June 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

in bypass_media mode, 2833 payload type modified by freeswitch!
by Qingquan Luo 14 Jun '11

14 Jun '11

Hi All, I use freeswitch to bridge one incoming call to other target phone number. I use bypass_media mode. So the rtp is not go through the freeswitch, By I notice that When LegB reply 183 or 200 SDP message. Freeswitch change it 2833 telephone-event payload type. and forwarding the message to caller. This make the caller use wrong dtmf payload type to send dtmf What wrong with that? How can fix it? Any information or help is welcome. Thanks Best Regards Qingquan -- Using Gmail? Please read this important notice: http://www.fsf.org/campaigns/jstrap/gmail?40922.

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FreeSWITCH Cookbook Question: Pocketsphinx
by Michael Collins 14 Jun '11

14 Jun '11

Hello all! I am looking for examples of mod_pocketsphinx being used in a real-world application. If you are using pocketsphinx ASR for something useful and can share some configs I would appreciate it. I'm soliciting ideas on how to make the pocketsphinx recipe in the cookbook be as truly useful as possible. I don't personally use it for anything and I've only seen the pizza demo in action, so I'm asking our intrepid community members for some thoughts. Email me off list, please. Thanks! -Michael

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Re: [Freeswitch-users] continue_on_fail failover with mod_lcr
by Ivan Kovacevic 14 Jun '11

14 Jun '11

Anyone? I am still wondering if there is a documented/undocumented reason why continue_on_fail does not work with piped failover? From: Ivan Kovacevic [mailto:ivank@rogers.com] Sent: June-09-11 4:38 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: continue_on_fail failover with mod_lcr Hi Everyone, I would like to implement fail-over for my outbound gateway, but I would like to be able to pick certain cause codes which qualify to stop trying next gateway (specifically when I have bad number and I am getting sip:404 NO_ROUTE_DESTINATION, UNALLOCATED_NUMBER). I was using 1.0.6, but I moved to the newest git about month ago. After searching through lists and spending several hours playing with continue_to_fail, failure_causes and fail_on_single_reject I was able to make it work by specifying cause codes for which I want to fail-over and omitting ones that qualify to stop trying next gateway in variable continue_to_fail: So this setup is working for me and in the case I have bad number and x.x.x.x returns sip:404 (NO_ROUTE_DESTINATION or UNALLOCATED_NUMBER) it is not trying y.y.y.y gateway. <include> <extension name="myextension"> <condition field="destination_number" expression="^(01144\d+)$"> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,SERVICE_NOT_IMPLEMENTED,CALL_RE JECTED,NO_USER_RESPONSE,EXCHANGE_ROUTING_ERROR,INCOMPATIBLE_DESTINATION,NETWORK_OUT_OF_ORDER,RECOVERY_ON_TIMER_EXPIRE"/> <action application="bridge" data="sofia/myprofile/$1(a)x.x.x.x"/> <action application="bridge" data="sofia/myprofile/$1(a)y.y.y.y"/> <action application="bridge" data="sofia/myprofile/$1(a)z.z.z.z"/> </condition> </extension> </include> However if I want to use "|" between my gateways - and the example below is not working. And no matter what x.x.x.x returns - it will try y.y.y.y and eventually z.z.z.z. <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,SERVICE_NOT_IMPLEMENTED,CALL_RE JECTED,NO_USER_RESPONSE,EXCHANGE_ROUTING_ERROR,INCOMPATIBLE_DESTINATION,NETWORK_OUT_OF_ORDER,RECOVERY_ON_TIMER_EXPIRE"/> <action application="bridge" data="sofia/myprofile/$1@x.x.x.x|sofia/myprofile/$1@y.y.y.y|sofia/myprofile/$1@z.z.z.z"/> Unfortunately, we have to use pipe for fail-over since we are using mod_lcr to choose between outbound gateways. Any suggestions? Thanks, Ivan

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CHANNEL_BRIDGE event missing
by george russell 13 Jun '11

13 Jun '11

Freeswitch has connected my call. The call has gone through the gateway and connected to the phone on the other end. Because the call has connected I thought CHANNEL_BRIDGE should of occured. This is the list of events captured. Why would CHANNEL_BRIDGE event be missing even though the call has been bridged? I have done a "events plain all" and filtered for aleg uuid and bleg uuid "filter Unique-ID (aleg uuid)" "filter Unique-ID (bleg uuid)" CHANNEL_EXECUTE / export CHANNEL_EXECUTE_COMPLETE / export CHANNEL_EXECUTE / export CHANNEL_EXECUTE_COMPLETE / export CHANNEL_EXECUTE / export CHANNEL_EXECUTE_COMPLETE / export CHANNEL_EXECUTE / export CHANNEL_EXECUTE_COMPLETE / export CHANNEL_EXECUTE / export CHANNEL_EXECUTE_COMPLETE / export CHANNEL_EXECUTE / export CHANNEL_EXECUTE_COMPLETE / export CHANNEL_EXECUTE / export CHANNEL_EXECUTE_COMPLETE / export CHANNEL_EXECUTE / bridge CHANNEL_PROGRESS / CALL_UPDATE / CALL_UPDATE / CALL_UPDATE / CALL_UPDATE / CHANNEL_ANSWER / SESSION_HEARTBEAT / CHANNEL_HANGUP /

