gt; The above works just fine for doing what I want to do, e. g. pass a
> parameter from an Asterisk dialplan context into a queue-triggered "agent
> just answered in the queue" Asterisk macro.
>
> Thanks for the reply!
>
> Kind regards
>
> Stefan
> -Origi
Hi,
the tcpdump starts with a pretty standard INVITE sequence:
10.0.0.121 -- INVITE --> 10.0.0.3
10.0.0.121 <-- 401 Unauthorized -- 10.0.0.3 // asterisk gives nonce in
WWW-Authenticate: header
10.0.0.121 -- ACK --> 10.0.0.3
After that, normally you would see a new INVITE from the phone
Hi,
maybe I am overlooking something, but channel variables should be thread safe,
shouldn't they?
I am using the following (sorry, in ael):
macro dial-queue (number) {
Set(_ORIG_UNIQUEID=${UNIQUEID});
Queue(${number},rCt,,,${timeout},,set-dst-agent);
..
}
// the
Hello everybody,
I am seeing a strange problem on Asterisk 1.8 with dnsmgr.
The number of entries in DNS Manager seems to be growing steadily and all are
pointing to the sama host - a SIP trunk to a local provider, which uses SRV
lookup.
So, when DNS manager refreshes, there are 6000+
On 14.09.2015, at 21:58, Sebastian Kemper wrote:
> So I got rid of the firewall rule that opened the RTP ports. And then it
> dawned on me that I don't even need to open the 5060 port. The REGISTER
> requests established a UDP connection that the kernel's conntrack module
On 26.11.2014, at 22:08, Antoine Megalla aa...@rocketmail.com wrote:
The asterisk installation went fine but as soon as I start asterisk
executable it loads everything and then after the Ready line the process
gets killed and when I try to run it again i get: /usr/sbin/asterisk :
command
On 22.11.2014, at 13:40, Yves A. yves...@gmx.de wrote:
I have a really strange problem which is driving me crazy for days now.
If I register my asterisk (tried all versions from 1.6 up to 13.x) with one
sip registrar,
everything works... calls go out and call come in... no 32 seconds limit.
... 'cause message file names start with 0 (msg.wav).
--
marie
On 05.10.2014, at 18:45, Leandro Dardini ldard...@gmail.com wrote:
Hello,
have you noticed the message num (VM_MSGNUM) is off by one?
For example, I receive the following message:
Just wanted to let you know you were
calls
correctly even when transferred early.
On Tue, Sep 23, 2014 at 1:41 PM, Michael Keuter li...@mksolutions.info
wrote:
Am 23.09.2014 um 19:49 schrieb Marie Fischer ma...@vtl.ee:
Hi everybody,
I'm looking for a solution for the following scenario:
• Asterisk queue
... and to continue my thought, if nothing else is possible, would it be a Very
Bad Idea to just delete the ABANDON log (queue_log goes to mysql via odbc)
automatically after it's created? In h extension?
--
marie
On 05.10.2014, at 20:42, Marie Fischer ma...@vtl.ee wrote:
Thanks for your
Hi everybody,
I'm looking for a solution for the following scenario:
• Asterisk queue
• At peak hours, there will be more callers then queue members/agents, so some
callers will spend some time on hold
• Agents should be able to choose which of the on hold calls to answer instead
of answering
On 01.09.2014, at 11:42, Lukasz Sokol el.es...@gmail.com wrote:
On 31/08/14 17:40, Marie Fischer wrote:
Le 29 août 2014 14:30, Marie Fischer ma...@vtl.ee a écrit :
we use OSX CardDAV server and its response is very slow, so we
ended up syncing all the CardDAV contacts to MySQL via cron
On 29.08.2014, at 22:44, Olivier oza.4...@gmail.com wrote:
Le 29 août 2014 14:30, Marie Fischer ma...@vtl.ee a écrit :
Hi,
we use OSX CardDAV server and its response is very slow, so we ended up
syncing all the CardDAV contacts to MySQL via cron. Asterisk dialplan then
runs a query
Hi,
we use OSX CardDAV server and its response is very slow, so we ended up syncing
all the CardDAV contacts to MySQL via cron. Asterisk dialplan then runs a query
defined in func_odbc.conf.
