Hi Geeks,
I am a beginner in asterisk, I read about native bridging option in
asterisk which allows the RTP streaming through the SIP media terminals
after initiating the call . I identified the following features are getting
affected
by this feature in my testing.
1) Call transfer.
2) M
Hello,
I am using Asterisk 1.4.21.2 in a Centos 4.8 with a kernel version
2.6.9-89.ELsmp. The processor type is Intel(R) Xeon(R) Quad Core CPU
E5410 @ 2.50GHz. with 4 GB of RAM
Sometimes, I get a strange behavior from asterisk: The CLI commands does not
work and Asterisk cannot receive calls. als
Hi Steve,
So is this a bug in Asterisk 1.6? Has anyone verified/reported this
issue?
Warm Regards
Venugopal G
HNM-SO WiMAX CPE VoIP IOT Team
Cell : +91-99723-99437
*
Hi Dave,
Sure enough my astdb does contain references to VM files as shown with
strings - doing the database dump however does not show the references.
I'm not sure about the internals of how Berk DB works, however I;m also
seeing references to lots of other data that really shouldn't be part of
Receiving a fax pstn - pstn with 1.6.2.6-rc2:
-- Executing [...@incoming-pstn-line:1] Answer("DAHDI/4-1", "") in
new stack
-- Executing [...@incoming-pstn-line:2] Wait("DAHDI/4-1", "3") in new
stack
-- Executing [...@incoming-pstn-line:3] Dial("DAHDI/4-1",
"DAHDI/g0,36") in new s
Zoa wrote:
On friday we finally released Attrafax under a GPL2 license.
It comes with its own set of modems and built in transparent gatewaying.
The solution should be quite stable as long as the line quality is ok.
(Some tools for measuring the line quality are included in the release,
as wel
Anybody work out how to fix this?
Asterisk 1.4.26.3
Sip Trunk inbound -> to Queuee -> Outbound to two sip stations, and one sip
trunk.
sip trunk caller answers, queue shows "ring+inuse" , core show channels
shows inbound/outbound
after caller hanges up, no channels in use, queue sti
Does anyone know if Asterisk can function as a voicemail system for a
Nortel Option 11 PBX? We will be connecting Asterisk to act as an IVR
before sending calls to the Nortel and as a Voicemail system in case the
user does not answer. That part is trivial, the only problem we have is
that
The backtrace is not useable. Try to rebuild Asterisk with the "Don't
Optimize" Option ("make menuconfig" and the the build options)
regards
klaus
Edwin Lam wrote:
> Philip A. Prindeville wrote:
>> On 03/08/2010 04:31 PM, Edwin Lam wrote:
>>> hi folks.
>>>
>>> i recently upgraded asterisk to 1.6
> Hi!
>
> > I've to deploy about 200 snom320 phones on a instalation.
> > Do you know any knid of tool to help me with this amount of phones?
> > I'm thinking in a provisioning tool which I use for setting up the
> > phones.
>
> Look here:
> http://www.voip-info.org/wiki/view/Asterisk+phone+snom#
I've just started switching my project to use confbridge instead of
meetme and app_conference (because of audio glitches that kept appearing
in those applications).
However, I can't find any way to interact with an existing confbridge
conference. Surely there's some equivalent to meetme's 'meet
On Mar 8, 2010, at 6:16 PM, sean darcy wrote:
>
> And without doing anything more, it now Just Works(TM). Sunspots possibly.
>
> sean
Glad it's working... those sunspots are nasty. :)
---fred
http://qxork.com
--
_
-- Bandwi
Fred Posner wrote:
> On Mar 5, 2010, at 1:01 PM, sean darcy wrote:
>
>> The issues are that sip doesn't work,
>
>
> What does "doesn't work" mean? In / Out? Both? Do you have a sip trace?
>
>> even though this same set up
>> worked with POTS dsl. IAX does (but gives lousy audio quality) so I
>
voip crazy wrote:
> Hello all,
>
> I've to deploy about 200 snom320 phones on a instalation.
> Do you know any knid of tool to help me with this amount of phones?
> I'm thinking in a provisioning tool which I use for setting up the
> phones.
>
> Any clue would be welcomed.
>
> Thanks.
>
> Voip-Craz
On Wed, Mar 10, 2010 at 3:15 AM, voip crazy wrote:
> Hello all,
>
> I've to deploy about 200 snom320 phones on a instalation.
> Do you know any knid of tool to help me with this amount of phones?
> I'm thinking in a provisioning tool which I use for setting up the
> phones.
>
> Any clue would be w
Hi!
> I've to deploy about 200 snom320 phones on a instalation.
> Do you know any knid of tool to help me with this amount of phones?
> I'm thinking in a provisioning tool which I use for setting up the
> phones.
Look here:
http://www.voip-info.org/wiki/view/Asterisk+phone+snom#Miscellaneous
Phi
Hello all,
I've to deploy about 200 snom320 phones on a instalation.
Do you know any knid of tool to help me with this amount of phones?
I'm thinking in a provisioning tool which I use for setting up the
phones.
Any clue would be welcomed.
Thanks.
Voip-Crazy
--
___
On 3/8/2010 12:55 PM, Kevin P. Fleming wrote:
> Dean Hoover wrote:
>
>> Our company has an Asterisk server where one of the T1 is connected to
>> an IVR. Asterisk is configured for FXO Loopstart, and the IVR is
>> configured FXS.
>
> This is under control of the dialplan, though... using Dial(DA
On 9 Mar 2010, at 12:21, Gopalakrishnaiyer Venugopal-Q16770 wrote:
> My SIP server (SONUS) is making a call to Asterisk DAHDI line with
> Caller Identity restricted. The asterisk is displaying the caller id
> of
> the caller eventhough they are not supposed to be shown.
