Dear all
I have asterisk 1.2 with mediant2k i have create SIP Trunk from
asterisk 2 mediant and my PRI terminated on Mediant 2000 E1 port now everthing
is fine but problem is when i call to somebody outside and he/she disconnect my
phone but my asterisk counitine ringing my SIP
I want unified Dialing i have both Type of PBX asterisk and avaya now i want to
dial from avaya to asterik with 4 digits extention and from asterisk to avaya 4
digits extention
But thing is that i have already asterisk runing with 2 digits
extention and i dont want to change any 2 di
Hi,
I'm trying to install asterisk(v1.2.22) with FreepBX(v2.2.3) with a
4-Port FXO Sangoma card A200.
I'm using Fedora 7 (x86_64) kernel version 2.6.22.1-27.fc7, but i'm
having these errors:
$ ztcfg -
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (
On Tue, Jul 24, 2007 at 02:09:33PM -0700, bilal ghayyad wrote:
> Hi List;
>
> I need to configure a softphone to be client and use
> it with Asterisk, which is the recommended one? Is it
> iax2?
My favorite IAX2 soft phone is kiax. Though I haven't tested that many.
--
Tzafrir Co
> Message: 1
> Date: Tue, 15 May 2007 23:01:24 -0400
> From: Frank Tarczynski <[EMAIL PROTECTED]>
> Subject: [asterisk-users] Asterisk 1.4.4 reproducibly dumps core on
> Solaris 10
> To: asterisk-users@lists.digium.com
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=ISO-
Hi,
after many issues we finally managed to make our system do outgoing
calls with perfect quality.
However I cannot detect *any* form of incoming call. when I use an
outside phone to call the E1 connected to the sangoma a102, I
instantly get a fast busy tone.
My /etc/zaptel.conf is:
loadzone=us
d
iaxcomm pro??
On Tue, 24 Jul 2007 19:40:45 -0400, Jim Archer <[EMAIL PROTECTED]> wrote:
>I tried several and had very poor luck with each I tried. These included
>IaxComm, IaxComm Pro, Diax and Firefly II. Also, One other one from I
>think Germany that had just changed it's name. All of the
Hi Vicky;
Thanks a lot for your reply.
Where to download Idefisk/zoiper?
Regards
Bilal
Idefisk/zoiper softphone is for IAX2 and it works fine
almost everytime
.
However there is more variety in sip softphones . I
think zoiper is
much
better than other iax2 softphones .
O
On 7/25/07, James R. Stevens <[EMAIL PROTECTED]> wrote:
Going over the needs of any PBX that replaces our current system (working
toward Asterisk) and have VM functionality question.
Currently when someone leaves a voice mail for a sales person (Who is in
the field) the system takes the VM and
Sorry for the excessive posts but this is important. There was a period
where txfax and rxfax were not updated. During that time we had problems
with segfaulting asterisk when we used newever Asterisk, SpanDSP and/or
Zaptel versions. Steve updated those applications in June and those bugs
seem to
On 7/24/07, Sylvain Boily <[EMAIL PROTECTED]> wrote:
Hi,
Install spanDSP 0.0.2-pre26 not 0.0.3.
Le mardi 24 juillet 2007 à 17:02 -0300, [EMAIL PROTECTED] a
écrit :
> Hi Everyone...
>
> I am running Asterisk 1.2.22 on Debian "Etch". I installed it from
> sources. I have also installed tiff-v3
Thank you!
Once I upgraded sox, it stopped seg faulting and I was able to do the
conversion.
Tzafrir Cohen wrote:
> On Wed, Jul 25, 2007 at 03:29:49PM -0400, dave cantera wrote:
>> eric
>> try this...
>> sox foo.wav -r 8000 foo.gsm resample -ql
>> # add -c1 to write the file in mono
>>
>> I can
On 2007 Jul 25 (Wed) at 14:57:27 -0700 (-0700), Peter Hessler wrote:
:On 2007 Jul 25 (Wed) at 17:39:11 -0400 (-0400), Jared Smith wrote:
::On Wed, 2007-07-25 at 14:05 -0700, Peter Hessler wrote:
::> -- Executing BackGround("Zap/10-1", "enter-ext-of-person") in new stack
::> -- Playing 'ente
On Wed, 2007-07-25 at 17:25 -0400, Alejandro Acosta wrote:
> May I do call report using the pinset instead of the extension?. I
> mean, to know how many call where made using the pin code X.
