I am helping a friend on one of his sip trunk and couldn't find the way
to resolve his problem.
His asterisk's problem is like this:
0. When incoming call to one of his sip trunk, Asterisk reply with "488
Not acceptable here". So the call get dropped.
1. Recently upgraded Elastix with Asterisk 1.4.33
2. Was working fine before the upgrade
3. There are total 4 SIP trunks connected to different providers. All
others works fine.
4. All codecs are allowed.
5. I setup his account on my Asterisk as a SIP trunk, both incoming and
outgoing call work fine. (So it is not his provider's problem)
6. I checked his FreePBX style multi sip*.conf files and all seem correct.
So what can I do to find out where went wrong on this sip trunk?
Thanks.
Jian
Hers is the debug out put:
===========================
<--- SIP read from 208.65.xxx.xxx:5060 --->
INVITE sip:160428xxxxx at 192.168.1.83:5060 SIP/2.0
Via: SIP/2.0/UDP
208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-ad239907e0915d0b-1---d8754z-;rport
Via: SIP/2.0/UDP
208.65.xxx.xxx:5061;branch=z9hG4bK-pcerhxpz5hr4addh;rport=5061
Max-Forwards: 69
Record-Route: <sip:208.65.xxx.xxx;lr>
Contact: "Anonymous"<sip:208.65.xxx.xxx:5061>
To: <sip:160428xxxxx at 208.65.xxx.xxx:5060>
From: "CID NAME"<sip:604777xxxx at
208.65.xxx.xxx:5060>;tag=kvspovbxperbwmfk.o
Call-ID: 12904465 at 208.xx.xx.xx~o
CSeq: 493 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
cisco-GUID: 4084071434-3712422367-2859401243-560159692
h323-conf-id: 4084071434-3712422367-2859401243-560159692
Content-Length: 109
v=0
o=Sippy 153068680 0 IN IP4 74.205.xxx.xxx
s=-
t=0 0
m=audio 34772 RTP/AVP 0
c=IN IP4 74.205.xxx.xxx
<------------->
--- (17 headers 6 lines) ---
Sending to 208.65.xxx.xxx : 5060 (NAT)
Using INVITE request as basis request - 12904465 at 208.xx.xx.xx~o
Found peer 'vsp06'
Found RTP audio format 0
<--- Reliably Transmitting (NAT) to 208.65.xxx.xxx:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-ad239907e0915d0b-1---d8754z-;received=208.65.xxx.xxx;rport=5060
Via: SIP/2.0/UDP
208.65.xxx.xxx:5061;branch=z9hG4bK-pcerhxpz5hr4addh;rport=5061
From: "CID NAME"<sip:604777xxxx at
208.65.xxx.xxx:5060>;tag=kvspovbxperbwmfk.o
To: <sip:160428xxxxx at 208.65.xxx.xxx:5060>;tag=as40501684
Call-ID: 12904465 at 208.xx.xx.xx~o
CSeq: 493 INVITE
User-Agent: 000e082e83c7Linksys/SPA2102-5.2.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '12904465 at 208.xx.xx.xx~o' in
6400
ms (Method: INVITE)
<--- SIP read from 208.65.xxx.xxx:5060 --->
ACK sip:160428xxxxx at 192.168.1.83:5060 SIP/2.0
Via: SIP/2.0/UDP
208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-ad239907e0915d0b-1---d8754z-;rport
Max-Forwards: 70
To: <sip:160428xxxxx at 208.65.xxx.xxx:5060>;tag=as40501684
From: "CID NAME"<sip:604777xxxx at
208.65.xxx.xxx:5060>;tag=kvspovbxperbwmfk.o
Call-ID: 12904465 at 208.xx.xx.xx~o
CSeq: 493 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '12904465 at 208.xx.xx.xx~o' Method: ACK
Really destroying SIP dialog
'2ded46615e78d7992c15bea726fae454 at 127.0.0.1' Method: REGISTER
astpbx*CLI> sip set debug off
SIP Debugging Disabled
