> What does this mean? I have a sipura 3000 with an analog line that I > have created as a trunk. Incoming calls make it to the sipura but not > to the pbx. However I can make outgoing calls but have no audio. I > thought it might be a codec issue so I set disallow=<blank> and > commented out the "allow=". I get the following in my logfile: > build_route: Contact hop: satelliteout > Jun 2 21:06:33 VERBOSE[2393]: -- SIP/satelliteout-af86 is ringing > Jun 2 21:06:33 VERBOSE[2393]: -- SIP/satelliteout-af86 answered SIP/100-6eab > Jun 2 21:06:33 VERBOSE[2393]: -- Attempting native bridge of > SIP/100-6eab and SIP/satelliteout-af86 > Jun 2 21:06:39 WARNING[2393]: Maximum retries exceeded on call > [EMAIL PROTECTED] for seqno 102 (Non-critical Response) > > Satelliteout is my outbound trunk and the call is being made from extension > 100. > Any idea what this means, I don't see anything that indicates an error > when running asterisk -rvvvvv in the console, this is taken from the > asterisk log files. >
Check to ensure canreinvite=no is defined in sip.conf entries for both the sipura and your phone. Might also do disallow=all and allow=ulaw in sip.conf and restart asterisk. See if either impacts the problem. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