You can map the pin code to the accountcode field in the CDR records.
The easiest way to do this is b
On 2007 Jul 25 (Wed) at 17:39:11 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 14:05 -0700, Peter Hessler wrote:
:> -- Executing BackGround("Zap/10-1", "enter-ext-of-person") in new stack
:> -- Playing 'enter-ext-of-person' (language 'en')
:> -- Executing WaitExten("Zap/10-1"
I thought it was the fios service but now I realize it's the snom 360!
It doesn't hang up random outgoing calls. It seems like it only happens
on outbound calls from phones that have been updated to 6.5.12 or
6.5.10. It didn't happen before, but I don't remember what version
firmware it was befor
On Wed, 2007-07-25 at 17:16 -0400, Steven wrote:
> My biggest issue with this is that the Iaxys will not generate DTMF tones
> onto the analog side..
Which type of IAXy do you have? I remember having a problem with this
over a year ago with one of the older IAXy boxes (the blue ones), but it
seem
On Wed, 2007-07-25 at 14:05 -0700, Peter Hessler wrote:
> -- Executing BackGround("Zap/10-1", "enter-ext-of-person") in new stack
> -- Playing 'enter-ext-of-person' (language 'en')
> -- Executing WaitExten("Zap/10-1", "10") in new stack
> ;; dialed '305' from the calling phone
> --
Hi All,
I think this question has been done before however I couldn't find anything
on the web.
May I do call report using the pinset instead of the extension?. I mean, to
know how many call where made using the pin code X.
Thanks in advance,
Alejandro Acosta,___
My biggest issue with this is that the Iaxys will not generate DTMF tones onto
the analog side..
I tried to use one to run an overhead paging system with, and I could not
select my zone.
I had to switch over to SIP to get a Linksys to play the DTMF.
It was still out-of-band to the Linksys, but
I'm trying to build the MySQL components in asterisk-addons but no
luck so far. I hope that you can help.
I have MySQL installed.
rpm -qa indicates:
MySQL-server-5.0.22-0
MySQL-devel-5.0.22-0
MySQL-client-5.0.22-0
rpm -ql MySQL-devel | grep client indicates:
/usr/lib/mysql/libmysqlclient.a
/usr
I'm in the process of setting up a 'phone tree', and are running into
some problems. My goal is for users to dial a phone number, the
asterisk system picks it up, plays the greeting, and users can type
whatever they want into the system.
What actually happens is users dial the phone number, as
On Wed, Jul 25, 2007 at 03:04:12PM -0500, Eric ManxPower Wieling wrote:
> Will that wrap the audio into the MS GSM container? The files will be
> e-mailed and Windows boxes don't seem to like standard GSM without silly
> additional software installed.
The .wav format (file calls it RIFF) is a c
John Novack wrote:
>
> Mike Wright wrote:
>
>>Just purchased a Motorola Wildcard X100P ...
>>but the button pressed generates no tone; on button release dialtone returns.
>>
>
> Sure sounds like polarity reversal.
>
Indeed it was. Punch block in the basement had tip and ring reversed.
Probabl
[EMAIL PROTECTED] vm]# sox -h
sox: Version 12.17.8
[EMAIL PROTECTED] vm]# sox msg.wav -g -t wav test.WAV
sox: Overriding output size to bytes for compressed data.
Segmentation fault (core dumped)
[EMAIL PROTECTED] vm]#
[EMAIL PROTECTED] vm]# sox msg.gsm -g -t wav test.WAV
Segmentation fau
I'm using asterisk 1.4.8 and Centos 5, calling over SIP channels, everything
works fine, except the sound quality sometimes not very good.
- Original Message -
From: "Garth van Sittert" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, July 25, 2007 12:53 PM
Subject: [asterisk-users] Asterisk-1.
An Asterisk recorded WAV49 file shows this:
./relocation/3425/busy.WAV: RIFF (little-endian) data, WAVE audio, GSM
6.10, mono 8000 Hz
I'm not all that familiar with exactly how to translate this into sox
output options.
dave cantera wrote:
> eric
> try this...
> sox foo.wav -r 8000 foo.gsm r
Before filing a bug report, I'd recommend you upgrade the machine as the
digum crew is sure to say it too. I remember a while back similar
"bad" wav files though way too intermittent to find a cause, though I
don't remember exactly what release, when exactly nor have I seen it in
several months.
