Hi,
I'm playing around with the auto attendant in the dial plan setup. When
creating a new attendant with configured holiday attendant, I cannot call the
attendant any longer. After a second the call gets dropped, no voice at all.
When I remove the holiday attendant, everything is working fine.
On Mon 02.Nov.09 10:19, Weigel, Stefan wrote:
>Hi Dan,
>
>just check your files under /etc/yum.repos.d/. After I had to handle this bug
>I had to reinstall the the original release* package, which
>sipxecs-release-4.0.2.rpm was replacing.
>It overwrites the /etc/issue, /etc/re
o.com]
Gesendet: Dienstag, 3. November 2009 20:26
An: Weigel, Stefan; sipx-users@list.sipfoundry.org
Betreff: RE: [sipx-users] Problems after upgrade from 3 to 4.02
Thanks for the response. I have decided to give up on the upgrade and have
configured a new sipx server in my virtual environme
onald | Systems Administrator | Brammo, Inc.
Email: jmcdon...@brammo.com | www.brammo.com
550 Clover Lane | Ashland, OR 97520
Phone: 541.482.9555 ext. 339 | Mobile: 541.531.0005 | Fax: 541.552.0414
MCSA 2003
-Original Message-----
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users
Hi Dan,
just check your files under /etc/yum.repos.d/. After I had to handle this bug I
had to reinstall the the original release* package, which
sipxecs-release-4.0.2.rpm was replacing.
It overwrites the /etc/issue, /etc/redhat-release so yum functionality wasn't
given.
On CentOS 5.4 reinstal
Hi,
we upgraded our SipXecs system this weekend and we found a bug when working
with a Patton SN4638 gateway:
in patton we're setting the display-name to show the originating number when
running into ACD.
With a setting in patton:
Looks Up For called-e164 Of || Modifies calling-name To
123
upgrade to 4.0.2 (or
4.0.3) then do a backup/restore so the version is the same. I think I would
also use the built-in backup/restore and not pgdump first. Realize the above
statement is, "if it were me".
On Mon, Oct 26, 2009 at 8:56 AM, Weigel, Stefan
mailto:stefan.wei...@allianz-
anz-warranty.com
Geschäftsführung: Andreas Rösing, Horst Ziegler
Amtsgericht München, HRB 175682
Für Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720
Von: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Gesendet: Montag, 26. Oktober 2009 14:05
An: Weigel, Stefan
Cc: sipx-users@list.sipfoundry.org
Betreff:
Hi,
I'm trying to move a SipXecs 3.10.2 installation on a Fedora-8 i386 system to a
freshly new installed CentOS-5.3 x86_64 with SipXecs 4.0.2.
I moved the files and restored the postgres databases from the pg_dump. But I
got some issues with SSL stuff (error in replicating).
So my general ques
liche Nachricht-
Von: Nikolay Kondratyev [mailto:k...@nstel.ru]
Gesendet: Mittwoch, 7. Oktober 2009 16:51
An: 'Scott Lawrence'; Weigel, Stefan
Cc: sipx-users@list.sipfoundry.org
Betreff: RE: AW: AW: [sipx-users] sipXbrige questions
Looks like I finally understand what is happening.
S
":442:SIP:DEBUG:azwslx11.azwarranty.int:SipClientUdp-19:FFFF:SipXProxy:"SipClient[SipClientUdp-19]::run
resPoll= 1 revents: fd[0]= 1 fd[1]= 0"
"2009-10-07T12:41:48.675254Z":443:SIP:DEBUG:azwslx11.azwarranty.int:SipClientUdp-19::SipXProxy:"SipClient[
09 13:40
An: Nikolay Kondratyev
Cc: Weigel, Stefan; sipx-users@list.sipfoundry.org
Betreff: Re: [sipx-users] sipXbrige questions
On Wed, 2009-10-07 at 14:49 +0400, Nikolay Kondratyev wrote:
> Stefan,
> Sometime ago I dug into rfc 3261 regarding "To" and "Request line" he
---
Von: Todd Hodgen [mailto:thod...@verizon.net]
Gesendet: Mittwoch, 7. Oktober 2009 09:33
An: Weigel, Stefan; sipx-users@list.sipfoundry.org
Betreff: RE: [sipx-users] sipXbrige questions
Just a thought here. The ITSP routing I believe is done by the domain name.
When you build the bridge, you ha
ht München, HRB 175682
Für Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720
---
-Ursprüngliche Nachricht-
Von: Josh Patten [mailto:joshpat...@gmail.com]
Gesendet: Dienstag, 6. Oktober 2009 18:54
An: Weigel, Stefan
Cc: 'Tony Graziano'; sipx
gericht München, HRB 175682
Für Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720
Von: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Gesendet: Dienstag, 6. Oktober 2009 13:51
An: Weigel, Stefan
Cc: Josh Patten; sipx-users@list.sipfoundry.org
Betreff: Re: [sipx-users] sipXbrige questions
A.) It appears you
ditive manner in that you add an alias to that
extension that matches the DID number so that DID matching can occur.
For example, the number you provided would have the alias 089123456978
added to it (or whatever DNIS information the telco is sending down the
line). The alias field is at the bottom of
Hi folks,
I'm playing around with the new 4.0.2 release and currently being stucked with
some issues related to sipXbridge:
a) I created a trunk with an itsp account for QSC (sip.qsc.de), it's
working so far, I can place calls an get calls from the outside. The only thing
are some error
Hi there,
again I'm stuck with a problem, setup as follows:
SIP Provider (sip.qsc.de) <--> WWW <--> Patton Box <--> SipXecs (with park
server) <--> Polycom Soundstation phones
I made sure all devices are using the G711u codec (otherwise I had problems
with white-noise). So my patton only speak
Hi,
we have a strange problem with white-noise with calls going through the ACD.
We tested various setups and SipXecs versions (stable to source repository),
always with the same result.
The setup:
VoIP Gateway (sip.qsc.de) <> Internet <-- --> Patton SN4638 <-- -->
SipXecs (with ACD) <--
19 matches
Mail list logo
