Good day. I'm trying to configure termination with The Asterisk thru Cisco AS5300 Gateway from the SIP softphone (X-Ten X-Lite) to POTS network. I think, I had recognise kind of problem: call is ringing in the POTS phone (so I guess SIP signalling is working ok?), but there is no voice in either sides. On the Asterisk PC I can see incoming RTP streams with tcpdump and tethereal, but I can't to see any RTP outgoing streams. There is G.711 a-law codec selected on Asterisk and Cisco, and I see it in sniffed RTP's.
How can I fix this trouble? I have attached configuration files's pieces. Best regards, Alexandr.
[203] type=friend host=dynamic username=203 secret=nastya nat=no canreinvite=no context=office callerid="Nastya" <203> [EMAIL PROTECTED] qualify=1000 disallow=all allow=alaw allow=ulaw allow=gsm [ciscoout] type=peer host=176.16.1.16 canreinvite=no qualify=1000 disallow=all allow=alaw
[office] exten => 8,1,SetCallerID(2393030) exten => 8,2,Dial(SIP/[EMAIL PROTECTED],30,rtTwW)
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