On Wed, Jul 25, 2007 at 03:29:49PM -0400, dave cantera wrote:
> eric
> try this...
> sox foo.wav -r 8000 foo.gsm resample -ql
> # add -c1 to write the file in mono
>
> I can't remember if you have to do something special in the recording
> too depends on your recorder.. oh, now I remember.
Will that wrap the audio into the MS GSM container? The files will be
e-mailed and Windows boxes don't seem to like standard GSM without silly
additional software installed.
dave cantera wrote:
> eric
> try this...
> sox foo.wav -r 8000 foo.gsm resample -ql
> # add -c1 to write the file in mon
Hi All
Has anyone experienced a crash specific to asterisk 1.2 and Centos 5
when using the misdn hfcpci module that comes with zaptel?
I have an asterisk pack based on asterisk-1.2.17 that I have been using
on dozens of machines that are rock solid and stable. Today when I
tried moving it to
eric
try this...
sox foo.wav -r 8000 foo.gsm resample -ql
# add -c1 to write the file in mono
I can't remember if you have to do something special in the recording
too depends on your recorder.. oh, now I remember. you have set
the recording to 16bit 14400 hz or something like that... if
Hi,
Does asterisk support RFC 3313 "*Private Session Initiation Protocol (SIP)
Extensions for Media Authorization* " QoS?
Is there any terminal/SIP phone support this? Thanks!
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
aste
On 23 Jul 2007, at 15:53, Matthew Brothers wrote:
> I am playing around with IAX encryption and have had good success.
> I read somewhere, that trunked packets are not encrypted. Does
> anybody know if this means the trunk packets themselves are not
> encrypted but the voice frames in them are e
Hi BaharatSamaria;
Thanks for your kindly email.
Are (Xlite and phoner) IAX or SIP? From where I can
download them (Xlite and phoner)?
Regards
Bilal
i am using Xlite and phoner
On 7/25/07, Milo? Kocbek <[EMAIL PROTECTED]>
wrote:
> My personaly best is idefisk
>
> Regards
>
> Milos
>
> 2007/7/24
Does anyone know what options you need to use with "sox" to output the
audio in the WAV49 format that Asterisk uses.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options vi
Mike Wright wrote:
> Just purchased a Motorola Wildcard X100P
Most probably you will be disappointed.
> and installed it into a clone PC running Fedora Core 6.
Another disappointment
Not to begin a religious argument, but CentOS 4 or 5 would be a better
choice.
Search the list archives for r
I have received the follwing info from my telco.
E1, PRI, CAS, HDB3, dss1
any help?
On 7/25/07, Erick Perez <[EMAIL PROTECTED]> wrote:
> Hi,
> While I wait for my unresponsive telco to provide some assistance, can
> you provide some configuration details for the following config?
> Sangoma 102 (d
Dave writes:Sounds like there's something wrong there. I don't remember the
beginning of the conversation as to when/how if it was stated. If all
your failing messages are 60 bytes, something's definitely going on.
Have you found a pattern to it's occurrence?
My original email was as follows:
--
Mike Wright wrote:
> With PC on or off:
>
> When analog phone is off hook I can hear dialtone but cannot dial out.
> When I press any button the dialtone quiets to barely audible but the
> button pressed generates no tone; on button release dialtone returns.
>
> However the telephone does ring o
Matt wrote:
> Is it possible to make Asterisk do inband DTMF over IAX?
No. The IAX2 protocol only supports DTMF out-of-band, and no IAX2
endpoint that I am aware of will 'listen' for inband DTMF in the media
stream.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuin
Sounds like there's something wrong there. I don't remember the
beginning of the conversation as to when/how if it was stated. If all
your failing messages are 60 bytes, something's definitely going on.
Have you found a pattern to it's occurrence?
> -Original Message-
...
> The sound fil
Short Answer: No.
Long Answer: Maybe. If you can get your device to send inband DTMF and
tell Asterisk you are using INFO or RFC2833 DTMF, then Asterisk should
just pass the DTMF as audio. Then if the call goes via IAX2 it should
be inband. This is an ungly hack, should not be supported in a
HI
Kindly can anyone plz tell me what will be the broadband architecture for
Asterisk, e.g; for a multinational company having offices in different far
location. What will the best solution or architecture to setup to go over
external PSTN lines accross many locations. Is ISDN is ok or it may need
Is it possible to make Asterisk do inband DTMF over IAX?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-use
Going over the needs of any PBX that replaces our current system
(working toward Asterisk) and have VM functionality question.
