the autoattendant but not any extensions. From the Sipxecs
system I can still dial any of my Asterisk stations.
*From:* Jhony Perez [mailto:jpe...@zbzoom.net]
*Sent:* Monday, May 18, 2009 10:05 AM
*To:* Dale Worley
*Cc:* sipx-users@list.sipfoundry.org
*Subject:* Re: [sipx-users] SipXecs 4 with Cisco gateway
I wanted to let you guys know that the issue with the Cisco gateway is
resolved.
The issue started when I moved from SipXecs 3.10.3 to SipXecs 4.0. The
issue was that I could call from the SipXecs extensions out to the Cisco
gateway and to phones on the Cisco Call Manager Express but when
On Mon, 2009-05-18 at 10:18 -0400, Jhony Perez wrote:
I wanted to let you guys know that the issue with the Cisco gateway is
resolved.
The issue started when I moved from SipXecs 3.10.3 to SipXecs 4.0. The
issue was that I could call from the SipXecs extensions out to the Cisco
gateway
Scott Lawrence wrote:
On Mon, 2009-05-18 at 10:18 -0400, Jhony Perez wrote:
I wanted to let you guys know that the issue with the Cisco gateway is
resolved.
The issue started when I moved from SipXecs 3.10.3 to SipXecs 4.0. The
issue was that I could call from the SipXecs extensions out
On Mon, 2009-05-18 at 10:18 -0400, Jhony Perez wrote:
After trying many things I found what I think it was the last piece of
the puzzle, on the Cisco gateway dial-peer (call leg) that points to the
SipXecs, I specified what codec to use, G711 and now I'm able to call
just fine. After doing
Dale Worley wrote:
On Mon, 2009-05-18 at 10:18 -0400, Jhony Perez wrote:
After trying many things I found what I think it was the last piece of
the puzzle, on the Cisco gateway dial-peer (call leg) that points to the
SipXecs, I specified what codec to use, G711 and now I'm able to call
On Mon, 2009-05-18 at 13:04 -0400, Jhony Perez wrote:
I didn't specify that on the incoming dial-peer but only on the
outgoing to SipXecs, incoming it is up to the calling side to propose
what codec to use based on what's configure as the prefer codec.
No, when an element makes a call, it may
Thank you very much for your help, before I do the snapshot, I'm going
to look at the trace with Sip Viewer and a packet capture with
Wireshark, I'll let you know my findings right away.
Jhony
M. Ranganathan wrote:
This can be investigated only with a sipx-snapshot generated snapshot.
Its
Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of M. Ranganathan
Sent: Monday, May 04, 2009 9:37 PM
To: Jhony Perez
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SipXecs 4 with Cisco gateway issues
On Tue, May 5
Thank you for your quick reply, based on your reply I got part of it
working but it broke other areas.
What I meant for internal extensions is extension that are services
running on the server itself, IE. AutoAttendant, VoiceMail, Park, Etc.
Yes, I did leave the Inbound Calls Destination
On Mon, 2009-05-04 at 23:16 -0700, Todd Hodgen wrote:
I've have the same issue that I have been working to figure out. Calls from
a Polycom phone to two different xlite phones work fine. Calls from xlite
to xlite, or xlite to Polycom get a re-order tone. Calls from xlite to ITSP
get reorder
If the call comes in through a gateway and gets to the AA but disconnects on
transfer, it's likely the gateway is not handling the REFER for the transfer
properly.
Jhony Perez jpe...@zbzoom.net 05/05/09 10:47 AM
Thank you for your quick reply, based on your reply I got part of it
working
When I do a debug on the gateway, all calls look exactly the same, the
one that works to the AA and the one that doesn't to the extension.
Also, the gateway config has no change and it was working fine with
SipXecs 3.10.3.
Tony Graziano wrote:
If the call comes in through a gateway and gets
This can be investigated only with a sipx-snapshot generated snapshot.
Its tough to tell whats going on without one. Please generate one
with the failing scenario and mail me with it. Remember to set the
logging level to sipxbridge to DEBUG and that of the Proxy server and
Registrar to INFO. I
Hello everyone,
I had a Cisco CME integrated with a SipXecs 3.10.3 working perfect. I
had a dial peer (call leg) pointing to the SipXecs and a SipTrunk
gateway on the SipXecs pointing to the Cisco CME and then the Dial Plan
and life was good (for the most part).
I got a brand new server
On Tue, May 5, 2009 at 12:05 AM, Jhony Perez jpe...@zbzoom.net wrote:
Hello everyone,
I had a Cisco CME integrated with a SipXecs 3.10.3 working perfect. I
had a dial peer (call leg) pointing to the SipXecs and a SipTrunk
gateway on the SipXecs pointing to the Cisco CME and then the Dial
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