Hey All,
I am working on a SIP Call bridging application. Every time I receive a
incoming call in Asteriskserver1 my AGI should alert to AsteriskServer2 and
AsteriskServer2 should callback to AsteriskServer1 and call should be bridged
on specified extension.
(making call in this way is
Dear All,
I have installed GNU gatekeeper in my machine. I tested the calls using
gatekeeper successfully.
Now I have tried to Disable the gatekeeper in oh323.conf file
gatekeeper=DISABLE
Now I have tried to call, but the connection is not established. I have
got
On Tue, Jul 14, 2009 at 02:10:47PM +0530, velusamy velu wrote:
Dear All,
I have installed GNU gatekeeper in my machine. I tested the calls using
gatekeeper successfully.
Now I have tried to Disable the gatekeeper in oh323.conf file
gatekeeper=DISABLE
Now I have
Is it possible to have several clients behind NAT to register to an
Asterisk-server with a public IP-address ?
When Asterisk receives an incoming call, how will it know @ which
private IP-address the client is reachable ?
I guess it is impossible for Asterisk to directly contact the private
jonas kellens wrote:
Is it possible to have several clients behind NAT to register to an
Asterisk-server with a public IP-address ?
When Asterisk receives an incoming call, how will it know @ which
private IP-address the client is reachable ?
I guess it is impossible for Asterisk to
Hello everybody,
I was wondering what is postponing the 1.4.26 release? I thought it was
scheculed for last week.
Is there something we can do to help to release this version?
There is no more issue reported on https://issues.asterisk.org/ for the time
being.
Best Regards,
-- --
Marc LEURENT
2009/7/14 gergis.rasmy gergis.ra...@gmail.com:
could anyone help explaining what does this error mean?
i get this error when make a video/ audio call from X-lite to Bria prof.
phone
rtp.c:1739 ast_rtp_read: Unknown RTP codec 126 received from '10.0.0.26'
Gres
To quote Counterpath, 126 is
jonas kellens schrieb:
Is it possible to have several clients behind NAT to register to an
Asterisk-server with a public IP-address ?
When Asterisk receives an incoming call, how will it know @ which
private IP-address the client is reachable ?
I guess it is impossible for Asterisk to
You could use global variables to record when and where parks occurred.
The issue I would see with this (besides perhaps being cumbersome) is that
you wouldn't have a way to undo the counters when a caller hung up instead
of coming off of park.
_
From:
Marc Leurent schrieb:
I was wondering what is postponing the 1.4.26 release? I thought it was
scheculed for last week.
Is there something we can do to help to release this version?
There is no more issue reported on https://issues.asterisk.org/ for the time
being.
No more issues are
Just been contacted by a UK Enum registrar looking for ITSPs to become
resellers of their Enum registration systems ...
Is anyone using Enum?
Does anyone (other than cynical old me) think that Enum is a spammers best
friend?
Has anyone received a spam VoIP call yet? (ie. one placed directly
Guys,
How would you block inbound call's? for example person who is calling me is
212-555-1212, and I would like to do not receive the calls from this person
and give them busy tone.
What should I write in asterisk config files? and in to witch file should I
write it???
It is a simple ex-girlfriend thing to do, assuming callerid is working
correctly.
- exten = s,1,answer
- exten = s,n/2125551212,Goto(torture|s|1)
- exten = s,n,Dial.
_
From: asterisk-users-boun...@lists.digium.com
VIP Carrier schrieb:
How would you block inbound call's? for example person who is calling me is
212-555-1212, and I would like to do not receive the calls from this person
and give them busy tone.
What should I write in asterisk config files?
core show function CALLERID
Verbose(1,###
Does anyone have any light to shed on:
c_avpair_new: unknown attribute
sip02 asterisk[7796]: rc_avpair_new: unknown attribute 1490026597
We are getting congestion errors on a Pri to telco, and not sure what is
going on.
Thanks
Cary Fitch
___
--
Cary Fitch wrote:
Does anyone have any light to shed on:
c_avpair_new: unknown attribute
sip02 asterisk[7796]: rc_avpair_new: unknown attribute 1490026597
We are getting congestion errors on a Pri to telco, and not sure what is
going on.
Doing a google search gave an indication that
Thanks, we agree.. have reset PRI on telco end and rebooted here and trouble
cleared... for a while anyway. Our PRI card seems to have issues.
CF
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Has anyone played with this phone? i cant seem to get it to work
properly, i manged to get it registered and can make calls from it, but
i havent been able to make it receive calls. Weird thing its that if you
make a call from it and while you are on that call you dial its number
does calls go
Here is what asterisk said
Sorry, but 1 syntax errors and 0 semantic errors were detected. It doesn't
make sense to compile.
On Tue, Jul 14, 2009 at 12:05 PM, Philipp Kempgen philipp.kemp...@amooma.de
wrote:
VIP Carrier schrieb:
How would you block inbound call's? for example person who
The assumption here is that you took Phillipps AEL snippet and put into
extensions.ael. Can you post what you put in?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of VIP Carrier
Sent: Tuesday, July 14, 2009 12:27 PM
To:
None of your stuff mentioned above is working!!!
