Displaying 1 result from an estimated 1 matches for "0823eab0".
2008 Feb 01
2
Asterisk 1.6 - Problems with SIP/REFER
...at 1. What is this message telling me?
What do I need to do to allow these phones to transfer calls between each other? Any help is greatly appreciated!
Here is the debug:
*CLI> == Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Executing [5878 at phones:1] Dial("SIP/5253-0823eab0", "SIP/5878") in new stack
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
Audio is at 10.7.10.1 port 19968
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.7.10.51:5060:
INVITE sip:5878 at 10.7.10.51:5060 SIP/2....