Currently when someone leaves a voice mail for a sales person (Who is in
the field) the system takes the VM and then in turn dials over a POTS
line and pages the sales
On Wed, 2007-07-25 at 09:22 -0700, Jaswinder Singh wrote:
> Idefisk/zoiper softphone is for IAX2 and it works fine almost
> everytime . However there is more variety in sip softphones . I think
> zoiper is much better than other iax2 softphones .
I like firefly, it can support g729 for free and S
THANKS!!!
I was looking for 1.4.9
Very much appreciated.
-E
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-
Hi all,
Really excited to be using Asterisk and learning about VOIP and PBX's.
I'm a complete beginner at telephony but have built and installed
Asterisk 1.4.5 and read several of the Asterisk books online and have
successfully connected to FWD with IAX2 and to GIZMO using SIP.
Just purchased a
aka. the FTP site is back up
dave cantera wrote:
> ed,
> do you positively have to have 1.4.0?
> just download 1.4.9 or 1.4.8... 1.4.0 is too old...
> I can email you 1.4.8, 1.4.5, 1.4.9...
> I just downloaded 1.4.9 from:
> http://www.digium.com/elqNow/elqRedir.htm?ref=http://ftp.digium.com/pub/a
ed,
do you positively have to have 1.4.0?
just download 1.4.9 or 1.4.8... 1.4.0 is too old...
I can email you 1.4.8, 1.4.5, 1.4.9...
I just downloaded 1.4.9 from:
http://www.digium.com/elqNow/elqRedir.htm?ref=http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.9.tar.gz
daveC
EdPimentl wrote
Idefisk/zoiper softphone is for IAX2 and it works fine almost everytime .
However there is more variety in sip softphones . I think zoiper is much
better than other iax2 softphones .
On 25/07/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:
Hi All;
Thanks for all replies :) -
But that means, sof
Regarding my issue with voicemails not being played, though they have
length messages in it:
Dave suggested the following:
I've got the exact same issue lately. Check the msg.txt file for
blank lines or 2 line caller I'd info.
That's causing my issue. Haven't figured out why yet but manually
r
Hello Fellow Asterisk Mailing ListMembers,
When I tried to download the latest version of Asterisk this is what I get:
http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz
Opening fileinfo database failed
http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz
Opening fil
Damon Estep wrote:
> Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is
> probably being send before the display IE arrives. The display IE is used for
> CNAM delivery, and should not exceed 15 characters.
>
> It is very common to put a message in the display IE that indicat
Matt wrote:
> If you have been affected by the SunRocket / ALLO folding issue,
> ChiliTech would like to extend our hand to you to help you in this
> time. We will transfer your numbers to us for no cost, and will
> match your SunRocket or ALLO rate. Please contact us at
> 1-866-678-6858 x 126
satish,
please clarify...
do you want people to dial 1171 on the avaya system to get to you?
do you want people to dial 1171 on the * box to get to you?
do you want people to dial 71 on either box to get to you?
daveC
satish patel wrote:
> Dear all
>
> I have asterisk 1.2 config
On Wed, 2007-07-25 at 17:09 +0200, Josu Lazkano wrote:
> Thanks Jared, but I don't understand this.
[snip]
> I want to do that when I enter on the 110 extension.
exten => 110,1,DISA(717171,international)
When you call extension 110, you'll hear a dialtone. This is your
indication to enter the pa
Hello,
I have tried during 3 days to reinstall asterisk.
BUT in fact the problem wasn't there.
I have reinstall everything and it works !!! and I guess it has always
worked the only problem was : my patience.
In fact when I call the line I have to wait 5 rings and asterisk detects the
call:
"*CL
Does asterisk 1.2.23 solve the problem did not say in the release notes.
Also Could this be a CentOS 5 problem maybe?
I am running CentOS 5 -Asterisk 1.2.22 and Zaptel 1.2.19
Otis
___
--Bandwidth and Colocation Provided by http://www.api-digital.com
Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is
probably being send before the display IE arrives. The display IE is used for
CNAM delivery, and should not exceed 15 characters.