On Tue, Jul 14, 2009 at 1:27 PM, VIP Carrier vipcarr...@gmail.com wrote:
Here is what asterisk said
Sorry, but 1 syntax errors and 0 semantic errors were detected. It doesn't
make sense to compile.
On Tue, Jul 14, 2009 at 12:05 PM,
what ever he have posted there I have added it, just changed DID
On Tue, Jul 14, 2009 at 1:39 PM, Danny Nicholas da...@debsinc.com wrote:
The assumption here is that you took Phillipp’s AEL snippet and put into
extensions.ael. Can you post what you put in?
On Monday 13 July 2009 17:19:15 Philipp Kempgen wrote:
Tilghman Lesher schrieb:
On Monday 13 July 2009 01:03:48 pm Philipp Kempgen wrote:
Philipp Kempgen schrieb:
Is Asterisk supposed to evaluate #exec's in an #include'd file?
The directive #exec is not permitted in an AEL
The Verbose and If statements should be kosher, the Dial is not.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of VIP Carrier
Sent: Tuesday, July 14, 2009 12:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Un-top-posting...
VIP Carrier schrieb:
How would you block inbound call's? for example person who is calling
me is 212-555-1212, and I would like to do not receive the calls from
this person and give them busy tone. What should I write in asterisk
config files?
On Tue, Jul 14, 2009 at
On Tue, Jul 14, 2009 at 12:44 PM, VIP Carriervipcarr...@gmail.com wrote:
what ever he have posted there I have added it, just changed DID
To help clear things up... what file did you add this to?
-jonathan
___
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John A. Sullivan III wrote:
Hello, all. I'm having a nasty problem with call parking in Asterisk
1.6.1.1 that smells like a bug. When the call returns, it seems to be
returning to a | delimited extension and failing. Here is the output
from the console:
Hi John.
I've just run into the
Marc Leurent wrote:
Hello everybody,
I was wondering what is postponing the 1.4.26 release? I thought it was
scheculed for last week.
Is there something we can do to help to release this version?
There is no more issue reported on https://issues.asterisk.org/ for the
time being.
There
Actually I am facing a problem with H.323 (the standard and the ooh323) with
Asterisk vesion 1.4.25 and I discover the following:
1) Using the standard h323 that come with Asterisk:
The chan_h323.so it is not existed in the /usr/lib/asterisk/modules after doing
the compilation and
Search the archives - we had a big discussion about this phone about six
months ago. If you make it work and want another one I will give you
special price!.
j
On Tue, 14 Jul 2009, Cesar Gonzalez wrote:
Has anyone played with this phone? i cant seem to get it to work
properly, i manged to
On Tue, Jul 14, 2009 at 11:47 AM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Monday 13 July 2009 17:19:15 Philipp Kempgen wrote:
Tilghman Lesher schrieb:
On Monday 13 July 2009 01:03:48 pm Philipp Kempgen wrote:
Philipp Kempgen schrieb:
Is Asterisk supposed to
It should be an easy one for many of the experts here.
On Mon, Jul 13, 2009 at 8:10 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
For a new project, I have written a dialplan and it is pretty straight
forward: The [dialout] context dials out a number, and h extension in this
context writes
Tilghman Lesher schrieb:
On Monday 13 July 2009 17:19:15 Philipp Kempgen wrote:
Is there a specific reason not to permit #exec in AEL files?
It wasn't coded that way, and it's parsed in a completely different way than
any other Asterisk configuration file. I don't know the reason Murf
Dear;
I would like to ask: when Asterisk was registering on the gnugk, both (asterisk
and gnugk) were on the same hardware machine and same IP address? Can they be
on the same IP address?
In case they were on the same IP address then: I am afraid the oh323 channel in
asterisk will respond
Zeeshan Zakaria schrieb:
For a new project, I have written a dialplan and it is pretty straight
forward: The [dialout] context dials out a number, and h extension in this
context writes the CDR. But what is happening is that if the callee hangs up
first, all values in the CDR are fine, but if
On Tuesday 14 July 2009 15:35:20 Philipp Kempgen wrote:
Tilghman Lesher schrieb:
On Monday 13 July 2009 17:19:15 Philipp Kempgen wrote:
Is there a specific reason not to permit #exec in AEL files?
It wasn't coded that way, and it's parsed in a completely different way
than any other
Benny Amorsen schrieb:
Last concern: Does setvar work even for transfers, like accountcode
does?
I can't answer your question, but transfer != transfer. Some use
a feature code in Asterisk, some initiate a transfer on their phone,
some use a way to call the Transfer() application.
Mixing it up
Philipp Kempgen schrieb:
Benny Amorsen schrieb:
Last concern: Does setvar work even for transfers, like accountcode
does?
