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[asterisk-users] ShoreTel <-> Asterisk Integration


 
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PostPosted: Tue Jan 29, 2008 4:15 pm    Post subject: [asterisk-users] ShoreTel <-> Asterisk Integration Reply with quote
Does anyone have experience using ShoreTel SIP trunks to integrate an
Asterisk system?

I am having trouble when the ShoreTel system transfers an incoming call
from a SIP trunk to the voicemail system. From the SIP traffic, it looks
like it negotiates a codec correctly, but once the RTP stream starts the
call drops or there is no audio. I see errors in Asterisk such as:

chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission
xxxxxxxxxxxxxxxxxxxxxx at 192.168.x.x for seqno 104 (Critical Request)

Has anyone run into this before or have any ideas?

Thanks,
Joe
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