It is very common to put a message in the display IE that indicates that the
CNAM info will
Hello,
After a day of test, I have finally reached to the same point, same errors,
same behaviour.
So I can give you the result of zap show channels :
Chan Extension Context Language MOH Interpret
pseudofrom-pstn fr default
1from-pstn
On Wed, Jul 25, 2007 at 12:51:13PM +, karim H wrote:
> Bonjour,
> I send a new post because my problem have completely changed
> Disappointed that it didn't work, I took a silly decision. I remove zaptel
> and asterisk from my ubuntu linux.
> And I tried to install it through synaptic (apt-get
Thanks Jared, but I don't understand this.
This is part of my extensiones.conf:
[incoming]
exten => 943712666,1,Wait(2)
exten => 943712666,2,Answer()
exten => 943712666,3,Background(/home/lazkano/bienvenido)
exten => 943712666,4,Wait(1)
exten => 943712666,5,Background(/home/lazkano/extension)
ex
On Wed, 2007-07-25 at 16:46 +0200, Josu Lazkano wrote:
> Hello, I want to know if is posible to call from a mobile to my
> Asterisk (national call) and then I insert an international number in
> my mobile and Asterisk call with a Voipbuster account.
Sure! You'll want to check out the DISA() dialp
Hello, I want to know if is posible to call from a mobile to my Asterisk
(national call) and then I insert an international number in my mobile and
Asterisk call with a Voipbuster account.
Is this possible?
Thanks a lot.
___
--Bandwidth and Colocation
Sorry I did not realize that your concern was when the line is
disconnected, as I didn't bother reading the whole message and as soon
as I read that the problem is about LCR I responded not reading even
one word further about the disconnected.
On 7/25/07, Vieri <[EMAIL PROTECTED]> wrote:
>
> ---
Hi,
While I wait for my unresponsive telco to provide some assistance, can
you provide some configuration details for the following config?
Sangoma 102 (dual E1) card
Location: Panama, Central America
Telco: Cable & Wireless Panama
Lastest stable asterisk 1.2.x compiled from sources
Site A in one o
Am Dienstag, den 24.07.2007, 11:26 -0500 schrieb John Faubion:
> > To prevent further missunderstanding please do not refer the SI-120
> as a snom
> > phone. If you need support please contact snom India.
>
> Tim,
>
> If it is sold by snom India, and one is to contact snom India, I can
> certa
I would like to build into our application a button to "park" and a
button to "unpark" calls.
Consider this scenario:
Agent A gets a call, and obtains a reference number. He needs to send
this client to another Agent.
Agent A pushes the "park" button. The call is then parked into a
designated
Hi,
I have done it twice, and it still doesn't work.
The biggest problem is that i can't install asterisk using sources as I done
before. I don't know why, I have "make" problems.
I would prefer to succeed in making asterisk work installed by synaptic.
Maybe someone have an idea with the error I
Dear all
I have asterisk 1.2 configuration and it is working fine but
thing is that i have alread Avaya setup and i have intergrate my Linuxbox
asterik with Avaya system avaya already use 4 digit dialplan (1644 example )
and in asterisk i have configure 2 digit dialplan ( 44 e
You can do one thing
* completely remove asterisk by going to synaptic in GUI
* once done,you can again re install
dnt use apt-get to remove asterisk as it doesnt remove it completely...
hope it works.
On 7/25/07, karim H <[EMAIL PROTECTED]> wrote:
> Bonjour,
> I send a new post because my problem
Bonjour,
I send a new post because my problem have completely changed
Disappointed that it didn't work, I took a silly decision. I remove zaptel
and asterisk from my ubuntu linux.
And I tried to install it through synaptic (apt-get debian/ubuntu).
I successfully install zaptel. And I can see it wo
Hi Robert,
which of the distros are you using as your base, dd-wrt ,
open-wrt ?
Dave
On Tue, 2007-07-24 at 17:19 -0400, Robert Augustyn wrote:
> Hi all,
> We have put together a firmware for a range of inexpensive routers.
> It has been configured to provide optimum VoIP performance.
> We ha
Hi!
I have an Asterisk Box that has 2 E1 connections: one to the PSTN and
one to a PBX. It is acting as a telephony gateway. I have a problem: the
PSTN sends information in the Display IE (in setup, information ,
etc.messages) that the PBX needs por internal processing.
