On 4/14/11 5:03 PM, m...@tdiehl.org wrote:
On Thu, 14 Apr 2011, Vahan Yerkanian wrote:
A word of notice: asterisk/digium yum repos xmls haven't been updated
yet (properly):
Yes, I noticed that also. For some reason the latest Dahdi rpms are
sitting in
the top level dir at
On 04/16/2011 08:47 PM, Ryan Wagoner wrote:
On Sat, Apr 16, 2011 at 1:56 AM, Steve Underwoodste...@coppice.org wrote:
On 04/16/2011 07:25 AM, Ryan Wagoner wrote:
On Fri, Apr 15, 2011 at 7:00 PM, sean darcyseandar...@gmail.comwrote:
Using spandsp-0.0.6-pre18, the Jan 22 release.
You
You must have 1.8+ its already been posted the 1.6 didn't get a backport fix
in the jabber protocol.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Saturday, April 16, 2011 3:57 AM
To:
dig @193.189.160.13 voip.siol
; DiG 9.6.0-APPLE-P2 @193.189.160.13 voip.siol
; (1 server found)
;; global options: +cmd
;; Got answer:
;; -HEADER- opcode: QUERY, status: REFUSED, id: 35478
;; flags: qr rd; QUERY: 1, ANSWER: 0, AUTHORITY: 0, ADDITIONAL: 0
;; WARNING: recursion requested but not
Hi all,
There is a feature very common in PBX called PADLOCK , and I'd like to set
up it on Asterisk 1.6. I have seen it in the internet but such scripts never
work to me. I am trying to do something like that:Create a password and
associate it with the callerid:
exten =
On Apr 16, 2011, at 9:27 AM, Steve Underwood wrote:
On 04/16/2011 01:24 PM, Oguzhan Kayhan wrote:
Hello,
We have a sip trunk end point with cisco media gateway.
VoIP works fine.
But when we try to send faxes thru this trunk, we simply can not.
Is there anybody experienced such problem
Hello, thanks or the quick replies.
I tried with both 1.6.2.9 and 1.6.2.17
My config is sipaxclient-asterisk-(siptrunk)-telcooperator-analogfax
All i know is telco uses cisco on their side..Not sure which version they
are using.
I got t38pt_udptl = yes parameter on sip.conf general.
Didnt make
I found a solution that works fine for me
Set(var1=${SHELL(shellcommand)})
Bye Daniel
Von: Daniel Knoll dan...@danielknoll.de
Datum: 16. April 2011 13:13:28 MESZ
An: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Betreff: write system command
FYI: I have T.38 disabled and have found the delay/jitter acceptable in our
environment over the network with g711ulaw passthrough.
- Jared
On Apr 16, 2011, at 11:28 AM, Oguzhan Kayhan wrote:
Hello, thanks or the quick replies.
I tried with both 1.6.2.9 and 1.6.2.17
My config is
I have a particular DID that when called will prompt the user to enter the
caller id that they want to be displayed followed by it prompting for the
phone number to dial. How would I go about getting thest calls logged in
both CDR and ARI? Currently, only the callerid information from the original
bumping once before sending it to the tracker.
Original Message
Subject: [asterisk-users] sip error logging
Date: Fri, 15 Apr 2011 03:39:23 -0400
I recently noticed that asterisk is not logging unknown sip connections.
I'm not sure if I've broken something or if asterisk
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi!
Are hints not yet implemented in res_jabber?
I have this here:
exten = 3000,hint,gtalk/gtalk_account/mari....@gmail.com
But the hint doesn't show any difference. It always shows online on the
phone and core show hints always shows that:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
has anyone managed to establish an XMPP connection to the facebook
Jabber servers?
I'd like to send messages on missed calls vie FB.
- -S
- --
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
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On 4/16/2011 8:20 PM, bayardo.sanc...@gmail.com wrote:
I listened to your email using DriveCarefully and will respond as soon as I can.
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stop it.
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Jeremy Kister
http://jeremy.kister.net./
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