I need to configure a SIP trunk between a UC560 and Asterisk.
I have already set up the Asterisk Trunk, incoming and outgoing routes.
At the moment all SIP calls proventienti dall’Asterisk and directed to the UC560 work regularly.
I have some problems on calls from dall’UC560 directed to Asterisk.
UC tells me that “The Number you have dialed is not in service”
I have enabled debugging on Asterisk SIP and I noticed the following:
pbxpi * CLI> <--- SIP read from UDP :/ / 192.168.0.227:57446 ---> ACK sip: firstname.lastname@example.org: 5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.227:5060; branch = z9hG4bK57231F From: "UC Extension" <sip:email@example.com>; tag = 1F7C8C-16B To: <sip:firstname.lastname@example.org>; tag = as707a2df5 Date: Fri, 21 Apr 2013 07:39:38 GMT Call-ID: 8F6EE81A-D97C11E2-8065DF0E-CD1EB93@192.168.35.227 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0
The problem seems to be that, typing 801 on a phone connected to the UC560 the “To” field contains "@192.168.0.230" (which is the IP address of the UC)
Unlike a SIP call made by a PC connected to UC560 with SotPhone attested to Asterisk I note the following:
<--- SIP read from UDP :/ / 192.168.10.12:5070 ---> ACK sip: SIP/2.0 email@example.com Via: SIP/2.0/UDP 192.168.10.12:5070; rport; branch = z9hG4bK2932784 To: <sip:firstname.lastname@example.org>; tag = as59a60f05 From: "SoftPhone" <sip:email@example.com>; tag = 7185 Call-ID: 1371751226-2784-COMPUTER-NAME@220.127.116.11 CSeq: 457 ACK Max-Forwards: 20 User-Agent: NCH Software Express Talk 4.28 Authorization: Digest Content-Length: 0
The Field “To” is properly compiled!!!
There’s a way to Edit the “To:” field of the incominc SIP Call?
Can you give me some guidance to solve the problem?
Thank you very much in advance