--
marie
On 29.08.2014, at 15:03, Lukasz Sokol el.es...@gmail.com wrote:
Hi,
Would it be hard /
Hello everybody,
is there any way to find out when the queue stats ('queue show' / AMI action
'QueueStatus') was last reset (by 'queue reset stats')? These counters would
make much more sense if I knew what timeframe they cover. ;)
--
marie
--
On 06.06.2013, at 15:05, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello,
when picking up an incoming call from one ip phone on another ip phone, the
call terminates after about 5 to 10 seconds.
When reading out the hangup cause variable in the h-extention of the
dialplan, the
On 23.05.2013, at 12:57, bilal ghayyad bilmar...@yahoo.com wrote:
There is no free channel to be used to have integration between asterisk and
skype? What is the software that I can use to send and receive chat messages
on skype network?
For voice calls, you could try Skype Connect, which
On 21.05.2013, at 0:05, Tommy Cooper tomcoope...@yahoo.com wrote:
Hi,
I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is
generating are failing. I am trying to run Sipp on the same machine as
Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command.
Do you have a peer
is running and listen to some message played
via Background().
- Forwarded Message -
From: Marie Fischer ma...@vtl.ee
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 22, 2013 1:16 PM
Subject: Re: [asterisk-users] Stress
On 04.05.2013, at 20:20, Olivier oza_4...@yahoo.fr wrote:
Le 2 mai 2013 13:23, Marie Fischer ma...@vtl.ee a écrit :
from time to time, we get so-called simplex / one-way audio calls, where one
party cannot hear the other. The only thing in common is that is does happen
with calls via SIP
Hello everybody,
from time to time, we get so-called simplex / one-way audio calls, where one
party cannot hear the other. The only thing in common is that is does happen
with calls via SIP trunk, not ISDN and not internal calls. Nothing strange in
verbose and SIP logs. Could even be some
On 09.04.2013, at 10:33, Shanavaz E A shanava...@yahoo.com wrote:
Hi,
From asterisk 1.8, the CDR table is not logging the unanswered or extn busy
calls which hit while in the queue. I am talking about this setting in the
cdr.conf :
; In brief, this option controls the reporting
On 09.04.2013, at 23:12, Thomas Perron thomas.per...@gmail.com wrote:
This seems basic but something is missing.
I dial from my cell phone to my DID and enter the context in extensions.conf
I am hoping to cascade through the plan and successfully automatically dial
the 1444 number
On 09.04.2013, at 23:43, Nick Khamis sym...@gmail.com wrote:
That's just it! Nothing! It just does not pass the 91 mark. There are
no failed calls during the test:
Successful call|0 |20802
Failed call|0 |0
Hello everybody,
I am having a problem with realtime SIP peers.
On Asterisk 1.8, I had SIP peers for external SIP providers configured in
database and additional register lines in sip.conf so they would register.
Now I upgraded to Asterisk 11.3.0, partly because of the promised
Hello everybody,
I am trying to find an intermittent SIP error with one provider and thought the
best first step would be to have sip set debug on for some days and check the
logs.
Everything gets logged nicely, but the SIP log clutters up the console quite
badly. Is it possible to have SIP
On 29.03.2013, at 15:05, Doug Lytle supp...@drdos.info wrote:
Marie Fischer wrote:
full = notice,warning,error,debug,verbose,dtmf,fax
You should have a log called full in:
/var/log/asterisk
Sure I do and happy with that. :)
The point is, I also have my Asterisk console full of SIP
-users
--
---
See our Docs, FAQs, etc at: http://snom.com/wiki
---
Sitz/Domicile:
snom technology AG Gradestraße 46 D-12347 Berlin
Sven Fischer
Hi,
try our latest beta version 6.5.2 which can be found here:
http://www.snom.com/wiki/index.php/Snom360/Firmware/Beta_Versions
http://www.snom.com/wiki/index.php/Snom320/Firmware/Beta_Versions
http://www.snom.com/wiki/index.php/Snom300/Firmware/Beta_Versions
Release Notes:
Berlin
Sven Fischer fax +49 30 39833111 PSTN/ENUM +49 30 39833434
mailto:[EMAIL PROTECTED] http://www.snom.com
--
--
---
See our Docs, FAQs
Hi,
maybe this helps ?
