Problems Extension with a Call In on Asterisk 1.6

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Asterisk Users 3 Comments

Hi

I request your help because i don’t have actually a solution at my problems.

I have a Asterisk Server in 1.6
Connected at a SIP Provider
This provider supply me 2 numbers:
003318364xxxx (official number)
081169xxxx (Nddi Number)

When i receive a call on the 081169xxxx, he don’t use
the extension. He use the 003318364xxxx extension.

SIP Debug:

< --- SIP read from UDP://91.121.xxx.xxx:5060 —>
INVITE sip:003318364xxxx@78.41.xxx.xxx:5060;transport=udp SIP/2.0
Allow: UPDATE,REFER,INFO
Call-ID:
04459-NK-5fa6f8a0-18641fd41@sip.myoperator.net
Contact:
Content-Type: application/sdp
CSeq: 1602837515 INVITE
From: “033426aaaaaa”
;tag=04459-CI-5fa6f8a1-6f03b5b60
Max-Forwards: 30
P-Preferred-Identity:

To:
User-Agent: Cirpack/v4.42s (gw_sip)
Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2
Content-Length: 481

v=0
o=cp10 130085910854 130085910854 IN IP4 10.7.1.121
s=SIP Call
c=IN IP4 91.121.bbb.bbb
t=0 0
m=audio 36146 RTP/AVP 18 4 0 8 125 111 101
b=AS:21
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000/1
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:125 CLEARMODE/8000/1
a=rtpmap:111 iLBC/8000/1
a=fmtp:111 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=sqn:0
a=cdsc: 1 image udptl t38

< ------------->

3 thoughts on - Problems Extension with a Call In on Asterisk 1.6

  • Hi Oliver ,

    This is a simple scenario with asterisk you can edit sip.conf and in peer
    entry, try to add,
    context=(desired_context for peer)

    and then into context write a dial-plan for given number and route a call or
    whatever you want to do.

  • Hi

    Anyone know a solution at my problems ?

    Thanks
    Olivier

    2011/3/23 Olivier CALVANO :

  • Hi

    Very thanks for your helps, that’s work very goo

    Bye
    Olivier

    2011/3/25 DHAVAL INDRODIYA :