The asterisk does not rela
--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
wrote:
> You cannot detect disconnected analog lines in
> Asterisk. You can't even
> determine of the lines have dialtone. All you can
> do is determine if
> there is a current active asterisk managed call.
You're right, unfortunately. ;-(
--- C F <[EMAIL PROTECTED]> wrote:
> This should do what you want:
>
> You can call it like this:
> exten =>
12125551212,1,Macro(dialoutbound,${EXTEN},8005551212,3-Zap/G1/-Zap/G2/-Sip/Nufone/)
>
> Hope this helps.
Thanks but it still won't work when the wires are
disconnected.
This is what I g
Hi All;
Thanks for all replies :) -
But that means, softphone in Asterisk is not that
good, I see all complains. Any advise?
Please Mr. "Time Bandit": What do u mean by "my IAX2"?
Is it your code or what?
Also Mr. "Rayan": I am noticing that you are advising
for SIP, what about IAX? Nothing su
If you have been affected by the SunRocket / ALLO folding issue,
ChiliTech would like to extend our hand to you to help you in this
time. We will transfer your numbers to us for no cost, and will
match your SunRocket or ALLO rate. Please contact us at
1-866-678-6858 x 126 or e-mail [EMAIL PROTE
Ok let's forget the asterisk dialplans for a second. Maybe I am
misunderstanding the 'national' option. When I set the
dialplan=national for the zaptel config what exactly does that do? I
was under the impression that re-wrote numbers.
On 7/24/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]>
On Wed, Jul 25, 2007 at 10:51:54AM +, karim H wrote:
> Hello,
> First, thanks for your interest and your help.
> This lines are what I see when I start asterisk using: sudo asterisk -cvvv
> (I am under ubuntu feisty).
> When I call my telefonic line connected to my X100P card, there is nothing
[EMAIL PROTECTED] wrote:
> exten => 200,1,SendDTMF(200w#86)
>
> But I don't know the path to take to get that to happen after the
> hangup() - which is needed to close down the voicemail() call.
I think a better approach would be to use the externnotify option in
voicemail.conf. This lets you
i am using Xlite and phoner
On 7/25/07, Miloš Kocbek <[EMAIL PROTECTED]> wrote:
> My personaly best is idefisk
>
> Regards
>
> Milos
>
> 2007/7/24, bilal ghayyad <[EMAIL PROTECTED]>:
> > Hi List;
> >
> > I need to configure a softphone to be client and use
> > it with Asterisk, which is the recomm
Hello,
First, thanks for your interest and your help.
This lines are what I see when I start asterisk using: sudo asterisk -cvvv
(I am under ubuntu feisty).
When I call my telefonic line connected to my X100P card, there is nothing
displayed in asterisk nothing at all, the call isn't detected.
I
My personaly best is idefisk
Regards
Milos
2007/7/24, bilal ghayyad <[EMAIL PROTECTED]>:
Hi List;
I need to configure a softphone to be client and use
it with Asterisk, which is the recommended one? Is it
iax2?
Regards
Bilal
__
On Wed, Jul 25, 2007 at 09:32:43AM +, karim H wrote:
> Hello,
> I followed the advices, I add 'channel=>1' in my zapata.conf.
> But it still doesn't work, I rmmod wcfxo and rmmod zaptel. I add the module
> again with modprobe wcfxo.
> Then I launch asterisk : asterisk -cvvv. But il doesn't det
HI,
I am using Vanlink voice gateway,with asterisk.Vanlink has 4 fxo and 4
fxs ports.Both vanlink and asterisk are in my local intranet.
I am able to call from a sip phone to analog phones connected to FXS
port of Vanlink Voice gateway.and able to make calls from analog
phones to sip phones.
But i
This 2 line code is doing what I wanted.
exten => 200,1,voicemail(200)
exten => 200,2,Hangup
What I've been told is that they want the 20 year old phone system to
light up the message bulb. (yea, a filament bulb, not an LED) To do
this you pick up on the line that goes into Asterisk and do a
Hello,
I followed the advices, I add 'channel=>1' in my zapata.conf.
But it still doesn't work, I rmmod wcfxo and rmmod zaptel. I add the module
again with modprobe wcfxo.
Then I launch asterisk : asterisk -cvvv. But il doesn't detect the call.
I change of policy, i decided to use genzaptel.conf
90 matches
Mail list logo
