- FreeSWITCH-users - FreeSWITCH Mailing Lists

Choppy VM when using loopback (on Debian lenny)
by Øyvind Albrigtsen 03 Feb '11

03 Feb '11

Hi, I'm using latest git-version of Freeswitch, and when I go to voicemail when calling a number the sound playback is choppy and it skips some of the digits in the number I called. If I set sleep to 2000 it sounds better, but it's still a little choppy and still skips some of the digits. This is with the default settings coming with the git version. When I call 4000 to listen to voicemail it sounds fine. The kernel is the default kernel which is compiled with CONFIG_HZ_250=y Regards Oyvind

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Simple_conference.lua
by John Smith 03 Feb '11

03 Feb '11

Hi, I’m trying to have the script http://wiki.freeswitch.org/wiki/Simple_conference.lua running on my FreeSwitch server. I get an error when FS wants to play any phrase given in this example: *[ERR] switch_ivr_play_say.c:150 Can't find macro conference_welcome* * *It seems normal as I didn’t find any reference to such phrases in the different xml phrase configuration files. There seems also to be no adequate wav file in sounds/en/us/callie/voicemail/8000. My question is: Do these files exist somewhere or does the example assume that we have to record our own wav files? Thanks John

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Is it possible to use att_xfer on channels already bridged via loopback?
by Dmitry Sytchev 03 Feb '11

03 Feb '11

Is att_xfer or mod_loopback is broken in FS-current? I use FreeSWITCH Version 1.0.head (git-7eceff4 2011-01-16 22-33-50 +0000) Seems there were no updates of att_xfer or mod_loopback since that. I use loopback channel as destination when doing att_xfer to re-enter dialplan. With loopback_bowout=false and loopback_bowout_on_execute=false this works. But when any of connected parties tries to do att_xfer again, all channels get hangup on transferer hangup. Scenario: A calls B, B answers A launches att_xfer via *7, B listens to MOH A dials C and we do att_xfer to loopback/C C answers, A hangs up to complete transfer C and B are now bridged via loopback, `show channels` shows 4 channels include 2 loopback legs. Now, C also tries to do in-call transfer with *7. C launches att_xfer via *7, B listens to MOH C dials D and do att_xfer to loopback/D D answers, C hangs up to complete transfer B and D are hung up instead of be bridged together. There are also issues with MOH wile running att_xfer, but they are not so important as att_xfer behavior itself. -- Best regards, Dmitry Sytchev, IT Engineer

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Caller ID using Fifo
by Joao Leme 03 Feb '11

03 Feb '11

Hi there, I would like to know if there is a way to see the caller ID on my Sip Client (X-Lite for example) of the caller that I answear from a Fifo queue? Thanks, John

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Fs runninf on Clound Service (XEN)
by dome＠tel.co.th 03 Feb '11

03 Feb '11

Dear All, I'm testing FS on http://cserver.cloudwww.com/ then give me IP : 61.x.x.x and in eth0 eth0 Link encap:Ethernet HWaddr 00:16:3E:2C:91:DC inet addr:10.6.32.11 Bcast:10.6.33.255 Mask:255.255.254.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:160267 errors:0 dropped:0 overruns:0 frame:0 TX packets:93251 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:189671769 (180.8 MiB) TX bytes:9669248 (9.2 MiB) So i can access 61.x.x.x without problem. but i want to know how to config sofia profile for binding 61.x.x.x in default config. i can send call to 61.x.x.x but when call out i got 10.6.32.11 and i can't find 61.x.x.x in sip header someone help me please BG Dome C.

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last git problem ?
by Madovsky 03 Feb '11

03 Feb '11

after every hangup now FS quits unexpeclty,without any error message....

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Voice quality monitoring via loopback on sip endpoint
by Marc De Corny 02 Feb '11

02 Feb '11

Hi all, I am thinking of a way of testing the quality of a voice call. Ideally i would like to create a call from FS with a loopback parameter that tells phone to answer the call automatically and loopback the RTP. This is a common functionality in most phoned (for example cisco) The sent and received packets could be compared for jitter, latency and packet loss and a result extracted. Does this already exist? I know there some specific tool that accomplish this bit they are expensive and awkward to use. Starting and retrieving the result from an api would be fantastic. Any ideas are welcome. Thanks Marc

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Re: [Freeswitch-users] last git problem ?
by Madovsky 02 Feb '11

02 Feb '11

maybe thies unusual log lines after hangup before FS quits 2011-02-02 14:58:35.549213 [DEBUG] switch_ivr_bridge.c:500 sofia/internal/9999999999999@default ending bridge by request from read function 2011-02-02 14:58:35.549213 [DEBUG] switch_ivr_bridge.c:494 sofia/internal/9999999999999@default ending bridge by request from write function nothing else ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, February 02, 2011 2:54 PM Subject: last git problem ? after every hangup now FS quits unexpeclty,without any error message....

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execute_on_answer
by Madovsky 02 Feb '11

02 Feb '11

possible to use execute_on_answer=<api> value; <api> value ??? thanks

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execute_on_ring executing on answer
by François Delawarde 02 Feb '11

02 Feb '11

Hi, Doing some testing with this morning's git (Fri Jan 28) I just found out that the execute_on_ring application runs when the destination answers instead of when it rings. So far, I can't seem to find out the reason. Could it be some configuration issue? Here a call log showing the phenomenon with a simple bridge: <action application="bridge" data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/192.168.10.1/sip:2103@192.168.10.22:5060"/> http://pastebin.freeswitch.org/15168 Thanks, François.

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