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regex groups
by Joseph L. Casale 13 Jun '11

13 Jun '11

I wrote up a quick dialplan and intended on using $0 for the whole number, so my regex was: <condition field="destination_number" expression="^0\d+$"> but $0 was empty until I rewrote it: <condition field="destination_number" expression="^(0\d+)$"> Is this expected, I assumed not and I was not looking for $1 etc I figured $0 to always get populated. Thanks, jlc

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originating a call from an outbound socket app
by Hector Geraldino 13 Jun '11

13 Jun '11

Hi all, I've been playing with FS around weeks, with some level of success. Right now I'm stuck trying to dial an extension, or to bridge a call to an external extension. Briefly, the scenario is the following: - Cisco CCM forwarding SIP calls to FS - FS in outbound mode, forwarding calls to a Java application using freeswitch-esl jar The java application answers the call, says a dialogue using TTS and receives input using DTFM or speech commands; all this is working pretty well. Now I'm trying to put this call in hold, originate a call to another extension and, depending on the result (call answered, not answered, destination user presses an specific number generating an expected DTMF), I need to bridge both calls and finish the session. I try several ways to get this done with no luck yet: using api originate to make the call with different arguments ({ignore_early_media, etc). I even tried to skip this step and just do the bridge between the inbound call and the external extension, but this didn't work either. I can't see any errors thrown by FS, and the result of the command is +OK, but nothing happens, the external phone never rings. The FS log only says: [DEBUG] switch_core_session.c:954 Send signal sofia/external/5655(a)192.168.8.1 [BREAK] <= this ext/IP address is from the inbound caller, not the one I'm trying to connect to. My first impression was that I might have something wrong with the originate/bridge parameters. I changed the dialplan XML file to just do the bridge, and it worked like a charm. <include> <extension name="test"> <condition field="destination_number" expression="6114|5121">  <action application="bridge" data="sofia/external/5562(a)192.168.2.14" /> </condition> </extension> </include> I don't know what path to follow here. I was unable to find relevant examples on google that guides me to solve this little issue. Any advices? Thanks all, Hector

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conference high cpu and audio delay
by Cameron Banta 13 Jun '11

13 Jun '11

Hello, I'm running FreeSWITCH Version 1.0.7 (hacked-20110606T040042Z) on Ubuntu 10.04 64 bit with kernel 2.6.35-22-server. I am running about 300 simultaneous calls spread out over about 50 conferences. I'm getting a cpu load of 20 to 30, and cpu utilization about 75% on each core. There's about a 2 second audio delay in the conference when it gets this loaded. Is this high load and audio delay just the limit of my cpu (dual core E5200)? I ask because I was able to get the same number of calls on this server with asterisk 1.6 with app_konference with a much lower cpu load - about 4 or 5 - and no audio delay. Thanks, Cameron

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X-Lite giving error while connecting to FreeSwitch Server
by ankIT WALiA 13 Jun '11

13 Jun '11

Hello all, I am very new to FreeSwitch and VOIP Technology. I installed the FreeSwitch and X-Lite on my Window 7 OS. I am using default FreeSwitch configuration. Nothing changed except adding Text to Speech module. When I am trying connecting to FreeSwitch Sever on localhost from X-Lite Softphone. I am getting the error on X-lite Account failed to enable. Account 1001 could not be enabled. Problem at server(SIP error 503). Try Again Later I have given the account setting at X-Lite as below: Account Name: 1001 Protocol: SIP UserID: 1001 Domain: localhost Password: 1234 Display Name: Test User Authorization Name: 1001 Domain Proxy: Register with Domain and receive calls: Checked Send Outbound via: DOmain I dont know what wrong I am doing and How I should debug this issue. Please help. Thanks Ankit

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Timer Config Off
by Rev Michael Carbone 13 Jun '11

13 Jun '11

Not sure what happened but recording a conference was working, now I get this error when I try to record a conference: 2011-06-13 15:33:16.337400 [ERR] mod_conference.c:2897 Error Opening File [recordings/conference_CallInLineRadio-localhost-2011-06-13-06_3.wav] 2011-06-13 15:33:16.337400 [ERR] switch_core_timer.c:117 Timer is not properly configured. any ideas?

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Listen to the conversation between a user and an IVR
by Yungwei Chen 13 Jun '11

13 Jun '11

Hi, I'm wondering if there's a way to listen to the conversation between a user and an IVR (javascript programs). Here's the background info. I have a javascript that first makes an outbound call to someone. And When the call is answered, the call is then connected another extension, which runs an IVR application written in javascript. I want to be able to listen to the live conversation between the person and the IVR. Is this supported? If so, how can I do that in javascript? Thanks.

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