I've looked through the phones and SIPx and there is no call forwarding
setup on the users or the phones.
in sub2.xml i can see and ACK Coming FROM 205 (message 28) but NOT to 205
On 03/02/2011 10:50, Tony Graziano wrote:
In looking at the trace, the ACK is going back to the original call
destination (205, at IP 10.7.6.214). Does this happen only when the
call is forwarded?
I'm confused by this (message 6 in sub2.xml):
SIP/2.0 100 Trying
From: <sip:[email protected]
<mailto:sip%[email protected]>>;tag=b6231ca6
To: <sip:[email protected] <mailto:sip%[email protected]>>
Call-Id: MDI4MTQ0MTE1MzJkNjE0Zjk1ZjlkY2Q4NWJkZjkzMGI.
Cseq: 2 INVITE
Via: SIP/2.0/UDP
10.7.6.214:2695;branch=z9hG4bK-d8754z-1f81618a1d15b762-1---d8754z-;rport=61599;received=78.32.98.42
Content-Length: 0
WHERE are you performing the forwards from? sipX or the phone?
On Thu, Feb 3, 2011 at 5:20 AM, xavier houghton
<[email protected] <mailto:[email protected]>> wrote:
HI List, i have a problem with a UA when recieving inbound calls.
The problem seems to be a missed routed ACK coming from SIPproxy.
I'm posting it here as it seems to be similar to a couple of bugs in the tracker
(XTRN-970 <http://track.sipfoundry.org/browse/XTRN-970> . I had earlier in
the week posted a similar issues in sipxuser list
I'm calling an internal extention from another internal extention.
the scenarion i have is as follows:
phone1 - X-lite on LAN10.7.6.214 <http://10.7.6.214>: ext 205
Phone 2 - Intertex ix67 ATA on LAN (using phone port)10.7.6.191
<http://10.7.6.191>: ext 206
My LAN is Behind a NAT. I have an intertex firewall with SIP module off
and no rewriting going on to replicate a standard remote setup with ALG off
SIPx hosted on VM on public IP address outside of this network. its running
4.2.1-01897 build 34 from ISO. the server is behind a 1:1 NAT
internet calling is setup for remote users and sipx knows its behind a nat.
the registrations look like this is sipxconfig
sip:[email protected]
<sip:[email protected]:63854;rinstance=231fc0b019b13367;x-sipX-privcontact=10.7.6.214%3A5546>
179
sip:[email protected]
<sip:[email protected]:63996;x-sipX-privcontact=10.7.6.191>
The symptoms are that after 60 seconds an inbound call to the UA stops.
here's what happens to the call
When i call phone 2 from phone 1 the call is setup and there is audio both
ways. after 62 seconds the call stops.
I've looked at the traces in sipviewr and see that on the call setup SIPX
get a 200 OK from phone 2 which is forwarded to Phone 1.
phone 1 replies with an ACk to sipproxy.
SIPx forwards the ACK to the Phone 2 LAN address, destination 10.7.6.191.
as the ACK never gets to the phone2, phone 2 resends the 200Ok a number of
times with the same scenario as above happening.
after a number of retires phone 2 sends a Bye and the call ends.
i have no issues the other way round. that is to say when calling Phone 1
from phone 2 i have no issues had the call up for around an hour.
I've included a snapshot attached.
I've also included a trace from SIPX.
Sub2.xml is the failled call.
Sub1.xml the working call for reference if needed.
help?
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