Hello All! There is description of my problem with Asteriks below. Asteriks CLI says: "File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded on call"
Sip debug on the server gives the next: Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.26:5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=106403508 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as0771c6f9 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 209 8523 is Cisco ATA-186 The sip.conf content: - - - - - [cisco8523] type=friend username=8523 secret=test nat=no host=dynamic canreinvite=no qualify=300 defaultip=192.168.0.26 - - - - - Why I place a call to Asteriks. I hear some invitation but connection brokes when retransmit exceed. Could anyone give some advice or solution. Thanks in advance -- Best regards Vlad _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
