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StefanFN

6 posts

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#74208 27-Dec-2010 15:40
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Hi,

this problem has been discussed in a few threads here and many other forums. But the suggested solutions (e.g. progressinband = yes) do not work for me.
My setup/tests:
 - Atcom IP02 device with latest switchfin compilation applied
 - 3 local extensions 6000 - 6002 .
 - 1 VFX trunk
 - inbound calls get all routed to ext 6001
 
 - Internal calling is possible
 - external calling is possible to national and international landlines and cellphones
 - internal transfer and call forwards are possible
 - inbound calling from cell phones and landlines is possible

 - Follow-me is possible to cellphones
 - diverting the call via extensions.conf (exten = s,1,Dial(${VFX_trunk}/<target#>) ) works
 - Call forwards initiated via phone (302 "Moved Temporarily") fail with no audio. Signaling/ringing works and the first button press accepting the call works too, but no conversation possible as both ends are just silent.

The audio fails only, when the call is initiated from a TNZ cell phone (tried 2degrees which is working). The cfwd destination is alos a TNZ cell phone.
In order to avoid the firewall, I made the PBX the DMZ host, i.e. no firewall to the internet just NAT (and 1-to-1 PAT).

Did anyone come across this problem? And if so did it get resolved? Otherwise, can anyone guide me the right path to troubleshoot this, please?

The CFWD functionality is required since we turn this on when we leave the office. This is a pre-programmable button at the extension phone, i.e. no website login for (de-)activation.

Many thanks,
--
Stefan

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StefanFN

6 posts

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  #421873 27-Dec-2010 15:55
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the deskphone where the cfwd is set up is a YeaLink T22.
I have the gut feeling the solution lies somewhere in the phone and is either proxy or codec related.
NAT Traversal on the phone is set to "Disable" as I am unsure if Switchfin provides a STUN server or which STUN server details to use. The OutBound Proxy is my Switchfin/Asterisk box.

Thanks,
--
Stefan



coffeebaron
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  #421889 27-Dec-2010 17:19
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What codecs are you using - is G729 enabled? If you haven't, set all codecs to ulaw/alaw only & try that.




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sbiddle
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  #421904 27-Dec-2010 17:59
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How are you configuring the SIP 302 forwarding in Asterisk?




StefanFN

6 posts

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  #421986 27-Dec-2010 22:37
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@coffeebaron: G729 is part of the codec list but is not exclusive. Will try limiting phones, trunks and extensions to ulaw and alaw only. Will try after that.

@sbiddle:
I did not have to configure 302 in Asterisk at all. I assume (haven't debugged the SIP session) that the 302 reply includes the cfwd destination which is interpreted and dialed by Asterisk.

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