http://www.snom.com/wiki/index.php/DHCP
http://www.snom.com/wiki/index.php/Massdeployment_Firmware_Release_5
For further questions in that regard feel free to contact us at
[EMAIL PROTECTED] !
Regards.
On Friday 02 June 2006 05:49, Remco Barendse wrote:
Hi list!
technology AG Gradestraße 46 D-12347 Berlin
Sven Fischer fax +49 30 39833111
mailto:[EMAIL PROTECTED] http://www.snom.com
---
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.html?L=1
Whitepapers at: http://www.snom.com/white_papers.html
---
snom technology AG Gradestraße 46 D-12347 Berlin
Sven Fischer fax +49 30 39833111
mailto:[EMAIL PROTECTED] http
Hi all,
I have a problem: On my internal S0 where phones are connected via HFC I get
all the number with a leading 0 (either from internal SIP phones or external
dialins via CAPI). I don't know where to look for this 0. Any ideas?
Greetings, Sven
--
Sven Fischer (Dipl.-Phys.) - FACIT
]
1
Unnumbered frame:
1 SAPI: 63 C/R: 1 EA: 0
TEI: 127EA: 1
1M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ]
5 bytes of data
== Primary D-Channel on span 1 down for TEI 64
--
Sven Fischer (Dipl.-Phys.) - FACIT Consulting GmbH
Hausinger Str. 6 - 40764 Langenfeld
Am Dienstag, 7. Februar 2006 09:38 schrieb Sven Fischer:
Hello all,
if I try to call from one phone on the internal S0 to another on the same
S0 using zaphfc, the bus is hung up. The called phone is ringing, but I
can't talk from one phone to the other. The error I get is:
-- Executing
Am Dienstag, 7. Februar 2006 12:55 schrieb Francesco Peeters (Asterisk):
On Tue, February 7, 2006 9:53, Sven Fischer said:
Am Dienstag, 7. Februar 2006 09:38 schrieb Sven Fischer:
Hello all,
if I try to call from one phone on the internal S0 to another on the
same
S0 using zaphfc
toy to play with, correct?
-Original Message-
From: Sven Fischer (support) [mailto:[EMAIL PROTECTED]
Sent: Friday, January 27, 2006 5:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Announcement: Snom 360 with integrated XML
O bjects
://www.snom.com/white_papers.html
---
snom technology AG Gradestraße 46 D-12347 Berlin
Sven Fischer fax +49 30 39833111
mailto:[EMAIL PROTECTED] http://www.snom.com
--
---
See our FAQs at: http://www.snom.com/faq0.html?L=1
Whitepapers at: http://www.snom.com/white_papers.html
---
snom technology AG Gradestraße 46 D-12347 Berlin
Sven Fischer fax +49 30
://www.snom.com/white_papers.html
---
snom technology AG Gradestraße 46 D-12347 Berlin
Sven Fischer fax +49 30 39833111
mailto:[EMAIL PROTECTED] http://www.snom.com
On Saturday 31 December 2005 01:57, Ross C wrote:
... and 2 Snom 320's (now discontinued I think).
No, they are not discontinued !!!
Regards,
Sven
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Asterisk-Users mailing list
To
This doesn't seem to be correct, too...
Sven
On Monday 02 January 2006 17:43, Ross C wrote:
Sorry!!
Just discontinued @ voipsupply.com I guess.
