Does forwarding work when you forward to an internal phone instead of
another trunk?
On 01/08/2012 12:02 AM, S.K.- G wrote:
Thank you Tony and Michael..
We have eliminated the softphone from our testing and we are currently using
Navigata for incoming DID and ISP telecom for outgoing calls .. Still same
problem.. Please advise??
The attached Wireshark was captured when 6477247514 called the Sipx DID 647
689 7627 ( ext. 7627 / Polycom IP 550) and gets forwarded after 10 sec to
416 9515329 .. My cell rang but no audio ..
SIPX 74.112.46.78
SBC 74.112.46.12
Incoming DID Navigata
Outgoing trunk: ISP Telecom
Please advise, Your help is greatly appreciated
Regards
Saad
-----Original Message-----
From: [email protected]
[mailto:[email protected]] On Behalf Of Tony Graziano
Sent: Saturday, January 07, 2012 10:10 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Call Forwarding ..
This is a very non-traditional deployment. SIPX is hosted and there is
an acme sbc sitting in the same network as sipx. all users are remote
(behind a firewall) though sipx is setup to use the remote sbc via
sipxbridge. The issue they are having is with hairpinned calls via the
acme and the itsp. I've suggested to them that they look at using a
second itsp for the forwarded calls to test whether the itsp supports
it, as some simply do not work. Additionally, I've specified their
softphone is incorrectly configured and should be removed from the
test environment until it is properly configured/registered.
4.4(latest)
On Sat, Jan 7, 2012 at 9:42 AM, Michael Picher<[email protected]> wrote:
What version of 4.4 are you on (check bottom of GUI and post here)?
On Sat, Jan 7, 2012 at 9:11 AM, S.K.- G<[email protected]> wrote:
Dear All,
We are experiencing a weird problem with the Trunk to trunk call
forwarding feature.
The call forwarding is not working when an outside caller gets forwarded
to another external number.. the call ends up with a dead air, seems like
the RTP stream is not getting connected properly as the SIP signaling
negotiation process is being completed properly and the receiver phone
rings
but no audio path when answered.
We have tried the resolution xx-9521 but didn't make any differences as
the INVITE From ITSP Account was checked ..
NAT Traversal, and Server behind NAT are checked in the internet calling
form.. SIPX 4.4 on VMWARE server, Polycom IP 650 station, X-lite 4.0
The network diagram: SIPX> SBC> remote Polycom phones(In DMZ)
Any helps will be highly appreciated
Regards
Saad Khankan
Toronto- Canada
_______________________________________________
sipx-users mailing list
[email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
Michael Picher, Director of Technical Services
eZuce, Inc.
300 Brickstone Square
Suite 201
Andover, MA. 01810
O.978-296-1005 X2015
M.207-956-0262
@mpicher<http://twitter.com/mpicher>
www.ezuce.com
----------------------------------------------------------------------------
--------------------------------
Hope to see you at the sipX CoLab! http://www.sipfoundry.org/sipx-colab
A gathering for - open source users, eZuce customers& eZuce partners
Get the inside track on 4.6 and a glimpse at the future of sipXecs!
_______________________________________________
sipx-users mailing list
[email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-users/
_______________________________________________
sipx-users mailing list
[email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-users/
_______________________________________________
sipx-users mailing list
[email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-users/
