Hi
My call flow is
T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN
Call is placed in reverse direction - from PSTN to T38 Gateway.
T38 GW reinvites with T38, and asterisk passes it along to the SIP provider.
The SIP provider challenges it and asterisk reponds to the Challenge w
2009/10/5 Jonathan Thurman
> Don't use them for Fax... I spent too much time trying to use one for
> a faxing ATA. (We went with the AudioCodes MP-124 instead, which
> rocks). We to have some analog phones and an analog IVR system hooked
> up to one with no issues. They are easy to configure
Hi Matt,
Thanks so much for your help. I tried lot of ways to trouble shoot the
issue, but finally I figured out that it was from the carrier side that they
had set the limit of 150. Till now I under the impression that they provide
just the bandwidth for the trunk, but they have the ability to li
On Mon, Oct 5, 2009 at 6:31 PM, Danny Nicholas wrote:
> $595 US. Cheap, but depends on the price of local dirt.
>
>
LOL... dirt in Argentina is cheaper.
--
Nicolás Gudiño
Buenos Aires - Argentina
___
-- Bandwidth and Colocation Provided by http://www.
B.Masoud @ SH wrote:
> Assume I have a Main Asterisk Server located in UK, and another box that
> have PSTN interfaces located in China, now the purpose is to FW calls
> through PSTN.
>
> Assuming I have a client who is calling from Japan to my main switch in UK
> and he is calling China, (japan ha
Hello,
I would like to know how Asterisk deal in this case:
Assume I have a Main Asterisk Server located in UK, and another box that
have PSTN interfaces located in China, now the purpose is to FW calls
through PSTN.
Assuming I have a client who is calling from Japan to my main switch in U
Angus Asterisk schrieb:
> It seems that the zaptel startup script does not work. I noticed at startup
> the line:
> /etc/init.d/zaptel: line 40: /etc/init.d/functions: No such file or
> directory
>
> Line 40:
> # Source function library.
> if [ $system = redhat ]; then
> . $initdir/functions
$595 US. Cheap, but depends on the price of local dirt.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of troxlinux
Sent: Monday, October 05, 2009 4:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussi
2009/10/5 CunningPike :
>
> I can add a recommendation for iSymphony - cheaper than dirt, easy to
> configure, and the users like it.
>
> CP
>
Hi , but this is free?
regardss
--
rickygm
http://gnuforever.homelinux.com
___
-- Bandwidth and Colocation
On Mon, Oct 5, 2009 at 11:34 AM, Ken D'Ambrosio wrote:
> Hey, all. Just wondering if there's a GUI out there -- preferably OSS,
> but I'll take what-have-you -- that
> a) can run on an Ubuntu/Debian box, and
> b) allows a receptionist to see what calls are in-process, and forward
> calls from th
Suse 11.1 for some reason won't install on the VIA box. After installing
get garbled text on screen.
I want to fix this as a learning experience.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Core show channeltypes:
SIP Session Initiation Protocol (SIP)yes yes
yes
Console OSS Console Channel Driver no yes
no
OOH323 Objective Systems H323 Channel Driverno yes
no
Skinny Skinny Client Control Protocol (Skinny) no
On Monday 05 October 2009 12:33:47 Danny Nicholas wrote:
> What are the limitations of ActionID? In all of the examples I see, it is
> usually 1 or some integer. Can it be a timestamp like uniqueid?
I use AMI as part of an external bash application and I usually specify the
ActionID to the some
There are plenty of good products out there, but I use my own
PERL/Apache/AMI interface for this
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Baker
Sent: Monday, October 05, 2009 2:10 PM
To: Asterisk Users Mailing L
We use iSymphony Asterisk Operator Panel with a great deal of success.
http://www.i9technologies.com/index.php?option=com_content&task=view&id=19&Itemid=40
Jason Baker
IT
Coordinator
Glastender, Inc.
5400 North Michigan Road
Saginaw,
Michigan 48604 USA
Phone: 989.752.4275 ext.
