Hello,
I have a SIP trunk between Asterisk and an Avaya IP500. I can extension dial without a problem; however, I can’t dial externally from Asterisk via the IP500. I verified that the SIP trunk is registered.
Avaya is looking for my auth name in the SIP INVITE. Asterisk sends the dialed number in the invite.
For example: If my auth is 0739 and my dialed number is 18005551212, I need the header to look like this:
INVITE sip:0739@192.168.2.230 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.202:5060;branch=z9hG4bK03a48ee8;rport
From: “AvrohomXPSoftphone” sip:0726@192.168.2.202;tag=as11986a58
To: sip:18005551212@192.168.2.230
Contact: sip:0726@192.168.2.202
Instead, the header looks like this:
INVITE sip:18005551212@192.168.2.230 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.202:5060;branch=z9hG4bK03a48ee8;rport
From: “AvrohomXPSoftphone” sip:0726@192.168.2.202;tag=as11986a58
To: sip:18005551212@192.168.2.230
Contact: sip:0726@192.168.2.202
Avaya rejects this call since the number 18005551212 is not defined in its database. Wildcards are not accepted.
My registration info from sip.conf:
[IP500]
type=peer
;fromuser=0739
;fromdomain=192.168.2.202
;username=0739
;Asterisk sip setvar = 0739
;Asterisk sip secret = 0739
host=192.168.2.230
port=5060
;incominglimit=3
;insecure=invite
;nat=yes
;canreinvite=no
secret=0739
qualify=yes
trunk=yes
[authentication]
register => IP500:0739@192.168.2.230/0739
in sip_nat.conf:
register => 0739:0739@192.168.2.230/0739
insecure=very
When I go to the asterisk console, I can verify that my SIP trunk is authenticated:
AsteriskCentos5*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
IP500/0739 192.168.2.230 N 5060 OK (7 ms)
AsteriskCentos5*CLI> sip show registry
Host Username Refresh State Reg.Time
192.168.2.230:5060 0739 105 Registered Tue, 07 Jul 2009 00:18:12
I’ve tried all sorts of tricks, such as modifying the caller id, or DNIS, but nothing helps.
In extensions.conf:
exten = _91XXXXXXXXXX,2,Set(CALLERID(DNID)=0739)
I welcome any assistance you can offer.
Avrohom