I am having a problem making calls through a SIP TRUNK.
I have the following setup
ALCATEL --- E1 (MGCP) ---> CALLMANAGER --- SIP TRUNK ----> ASTERISK
When I call from Asterisk to Alcatel or Callmanager, it works fine.
When I call from an IP Phone in Callmanager to a DN in Asterisk it works fine.
But When i make a call from Alcatel to Asterisk, the call fails.
I ran a sniffer on Asterisk to see the traffic flow between the Callmanager and Asterisk, and I found that the "from:" field of the message header on the INVITE packet is showing the extension number I am calling as the originating number.
When I am calling to extension 8050 in Asterisk (always from the Alcatel PBX) , I see the following message:
Via: SIP/2.0/UDP [Callmanager's IP Address]:5060;branch=z9hG4bK151f2a3e
From: "User" <sip:8050@[Callmanager's IP Address]>;tag=16778787
To: <sip:8050@[Asterisk's IP Address]>
Date: Fri, 07 Mar 2008 19:20:49 GMT
Call-ID: 96cf0400-1de10a7e-4a2-500880a@[Callmanager's IP Address]
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK
CSeq: 101 INVITE
Remote-Party-ID: "User" <sip:8050@[Callmanager's IP Address]>;party=calling;screen=no;privacy=off
Contact: <sip:8050@[Callmanager's IP Address]:5060>
The sniffing shows that when Asterisk receives the INVITE with that DN, it then asks for Proxy Authentication, and Callmanager does nothing.
When I make a direct call from Callmanager to Asterisk, the INVITE packet shows the correct information in the header, and the call proceeds without issues.
It's obvious to say that Extension 8050 does not exist in Callmanager nor in Asterisk.
Has anybody seen something like this?
I have Callmanager 4.1(3)SR3. I have been browsing the bug toolkit but couldn't find any clue or reference to a similar problem. I am NOT free to apply the latest service Release if I cannot guarantee that it's the solution to the problem, that's why I am asking for your assistance.
Can anybody help me out?
Best regards to all the NetPro community!