228
Fax: 989.
B.Masoud @ SH schrieb:
> I have defined the card g0 to have 24 channels, but
> every time I try to call, if all ports are off the call always go to the
> first port, how can I balance the calls over all ports???
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels#DialingaGroup
Dial(Dahdi/r0
You don't need to run make clean the 1.4.26.2 folder. Just do
./configure & make install in the 1.4.25 folder.
When you run make you are just compiling the source into binaries in
that folder. You can have a number of these source folders and they
won't conflict. Make install is what actually copi
Thanks,
I made the zone, and now call disconnect works ok!
i have one last problem, I have defined the card g0 to have 24 channels, but
every time I try to call, if all ports are off the call always go to the
first port, how can I balance the calls over all ports???
Any suggestions appreciated.
On Mon, Oct 5, 2009 at 2:34 PM, Ken D'Ambrosio wrote:
> Hey, all. Just wondering if there's a GUI out there -- preferably OSS,
> but I'll take what-have-you -- that
> a) can run on an Ubuntu/Debian box, and
> b) allows a receptionist to see what calls are in-process, and forward
> calls from the
On Mon, 2009-10-05 at 12:33 -0500, Danny Nicholas wrote:
> What are the limitations of ActionID? In all of the examples I see, it is
> usually 1 or some integer. Can it be a timestamp like uniqueid?
It is simply a unique string. You can make it a timestamp if you'd
like, but I doubt that means
Hey, all. Just wondering if there's a GUI out there -- preferably OSS,
but I'll take what-have-you -- that
a) can run on an Ubuntu/Debian box, and
b) allows a receptionist to see what calls are in-process, and forward
calls from their phone to somewhere else.
Thanks!
-Ken
--
This message has
Danny Nicholas schrieb:
> What are the limitations of ActionID? In all of the examples I see, it is
> usually 1 or some integer. Can it be a timestamp like uniqueid?
AFAICR ActionID is a string. Probably limited to 255 characters or
something.
> -Original Message-
> From: asterisk-users
What are the limitations of ActionID? In all of the examples I see, it is
usually 1 or some integer. Can it be a timestamp like uniqueid?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith
Sent: Monda
On Mon, 2009-10-05 at 14:55 +, Anahi Ludueña wrote:
> I'm executing some parallel Originate async, is there a way to know
> the result of each originate?...
> I was looking at the OriginateResponse event, but I don't know how to
> get it from my web service. Also, if I have 3 originate in paral
Each is independent of the other. The important things are to make sure
asterisk is not running when doing make install and to clean
/usr/lib/asterisk/modules before make install.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behal
I currently have asterisk-1.4.26.2 installed and working. It was sugguested I
try asterisk-1.4.25 to see if it fixes my SIP dtmf problems.
What is the method to downgrade?
Do I just do in the asterisk-1.4.25 folder:
make clean
./configure
make install
Or do I need to 'make clean' in the aster
At 02:47 AM 10/5/2009, you wrote:
TDM04. The original 4 channel card with 4 red cards installed.
>Are you series???
>My card is FXO TDM2400, I am sure its designed to forward calls to pstn!!!
>
>
>
>At 04:32 PM 10/4/2009, you wrote:
> >Hi
> >I installed TDM24 card, made ZAP (DAHDI) trunk, and set
Anyone working on this? Would love to have a "click to talk" that would
operate with my Grandstream video phones.
j
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register N
On Mon, 5 Oct 2009, James Lamanna wrote:
> Hi,
> I noticed that Dahdi starting producing these error messages:
>
> ERROR[29250] chan_dahdi.c: No more room in scheduler
> ERROR[29250] chan_dahdi.c: Asked to delete sched id -1???
>
> during which time I could not send any calls or receive calls on
You would have to be able to query an AMI interface for results (PHP, PERL,
etc)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Monday, October 05, 2009 10:46 AM
To: asterisk-users@lists.digium.com
Subje
Doug,
I have tested both ends and got the same results. I was able to using FOP to
drag to Conferences, just not the Parking Lot. Another strange thing I found
is this:
- On ext 5134, I call ext 5334
- 5334 picks up the call
- using FOP, I drag 5334 and drop it back to 5334.
