Hello,
I have a Google Voice phone number and want to connect it to my asterisk box
to have calls handled to my SIP account.
When I call the number I receive the correct INCOMING request on Jabber
portion of asterisk, but the call is not connected to the gtalk part.
JABBER: asterisk INCOMING: htt
Hey Guys,
i would like to write the output from my bash script into a Variable, that is
readable by Asterisk
Using this:
Set(var1=${FILE(/dev/shm/tempfile.txt,0,6)})
is not very helpful because this command reading fixed character length. If i
read 6 characters and in the file only 3 i get "1
Ive found trying to make this work is easier (esp in a LAN environment if you
disable fax relay in the dial-peer and set it for passthrough). This solved the
intermittent problems I had as well as the unknown rtp issues.
I can post an example when not on my mobile.
Jared Mauch
On Apr 16, 201
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 http://packages.a
Hi,
I have Asterisk 1.4.10 under LMCE (upgrade is not an option) and have this
strange error appearing in full log :
[Apr 16 14:35:48] NOTICE[10802] chan_sip.c:-- Registration for
'num...@voip.siol' timed out, trying again (Attempt #22)
[Apr 16 14:35:48] WARNING[10802] chan_sip.c: No s
On Sat, Apr 16, 2011 at 1:56 AM, Steve Underwood wrote:
> On 04/16/2011 07:25 AM, Ryan Wagoner wrote:
>>
>> On Fri, Apr 15, 2011 at 7:00 PM, sean darcy wrote:
>>>
>>> Using spandsp-0.0.6-pre18, the Jan 22 release.
>>>
>> You might try using spandsp-0.0.6-pre17. That version works great for
>> me
On 04/16/2011 08:47 PM, Ryan Wagoner wrote:
On Sat, Apr 16, 2011 at 1:56 AM, Steve Underwood wrote:
On 04/16/2011 07:25 AM, Ryan Wagoner wrote:
On Fri, Apr 15, 2011 at 7:00 PM, sean darcywrote:
Using spandsp-0.0.6-pre18, the Jan 22 release.
You might try using spandsp-0.0.6-pre17. That
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 and solved?
How should i set sip.conf and udptl.conf.
I alrea
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: asterisk-users@lists.dig
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 request
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 => _*11*,1,Set(D
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 suc
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 an
I found a solution that works fine for me
Set(var1=${SHELL(shellcommand)})
Bye Daniel
> Von: Daniel Knoll
> Datum: 16. April 2011 13:13:28 MESZ
> An: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Betreff: write system command output into a variable
>
> Hey Guys,
>
> i would
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 sipaxclie
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 its
-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:
60
-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
V
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On 4/16/2011 8:20 PM, bayardo.sanc...@gmail.com wrote:
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stop it.
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