>
> Kindly throw some lig
Hi Bob,
Thanks for replying. I've thought of doing that, but softkeys are limited
and for a phone with many call appearances (4-5) that would be using many of
the softkeys.
Mike
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.
Hi,
I am having a problem with (Asterisk is crashing) with a Fritz card PCI
/ chan_capi.
Receiving Calls from PSTN works, but outbound calls make asterisk crash
(Speicherzugriffsfehler/Segmentation fault). The crash occurs upon
dialing with the other phone not even ringing.
I hereby ask if some
Will Payne wrote:
> it just seemed like a 'I know this is wrong, but...' comment :)
> Quoting entire emails is bad, m'kay. Quoting whole threads is worse. If you
> snip the quote down to the relevant portion, you can reply where you like,
> regardless of what's gone on beforehand.
>
> (Surely th
Gopalakrishnaiyer Venugopal-Q16770 wrote:
>
> Caller Identity restricted. The asterisk is displaying the caller id of
> the caller eventhough they are not supposed to be shown.
>
>
core show application setcallerpres
hylafax*CLI>
-= Info about application 'SetCallerPres' =-
[Synopsis]
Set
Hai All,
My SIP server (SONUS) is making a call to Asterisk DAHDI line with
Caller Identity restricted. The asterisk is displaying the caller id of
the caller eventhough they are not supposed to be shown.
Kindly throw some light on this issue
Regards
Venugopal
--
___
On 9 Mar 2010, at 11:47, SIP wrote:
> Different entirely. People who switch to bottom posting on a top-posted
> thread make things MUCH harder to read by being needlessly pedantic.
it just seemed like a 'I know this is wrong, but...' comment :)
Quoting entire emails is bad, m'kay. Quoting who
Will Payne wrote:
> On 8 Mar 2010, at 22:08, Dave Poirier wrote:
>
>
>> Top posting to remain consistent...
>>
>
>
> I drop litter because everyone else does.
>
> ;)
>
> W
>
>
Different entirely. People who switch to bottom posting on a top-posted
thread make things MUCH harder to read
Hello!
I have problems with audio in conference zap sip, I have choppy audio. I
believe this problem is cause by de echo canceller from the fonebridge that
I use in my system.
Can someone explain me how I can disable the echo canceller form the
fonebridge?
I'm using dual port T1/E1 foneBRIDGE2
Hello
Im configuring an asterisk peer, wich uses port 5060 to send and port
5061 to receive signaling.
So, wich port should I put in my asterisk SIP trunk configuration?
port = 5060
or
port = 5061
?
Thanks
Regards
Joao Pereira
--
_
Hello
Im configuring an asterisk peer, wich uses port 5060 to send and port
5061 to receive signaling.
So, wich port should I put in my asterisk SIP trunk configuration?
port = 5060
or
port = 5061
?
Thanks
Regards
Joao Pereira
--
_
On Tue, Mar 9, 2010 at 10:38 AM, Gopalakrishnaiyer Venugopal-Q16770
wrote:
> Hi,
>
> Yes the public number is connected via DAHDI.Also for incoming fax do we
> need to make any changes?
>
no
--
_
-- Bandwidth and Colocation P
Hi,
Yes the public number is connected via DAHDI.Also for incoming fax do we need
to make any changes?
Warm Regards
Venugopal G
HNM-SO WiMAX CPE VoIP IOT Team
Cell : +91-99723-99437
***
On Tue, Mar 9, 2010 at 6:37 AM, Gopalakrishnaiyer Venugopal-Q16770
wrote:
> HI,
>
> Do we need to make any changes to the chan_dahdi.conf to make sure that the
> asterisk detects fax calls?As mentioned below I will be connecting an analog
> fax machine to the DAHDI channel and will be dialling t
Hi,
This is the output from queue show 28:
47 (DAHDI/g0/12345678) (realtime) (Ringing) has taken no calls yet
Why is the devicestate "Ringing" when no channels is calling this
number, and the queue says "has taken no calls yet"?
Is it picking up the general state of a random channel on g0 in
1) elastix
2) contacq (but there is still a stable version)
2010/3/8 Edwin Quijada
>
> gNUDIALER
> *---*
> *-Edwin Quijada
> *-Developer DataBase
> *-JQ Microsistemas
> *-Soporte PostgreSQL
> *-www.jqmicrosistemas.com
> *-809-849-8087
> *-
Hi all, I'm new in asterisk and I got to set up a dundi config for my work.
I have 2 PBX for the test, the two PBX are in the same local network
PBX A : 192.168.199.23
PBX B : 192.168.199.21
my config files : (on PBX B , the config files on PBX A looks like it)
/etc/asterisk/dundi.conf
[general
On 8 Mar 2010, at 22:08, Dave Poirier wrote:
> Top posting to remain consistent...
I drop litter because everyone else does.
;)
W
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? J
On 9 March 2010 07:58, Peter Childs wrote:
> On 8 March 2010 15:34, Olle E. Johansson wrote:
>>
>> 8 mar 2010 kl. 11.13 skrev Peter Childs:
>>
>>> On 5 March 2010 13:48, Jim Dickenson wrote:
At an Asterisk CLI use the command "manager show commands".
>>>
>>>
>>> Life is rarely that simple,
On 8 March 2010 15:34, Olle E. Johansson wrote:
>
> 8 mar 2010 kl. 11.13 skrev Peter Childs:
>
>> On 5 March 2010 13:48, Jim Dickenson wrote:
>>> At an Asterisk CLI use the command "manager show commands".
>>
>>
>> Life is rarely that simple, and this does not really answer the question.
>>
>
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