Re: [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits

2007-04-02 Thread Raj Jain
I found a subtle difference between the two traces you sent (the call that works and the call that gets dropped). This may or may not be what's causing the problem. The call that gets dropped had a retransmission of INVITE from UAC to UAS (and therefore retransmission of 200 OK from UAS to UAC).

Re: [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits

2007-04-01 Thread kjcsb
One potential reason could be that the ACK request being sent to Asterisk is malformed. Notice branch=0 in the top Via. This should start with z9hG4bK magic cookie since the INVITE was an RFC 3261 transaction. While branch=0 is valid in RFC 2543, I don't think an INVITE can start-off as RFC

[asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits

2007-03-29 Thread kjcsb
I have the following scenario: PSTN gateway (202.180.nnn.nnn) - OpenSER 1.0.1 (147.202.nnn.nnn) - Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, often (but not always) Asterisk repeatedly sends a SIP 200 OK message and eventually hangs up the call. sip.conf [general] port

Re: [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits

2007-03-29 Thread Raj Jain
One potential reason could be that the ACK request being sent to Asterisk is malformed. Notice branch=0 in the top Via. This should start with z9hG4bK magic cookie since the INVITE was an RFC 3261 transaction. While branch=0 is valid in RFC 2543, I don't think an INVITE can start-off as RFC 3261

[asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits

2007-03-29 Thread kjcsb
I have the following scenario: PSTN gateway (202.180.nnn.nnn) - OpenSER 1.0.1 (147.202.nnn.nnn) - Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, often (but not always) Asterisk repeatedly sends a SIP 200 OK message and eventually hangs up the call. sip.conf [general] port =