I can't answer your question, but transfer != transfer. Some use
a feature code in Asterisk, some initiate a transfer on their phone,
some use a way to call the
Howdy,
Getting ready to play with QoS settings. We have an asterisk 1.4.23
server running in a colo bunker in the US Virgin Islands under a large
radio tower. That tower has multiple sector radio/antenna pairs that
blanket a valley in 802.11a. The customers have directed dishes aimed at
Jeff LaCoursiere wrote:
Search the archives - we had a big discussion about this phone about six
months ago. If you make it work and want another one I will give you
special price!.
j
Jeff, yeah i saw the posts, i followed Bob Pierce config and had no
luck, BUT it just started to
Tilghman Lesher schrieb:
On Tuesday 14 July 2009 15:35:20 Philipp Kempgen wrote:
Tilghman Lesher schrieb:
On Monday 13 July 2009 17:19:15 Philipp Kempgen wrote:
Is any *.conf file (which permits #exec) guaranteed to be read before
extensions.ael? It would then be possible to (ab)use an
This was fixed in the 1.6.1 SVN, and I would guess that it was also fixed in
the 1.6.0.
SVN log:
r189951 | russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines
Fix call parking callback. Pipes - Commas.
You will have to create a patch against the 1.6.0 source, but you could
start
On Tue, 14 Jul 2009, Cesar Gonzalez wrote:
Jeff LaCoursiere wrote:
Search the archives - we had a big discussion about this phone about six
months ago. If you make it work and want another one I will give you
special price!.
j
Jeff, yeah i saw the posts, i followed Bob Pierce config
Hi guys,
Hope someone can help me with this... I got an Asterisk with TE120P card...
using dahdi 2.1.0.4 and asterisk 1.4.24.
The problem is the link between the card and the telco... suddenly through
the day looses connection... and the people on site just power cycle the PC
and the problem got
I think an equally interesting question is whether the Federal Trade
Commission (and foreign equivalents) draw a distinction between calls to
E.164 numbers based on their transport technology. In other words, is there
a legal difference depending on whether the call touches the PSTN vs. being
Jonathan Thurman wrote:
This was fixed in the 1.6.1 SVN, and I would guess that it was also
fixed in the 1.6.0.
SVN log:
r189951 | russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines
Fix call parking callback. Pipes - Commas.
You will have to create a patch against
Barry L. Kline wrote:
I'll figure out how to make this patch
against 1.6.0.10.
That was a trivial fix. I hope that they permanently add that patch to
the 1.6.0.x series.
Thanks again Jonathan.
Barry
___
-- Bandwidth and Colocation Provided
On 13/7/09 2:23 PM, eric weaver wrote:
I am doing a little application to originate a call through Asterisk via
AMI (Perl Asterisk::Manager).
It logs in successfully, does an originate command with
Exten: 0020 (which is set up to answer and wait for 60 then hang up)
Channel:
On Tue, Jul 14, 2009 at 2:19 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Dear;
I would like to ask: when Asterisk was registering on the gnugk, both
(asterisk and gnugk) were on the same hardware machine and same IP address?
Can they be on the same IP address?
If I understand your
On Tue, Jul 14, 2009 at 06:46:50PM -0500, Karl Fife wrote:
[snip]
missed the original message
- Original Message -
From: Gordon Henderson gordon+aster...@drogon.net
To: Asterisk Users Mailing List Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 14, 2009 9:14 AM
Turns out I was using the wrong units in the TIMEOUT parameter to the
Manager Originate command... It was supposed to be milliseconds and I put
15. D'o Was timing out before it got started.
Now it connects but odd things happen. But there are two NATting firewalls
between the two
On 15/7/09 1:34 PM, eric weaver wrote:
Turns out I was using the wrong units in the TIMEOUT parameter to the
Manager Originate command... It was supposed to be milliseconds and I
put 15. D'o Was timing out before it got started.
Now it connects but odd things happen. But
The answer, quickly, is No, ENUM is not safe from spam. But there
is security in obscurity at the moment. Since nobody really uses
ENUM, it's not been brought to the attention of phone spammers.
However, witness AOL AIM, or Skype - now that people know it exists
and there are millions
Is there a way to transfer a call, while in the middle of the call, using
DTMF?
___
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On 15/7/09 3:07 PM, Michael wrote:
Is there a way to transfer a call, while in the middle of the call, using
DTMF?
Yep, just pass the t or T options to the dial command and set it up in
/etc/asterisk/features.conf
--
Cheers,
Matt Riddell
Director
Yes,
In the features.conf under featuremap you need the blindtransfer un-commented
blindxfer = ##
Then in your extensions.conf you need to have at least a capital T
exten = example,1,Dial(ZAP/4/12345,,T)
Then during the call you can press ## and asterisk will say transfer.
Then dial in the
Hi
The subject line says it all how do I enable this style of call.
Pointers to the dns setup and asterisk setup would be great
or even search words for google, as I am not sure how to search for this
type of request.
Alex
--
There is no instance of a country having benefited
from
Alex Samad wrote:
Hi
The subject line says it all how do I enable this style of call.
Pointers to the dns setup and asterisk setup would be great
or even search words for google, as I am not sure how to search for this
type of request.
Alex
Alex,
Here's a good place to start.
On Wed, 2009-07-15 at 14:34 +1000, Alex Samad wrote:
Hi
The subject line says it all how do I enable this style of call.
Pointers to the dns setup and asterisk setup would be great
or even search words for google, as I am not sure how to search for this
type of request.
Alex
snip
If
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