[asterisk-dev] Curly Asterisk <-> Voxbone problem

[Mark Lynch]

Voxbone have send me some more details on what they are seeing when a call
is made.

The good machine is setting the following
c=IN IP4 203.206.181.174.

whereas the bad machine is setting:
c=IN IP4 81.201.82.39.

Which seems very wrong.  I've tried pretty much everything I can think of -
any clever ideas I could try before I rebuild the machine with i386 version?

-- Message from voxbone --

Now looking at the traces I took for both the good equipment
203.206.181.174vs the bad equipment
63.138.188.91, it seems your are telling your equipment to send RTP to our
SBC server. Here is the 200 OK messages you are sending to us:

Good equipment 203.206.181.174

U 203.206.181.174:5060 -> 81.201.82.39:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 81.201.82.39:5060
;branch=z9hG4bK0caef7a1b9d235f5282b5358e028689d;received=81.201.82.39.
From: "anonymous" <sip:0 at voxbone.com <sip%3A0 at voxbone.com>>;tag=37387.
To: <sip:61390015580 at 203.206.181.174 <sip%3A61390015580 at 203.206.181.174>
>;tag=as4e5b9cf8.
Call-ID: 73192f5a181daa0d4f28e96c4fb077e2 at 81.201.82.39.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Contact: <sip:61390015580 at 203.206.181.174<sip%3A61390015580 at 203.206.181.174>
>.
Content-Type: application/sdp.
Content-Length: 268.
.
v=0.
o=root 4719 4719 IN IP4 203.206.181.174.
s=session.
c=IN IP4 203.206.181.174.
t=0 0.
m=audio 10892 RTP/AVP 8 0 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


Bad Equipment:

U 63.138.188.91:5060 -> 81.201.82.39:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 81.201.82.39:5060
;branch=z9hG4bK955740fe5bd279f8512f98257acde109;received=81.201.82.39.
From: "anonymous" <sip:0 at voxbone.com <sip%3A0 at voxbone.com>>;tag=86815.
To: <sip:61280147490 at 63.138.188.91 <sip%3A61280147490 at 63.138.188.91>
>;tag=as587bc4c0.
Call-ID: 0889d6eabd1218a4bb8052349b347f71 at 81.201.82.39.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Contact: <sip:61280147490 at 63.138.188.91 <sip%3A61280147490 at 63.138.188.91>>.
Content-Type: application/sdp.
Content-Length: 263.
.
v=0.
o=root 6299 6299 IN IP4 63.138.188.91.
s=session.
c=IN IP4 81.201.82.39.
t=0 0.
m=audio 16674 RTP/AVP 8 0 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

Please look at the c=IN IP4 section, (connection information), for the good
equipment your are telling us to send audio to 203.206.181.174 while on the
bad equipment you are telling us to send audio to 81.201.82.39 which RTP is
not supported. Please make the appropriate changes on your side. Thank you.


-- 
OnScreen and Voice Learning and Assessment
www.learnosity.com
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