Wrong callee id with aterisk 11.5

Asterisk Asterisk Support
1

Hi All,
i have installed asterisk 11.5.1 on a Centos 6.4 server using a TE120P digium card to interconnect 2 sites. at the second site, a traditional pbx is connected to the TE120P card.
when i send calls from server 1 to asterisk server, the callee number isn’t sent. asterisk try to call the server ip address instead of the extension we need.
i did capture with wireshark wich show that extension is well transmitted:as shown below:
[color=#400000]To: sip:1001@41.xxx.xxx.[/color]
this is the message i get in asterisk cli:
[color=#0000FF]“Call from ‘SWITCH_PARIS’ (88.163.xxx.xx:5060) to extension ‘192.168.1.14’ rejected because extension not found in context ‘incoming’.”[/color]

here is the incoming context:

[color=#BF0000][incoming]
exten => _1XXX,1,Dial(DAHDI/g1/${EXTEN})
exten => _1XXX,2,Playback(invalid)
exten => _1XXX,3,Hangup[/color]

NB: calls from asterisk to other site are ok.
Please help me solving this issue. :cry:

Best Regards;

2

SIP routing is based on the INVITE line, not the To: line.

3

these are captures results i did:

  • from the caller side : INVITE sip:1001@41.202.x.x SIP 2.0
  • from asterisk (receiver side) : sip:192.168.1.14@41.202.x.x SIP 2.0

visually something is changing the called extension at the asterisk side to his internal ip address !!!

Regards;

4

Looks like you have a broken router; at a guess, it assumes that SIP URIs are always in the, degenerate, sip: form. It is generally advised that you turn off SIP-ALG in routers.

5

Hi david55,

effectively, the problem was due to the router. i changed with a Dlink dsl 2570u and all work well now. thans very much for your support.

Regards;