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?
Hers is the debug out put:
<--- SIP read from 208.65.xxx.xxx:5060 --->
INVITE sip:firstname.lastname@example.org:5060 SIP/2.0
From: “CID NAME”<604777xxxx>;tag=kvspovbxperbwmfk.o
CSeq: 493 INVITE
o=Sippy 153068680 0 IN IP4 74.205.xxx.xxx
m=audio 34772 RTP/AVP 0
c=IN IP4 74.205.xxx.xxx