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From looking at
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I had it setup all internally, I was able to call
Can the call forward busy and call forward no answer settings be change
on the SipX??
I know that it can be done on the phones themselves but with server
based call forwarding, I much prefer to do it on the server. I done some
search but everything always points to the call forwarding but
Hello everyone,
We have an issue where we cannot install SipX 4.04 from the single CD
install on a Dell PE R410 servers. After trying multiple things we ended
up installing VMware ESXi and SipX on top of it and assign all resources
to SipX.
Now we are having an issue where the Java
Tony thank you for the quick response.
We did try doing a CentOS 5.4 install and installing SipX from Yum but
then we had some errors with file and data replication that we never had
with the single CD install. I had a small installation in the past
running 3.10 on top of ESXi and the only
On 2/24/2010 10:38 AM, Scott Lawrence wrote:
On Wed, 2010-02-24 at 09:46 -0500, Jhony Perez wrote:
We have an issue where we cannot install SipX 4.04 from the single CD
install on a Dell PE R410 servers. After trying multiple things we ended
up installing VMware ESXi and SipX on top
Hello everyone,
I have a group of Linksys phones (about 40) and after getting them to
register which took a while but I understand now the issue with the
upper case config file.
The issue is when we dial the voice mail extension we get asked to enter
the extension, once we enter the
Of Jhony Perez
Sent: Monday, January 18, 2010 4:16 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Linksys Phones (spa922,spa942 and spa962) can't
login to VM
Hello everyone,
I have a group of Linksys phones (about 40) and after getting them to
register which took a while but I
, Picher, Michael wrote:
You must use in-band dtmf...
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jhony Perez
Sent: Monday, January 18, 2010 4:16 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx
Hello everyone,
I'm installing SipX 4.02 single CD install on a Dell PE R410 server
with 4GB, 1066MHz, Dual Ranked UDIMMs memory, Intel Xeon E5502, 1.86 GHz
processor and 2 SAS 146GB 15K RPM HHDs.
When I boot from the disk, it goes to the SipX boot screen, I hit enter
and it start to run
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
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
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
The setting to specify how many digits the extensions are is under the
System -- Dial-Plan -- VoiceMail.
Then just change the VoiceMail and AA extensions.
Jhony
Dale Worley wrote:
sipXecs has a configuration setting to specify the number of digits in
an extension. (Although I can't find
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
call the SipXecs AA the DTMF tones work fine but when I type an
extension the AA attempts to transfer and it disconnects.
Thanks again...
Jhony
M. Ranganathan wrote:
On Tue, May 5, 2009 at 12:05 AM, Jhony Perez jpe...@zbzoom.net wrote:
Hello everyone,
I had a Cisco CME integrated
Mark,
This sounds like the same issue I'm having with the Cisco gateways, I
don't have a fix for it yet but you can see the SipXecs 4 with Cisco
gateway issues (Jhony Perez) in the list.
When you call in from the Asterisk try sending the call to the SipXecs
Auto Attendant, if this works
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 but it broke other areas.
What 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
with the
Service console for cpu time.
-M
Jhony Perez jpe...@zbzoom.net 01/05/09 6:26 PM
Hello everyone,
I have SipX 3.10.2 with CentOS 5 running on a VMWare ESXi server, I've
assign 1GB of RAM and selected 2 CPU during the install, the system
seems to run just fine but we started having intermittent
Hello everyone,
I have SipX 3.10.2 with CentOS 5 running on a VMWare ESXi server, I've
assign 1GB of RAM and selected 2 CPU during the install, the system
seems to run just fine but we started having intermittent issues with
voicemail quality, at first it was only on the playback of the
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