Thx for the correction.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sven Fischer
(support)
Sent
On Friday 23 December 2005 00:39, Steven Ringwald wrote:
On Thu, 2005-12-22 at 23:34 +0100, Christian Stredicke wrote:
Try loading
http://phone-ip-address/line_sip.htm?settings=saveuser_dp_str1= (if
that was in the line 1) while the phone boots up (keep your finger on
the reload button).
in brackets
(snom320).
Regards,
Sven Fischer
On Wednesday 12 October 2005 20:20, Franklin Webb wrote:
Greetings fellow list members,
It seems like a lot of people have been having trouble getting
indicators working on the Snom phones, myself included. Recently I was
able to get the desktop
---
snom technology AG Gradestraße 46 D-12347 Berlin
Sven Fischer fax +49 30 39833111tel +49 30 39833444
mailto:[EMAIL PROTECTED] http://www.snom.comsip:[EMAIL PROTECTED
Hi
all
i'm just starting
to setup my "own" asterisk. My first
question is, if there is any reason to choose aspecial linux
distribution or if it doesn't mater which
distribution i chosse. Is there anything i should be aware
of?
Thanks a lot for
your help!
Greetings
Frank
technology AG Pascalstraße 10b D-10587 Berlin
Sven Fischer fax +49 30 39833111
mailto:[EMAIL PROTECTED] http://www.snom.comsip:[EMAIL PROTECTED
Fischer fax +49 30 39833111
mailto:[EMAIL PROTECTED] http://www.snom.comsip:[EMAIL PROTECTED]
---
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Asterisk-Users mailing list
[EMAIL PROTECTED]
http
special offers for a snom 4S SIP Registrar/Proxy SME
Version! For further information contact: [EMAIL PROTECTED] !
---
snom technology AG Pascalstraße 10b D-10587 Berlin
Sven Fischer fax +49 30 39833111
! For further information contact: [EMAIL PROTECTED] !
---
snom technology AG Pascalstraße 10b D-10587 Berlin
Sven Fischer fax +49 30 39833111
mailto:[EMAIL PROTECTED] http://www.snom.comsip
] !
---
snom technology AG Pascalstraße 10b D-10587 Berlin
Sven Fischer fax +49 30 39833111
mailto:[EMAIL PROTECTED] http://www.snom.comsip:[EMAIL PROTECTED
Hi,
On Monday 11 October 2004 19:12, Dave Cotton wrote:
On Mon, 2004-10-11 at 11:51 -0500, Mike Meyer wrote:
Someone pointed me here
http://www.snom.com/downloads/share
http://www.snom.com/download/share
!
That where the SNOM support team sent me. Seems that they may be
Hi,
no, it is
http://www.snom.com/download/share
!!!
Sven
On Monday 11 October 2004 17:18, Alex Barnes wrote:
Someone pointed me here
http://www.snom.com/downloads/share (had to guess at URL as the Snom
site appears to be down or uber slow but if that's not it its damn close
:-P )
Hi,
do a SIP trace or PCAP trace of the scenario via the webinterface and you will
see exactly, what is going on...
Regards,
Sven
On Thursday 14 October 2004 21:53, Magnus Jungsbluth wrote:
Hi,
I'm running the 1.0 release of Asterisk an it is working nicely with our
snom 105 phones. Hold
fkey4: dest sip:[EMAIL PROTECTED];user=phone
/pre
/html
BTW did you saw our FAQ regarding massdeployment ?
kind regards,
Sven Fischer
Dear Friends of Ubiquity Software:
As you may have noticed, Ubiquity Software began using the web domain
ubiquity.com earlier this year in addition
Hi,
On Friday we changed our Telco-Provider (from German Telekom to Mnet) and recieved new
Numbers. I changed the extensions in extension conf to match the new numbers. But i
always get:
Jul 18 12:10:39 WARNING[245776]: pbx.c:1780 ast_pbx_run: Channel
'CAPI[contr1/89064934]/0' sent into
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