Jane Reeder
2005-Jun-30 00:01 UTC
[Asterisk-Users] Sipura 3k answers then immediate busy signal
I have a sipura 3000 that I am using just to send calls to my mac asterisk server. When you call the phone it rings, answers, and then goes right to a busy signal. Any ideas? Thanks for your help! Jane At the console in verbose mode I get: *CLI> DEBUG[8501248]: File chan_sip.c, Line 663 (create_addr): Setting NAT on RTP to 0 DEBUG[8501248]: File chan_sip.c, Line 554 (__sip_ack): Stopping retransmission on '63e5425660664f565ce2c88b2cdc4d51@ipaddressofasteriskserver' of Request 102: Found *CLI> DEBUG[8501248]: File chan_sip.c, Line 3898 (check_user): Setting NAT on RTP to 0 DEBUG[8501248]: File chan_sip.c, Line 554 (__sip_ack): Stopping retransmission on '9a4ae229-c06315ff@ipaddressofsipurabox' of Response 101: Found DEBUG[8501248]: File chan_sip.c, Line 3898 (check_user): Setting NAT on RTP to 0 DEBUG[8501248]: File chan_sip.c, Line 4950 (handle_request): Check for res for 400 DEBUG[8501248]: File chan_sip.c, Line 980 (find_user): Call from user '400' is 1 out of 0 DEBUG[8501248]: File chan_sip.c, Line 554 (__sip_ack): Stopping retransmission on '9a4ae229-c06315ff@ipaddressofsipurabox' of Response 102: Not Found DEBUG[8501248]: File chan_sip.c, Line 663 (create_addr): Setting NAT on RTP to 0 DEBUG[8501248]: File chan_sip.c, Line 554 (__sip_ack): Stopping retransmission on '417f48ea0b8c1c3e1b92305e1aa57976@ipaddressofasteriskserver' of Request 102: Found -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050630/d5844f27/attachment.htm