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Jan Pichrt

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Hi, I am testing Kerio Operator for our company. Running on virtual on MS technology (converted vmware image)... I succesfuly connected Operator to SIP account of our VoIP provider (SIPY.cz) and created testing IVR. I have one line which is redirected directly to the IVR. Now, when I make the call, I hear nothing... just 20 second of silent and than hang up.

Inside my "Warning" log:
[07/Feb/2011 00:06:52] Feb  7 00:06:52 kerio asterisk[1673]: WARNING[3289]: channel.c:2811 in set_format: Unable to find a codec translation path from g722 to gsm

[07/Feb/2011 00:06:52] Feb  7 00:06:52 kerio asterisk[1673]: WARNING[3289]: file.c:912 in ast_streamfile: Unable to open /var/lib/asterisk/kerio_sounds/Kerio-MoH3 (format 0x1000 (g722)): No such file or directory

[07/Feb/2011 00:06:52] Feb  7 00:06:52 kerio asterisk[1673]: WARNING[3289]: pbx.c:5730 in pbx_builtin_background: ast_streamfile failed on SIP/999998743-08628158 for /var/lib/asterisk/kerio_sounds/Kerio-MoH3


And on my VoIP phone I recived error "701"... (hang up message ofc Rolling Eyes )

Is possible the connect to the machine and check the file permissions and file availability?

/edit: this error persist only when I am calling from SIPY.cz network... when I use a GSM service (T-Mobile) the IVR is playing, but not smoothly Sad you can try my test line at 00420 359 604 995 (CZ SIPY.cz)

What is wrong? Crying or Very Sad

And my second question: is possible automatically export list of calls to the file and send etc. to the FTP or another share (we are using this feature for automatically call accounting with SPA IS)?

File with structure like this:
2009-08-04 14:42:36       351XXX200     359XXX634         ANSWERED   23  


[Updated on: Mon, 07 February 2011 11:25]

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Vladimir Toncar (Kerio)

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Hi,

This appears to be an issue we have already encountered with the iLBC codec. It seems that Asterrisk does not by default transcode to/from iLBC and G.722 even though they are free. Until we are able to fix this, the work-around is to disable these two codecs.

The smooth playback depends on precise timing which is difficult to guarantee in virtual machines. We were able to solve that in VMWare using Asterisk's zt_dummy module so the virutal machine is able to handle at least a small number of concurrent calls. They same approach might not work in the MS virtualization.

Vladimir
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Jan Pichrt

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ok, thanks for your reply... i go move this virtual to another one (vmware). I try disable codecs too... than report back Smile

/edit: i am disabled this two codecs and still not hear IVR... Sad

[Updated on: Mon, 07 February 2011 13:03]

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Vladimir Toncar (Kerio)

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Hi,

Did you disable the codecs on the interface that handles your incoming calls?

Vladimir
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Jan Pichrt

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yes... see attachment, is that correct?

  • Attachment: operator.png
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  •  
Vladimir Toncar (Kerio)

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Hi,

Can you try to disable all other codec with the exception of G.711 A-law, G.711 U-law, and GSM? Maybe the incoming call uses G.729 or G.723.1 where transcoding is also not possible.

Vladimir
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