- 5334 gets disco
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
James Lamanna wrote:
> This is with dahdi 2.2.0 and asterisk 1.6.0.10.
>
> Any ideas on this issue?
Check to see if this is a bug that has been fixed in > 1.6.0.10. I
think the current is 1.6.0.15 and there has been significant bug fixes
since your
Don't use them for Fax... I spent too much time trying to use one for
a faxing ATA. (We went with the AudioCodes MP-124 instead, which
rocks). We to have some analog phones and an analog IVR system hooked
up to one with no issues. They are easy to configure if you just need
to hook up some anal
Thanks Danny, but how can I get it from my web service?
Anahi Ludueña
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 5 Oct 2009 10:03:41 -0500
Subject: Re: [asterisk-users] OriginateResponse Event
Each response set has a uniqueid field
that des
Hi,
I noticed that Dahdi starting producing these error messages:
ERROR[29250] chan_dahdi.c: No more room in scheduler
ERROR[29250] chan_dahdi.c: Asked to delete sched id -1???
during which time I could not send any calls or receive calls on at
least one of my Dahdi spans.
The only way to clear t
You are playing the prompt with Background or Playback? Please post the
dialplan snippet.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bart Fisher
Sent: Monday, October 05, 2009 10:03 AM
To: asterisk-users@lists.digium.c
Hello,
I'm looking for info about interconnecting asterisk to QSIG GF enabled PABX over
PRI .
Any information and pointers will be helpful.
The very first first question: does asterisk support QSIG BC and GF natively
i see that it is supported through CAPI enabled cards but what about support
th
I have a simple dialplan. Using the read cmd, I ask caller for his passcode.
If the caller is calling from an plain old analog phone, his dtmf is not heard
until the prompt stops playing. SIP phones work correctly. I've trird
everything I found searching the net. I've tried all the dtmfmode. I
Philipp Kempgen schrieb:
> Klaus Darilion schrieb:
>> forgot to mention this happens on Asterisk 1.4.26.1
>>
>> Klaus Darilion schrieb:
>>> Hi! I have a problem with "jump" in AEL:
>>>
>>> _+43123456789! => jump +22;
>>> +22 => { NoOp(); }
>>>
>>> -> OK
>>>
>>> _+43123456789! =>
Each response set has a uniqueid field that designates the start time and
call sequence of the call. Unless you manage to start 36K calls
simultaneously, you can track each call with this.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Steve Edwards schrieb:
> On Mon, 5 Oct 2009, Klaus Darilion wrote:
>
>> forgot to mention this happens on Asterisk 1.4.26.1
>>
>> Klaus Darilion schrieb:
>>> Hi! I have a problem with "jump" in AEL:
>>>
>>> _+43123456789! => jump +22;
>>> +22 => { NoOp(); }
>
> Don't you need anothe
Hi people,
I'm executing some parallel Originate async, is there a way to know the result
of each originate?...
I was looking at the OriginateResponse event, but I don't know how to get it
from my web service. Also, if I have 3 originate in parallel, how can I
identify the OriginateResponse of
I want to know what dahdi_dynamic and dahdi_transcode modules are for.
What are they purpose?.
