAJKQT6QPSI3BXJJ4A&Signature=bnJgjbnxqlgsdYIcKIbYUa9gqao%3D/ The result is: 2012-10-04 20:27:28.891298 [DEBUG] switch_ivr.c:605 sofia/external/+15752222222(a)67.231.8.195 Command Execute playback(http://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2… EXECUTE sofia/external/+15752222222(a)67.231.8.195 playback(http://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2… 2012-10-04 20:27:28.903180 [ERR] mod_httapi.c:2545 File at url [http://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2Fblind_wi…] is unreachable! If I use the Browser with same URL, I'm able to listen to the sample audio. Any idea about what is wrong? Thanks in advance, Leonardo

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simple install question
by Sean Devoy 01 Feb '13

01 Feb '13

Hi, I want to switch from the stable release version to the git head and rebuild in hopes of ridding myself of the delay in connection of the audio. On almost all my calls I must hesitate about 1 second before saying hello or the caller does not hear me. I have seen many discussions saying that this is probably fixed in the git root. So here are my questions: The procedure appears to be: Log in to root cd /usr/local/src/freeswitch git checkout master Which responds: Branch master set up to track remote branch master from origin. Switched to a new branch 'master' Then just: make current Is that correct? Do I need to stop the running FS while running this? If so, how long will I be down? (ball park it) I realize this is a RTFM kind of question, but I can't risk an outage. I need confirmation from someone who understands. Thanks, Sean

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