Perhaps this would help.
http://wiki.sipfoundry.org/display/sipXecs/Call+Redirectors
Each redirect is a hop.
On 11/09/2011 08:14 PM, Paul Kramer wrote:
Thank you both for your kind input. I'm very impressed at the
promptness of replies on the list so far!
I've created a new issue at
http://track.sipfoundry.org/browse/XX-9954, however this is the first
issue I have created so there could be problems with it.
Apart from the "Too many hops" error message in the sipxproxy log, was
there any other indication that the problem could be related to
max-forwards? In the pcap file I can see the original invite from the
ITSP contains a max-forwards value of 10 which i'm guessing also
helped in diagnosing the problem.
Do you know if there's a resource available describinig the flow of
data between each SipXecs component? I'm looking at the merged.xml
file in sipviewer and must say i'm quite confused.
Thanks again for your help.
Paul
On Wed, Nov 9, 2011 at 9:27 PM, Joegen Baclor <[email protected]
<mailto:[email protected]>> wrote:
Max-Forwards strikes again. Your ITSP is sending a Max-Forwards
of 10 which is consumed by the hops required to eventually land
the call to the IVR. The reason why it does not show when not
using aliases is because it takes a lesser amount of hop and does
not consume the max forwards. I have discussed this with the
developers and we agreed that we can introduce a
reset-max-forwards=value parameter in the ITSP account setting.
This would allow calls coming from these ITSPs with very low max
forwards to be rewritten.
Please create a tracker for this in jira and attach the logs
oisted here.
On 11/09/2011 04:19 PM, Paul Kramer wrote:
Ok here is the log with enhanced debugging. I had a quick look at
the voicemail dialplan and everything seemed to be in order.
On Tue, Nov 8, 2011 at 1:18 PM, Paul Kramer
<[email protected] <mailto:[email protected]>> wrote:
Hi all, this is my first post on the forums and I'm a little
wet behind the ears - might have to be a bit patient with me
Laughing
I'm having an issue with calls made from the PSTN coming in
through the Internode (Australian) ITSP trunk I have setup in
sipxecs. When I forward calls to an extension via System ->
Servers -> *server* -> Sip Trunking -> Incoming Calls
Destination, calls forward to VM successfully. However when I
apply my ITSP DID to any extension, the extension rings and a
call can be established but when the call is forwarded to VM,
it is dropped at the far end. VM forwarding also works as
expected during internal calls.
Attached are two sets of logs and traces: one with aliases
configured and one with calls passed straight through to an
extension.
Thanks in advance for your help.
Paul
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