Hello all, Over the weekend, I setup and linked an Asterisk box at another site to the Asterisk box here.
The phones here are a mixture of Cisco 7940/7960 and Grandstream BT-100 phones. The phones at the other end are Grandstream BT-100 SIP phones. The Cisco phones run SIP 7.1 (upgraded last Friday from 6.1), the Grandstream phones run 1.0.4.68. Both Asterisk boxes are running stable CVS "CVS-06/07/04-16:18:54". The phones all use G711u to their respective Asterisk boxes. GSM is used between the Asterisk boxes. The boxes are 60ms apart. When on a call, the quality of the call is perfect. I've spent around 3 hours over 2 days on calls between the two systems. Today when I was on a call I noticed by accident that if you press a button (1 through 0 and */#) the call drops out after a few seconds. With the help of a colleague at the remote site we discovered the following: Grandstream <-> Asterisk <-> IAX <-> Asterisk <-> Grandstream. Any phone presses button, the call drops out. Cisco <-> Asterisk <-> IAX <-> Asterisk <-> Grandstream. Cisco presses button, call drops out. Grandstream presses button, call doesn't drop out. It doesn't seem to matter which side originated the call. What can be seen with SIP debug is 5 of these: Retransmitting #1 (no NAT): INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0549e133 From: "Shaun Ewing" <sip:[EMAIL PROTECTED]>;tag=as3dbbd61d To: <sip:[EMAIL PROTECTED]>;tag=ba97574145fedb8a Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 104 INFO User-Agent: Asterisk PBX Content-Type: application/dtmf-relay Content-Length: 24 Signal=7 Duration=250 I$P to 192.168.0.254:5060 We then get something like: Jun 7 14:34:06 WARNING[1125329600]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Request) set_destination: Parsing <sip:[EMAIL PROTECTED];user=phone> for address/port to send to set_destination: set destination to 192.168.0.254, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0549e133 From: "Shaun Ewing" <sip:[EMAIL PROTECTED]>;tag=as3dbbd61d To: <sip:[EMAIL PROTECTED]>;tag=ba97574145fedb8a Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 105 BYE User-Agent: Asterisk PBX Content-Length: 0 All the other debug bits indicate that the call terminated normally. I was wondering if anybody had experienced anything like this before? Regards, Shaun _______________________________________________ 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