I have read in this thread:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg180595.html
That dahdi_transcode is for the TC400B transcoder card. But this does not
seems to be true, because
>
> I cant find Zapata.cfg
You have a DAHDI installation thus you have to find chan_dahdi.conf.
it should be located under /etc/asterisk
Regarding the configuration for tones you have to check indications.conf file
Best regards,
Nini
___
-- Bandwidth
Scott L. Lykens wrote:
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>> boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
>
>> I am working on getting this situation resolved and should have new
>> releases of FFA out at the end of thi
Hello,
My configuration is :
Card 0 - kernel dummy sound card
Card 1 - my soundcard
I have a jackd running in background. My jackd launch command is :
jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0
--capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2
--outchanne
Klaus Darilion schrieb:
> forgot to mention this happens on Asterisk 1.4.26.1
>
> Klaus Darilion schrieb:
>> Hi! I have a problem with "jump" in AEL:
>>
>> _+43123456789! => jump +22;
>> +22 => { NoOp(); }
>>
>> -> OK
>>
>> _+43123456789! => jump 22;
>> 22 => { NoOp(); }
On Mon, 5 Oct 2009, Klaus Darilion wrote:
> forgot to mention this happens on Asterisk 1.4.26.1
>
> Klaus Darilion schrieb:
>> Hi! I have a problem with "jump" in AEL:
>>
>> _+43123456789! => jump +22;
>> +22 => { NoOp(); }
Don't you need another semi-colon after the closing brace?
S
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
> I am working on getting this situation resolved and should have new
> releases of FFA out at the end of this week, but in the meantime if
Danny Nicholas schrieb:
> Sipregisterattempts would seem to be the simplest way to do this. It is 0
> by default, changing it to 5 would stop the hacker after 5 tries.
wrong.
registerattempts wokrs the other way round - if Asterisk is the client
and registers to another SIP proxy.
regards
kl
forgot to mention this happens on Asterisk 1.4.26.1
Klaus Darilion schrieb:
> Hi! I have a problem with "jump" in AEL:
>
> _+43123456789! => jump +22;
> +22 => { NoOp(); }
>
> -> OK
>
> _+43123456789! => jump 22;
> 22 => { NoOp(); }
>
> -> OK
>
> _+43123456789! =
Hello,
I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.
isb177*CLI> features show
Builtin Feature Default Current
--- --- ---
Pi
Man, thanks a lot!
I just changed the name to g0 instead of DGTDM24 and it worked!!
I would like to know where I can set the configuration for line tones( dial
tone, call and busy tone) and where I can change different setting for
polarity / current disconnect etc.. of the line?
I cant find Zapat
>> DAHDI/DGTDM24/966505103250
This (DGTDM24) is strange. Could you provide the setup of the DAHDI trunk?
You should have something like DAHDI/g0/96 or DAHDI/10/96
Here are more info on this subject:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg226642.html
HTH,
Ioan (Nin
Are you series???
My card is FXO TDM2400, I am sure its designed to forward calls to pstn!!!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira
Sent: Monday, October 05, 2009 5:07 AM
To: Asterisk Users Mailing
hin lee wrote:
> Whenever I try to drag calls to the Parking Lot or On Hold, FOP would
> drop my calls. I have searched online and have found similar problem,
> such as the link below. I have tried their solution but still the FOP
> is not working correctly. I even installed the HUDLite server
fba098", "Using
CallerID "100" <100>") in new stack
-- Executing [966505103...@from-internal:2] Set("SIP/100-08fba098",
"_NODEST=") in new stack
-- Executing [966505103...@from-internal:3] Macro("SIP/100-08fba098",
"record-en
Hi,
In this
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/204725/focus=221375dating
from 2008, experiences with Grandstream GXW4024 were asked.
Has anyone something up-to-date to share about this ?
Regards
___
-- Bandwidth and Colocatio
Martin Stubbs wrote:
> Hi,
>
>
> If I connect to the USB modem with minicom and issue the ATDxxx; command
> with a semicolon at the end to signify a voice call I get the same error
> response.
>
> Could someone else with this type of USB modem tell me if that command should
> work in mini
Hi! I have a problem with "jump" in AEL:
_+43123456789! => jump +22;
+22 => { NoOp(); }
-> OK
_+43123456789! => jump 22;
22 => { NoOp(); }
-> OK
_+43123456789! => jump 22;
_22 => { NoOp(); }
-> OK
_+43123456789! => jump +22;
_+22 => { NoOp(); }
60 matches
Mail list logo
