Hi,thanks Jean-Yves, but I've already found that page (googling), but I asked because following those instruction I couldn't find the SIP settings.Maybe are not present on my N70?Well I'll investigate*## on my mobile says:
V 2.0539.1.219-10-05RM-84Any hints?Thanks2006/8/1, Jean-Yves Avenard
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
The only thing I have noticed is that some of my posts do not make it to the
list, so I send many of my posts directly to the list.
I have the same situation right here.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Check this for a detailed description:
http://en.wikipedia.org/wiki/Berkeley_DB
Copy/paste
Berkeley DB (DB) is a high-performance, embedded database library with bindings
in C, C++, Java, Perl, Python, Tcl and many other programming
On Mon, Jul 31, 2006 at 05:24:02PM -0400, Matt Florell wrote:
On 7/31/06, Julio Arruda [EMAIL PROTECTED] wrote:
Matt Florell wrote:
Yes, that is very confusing :)
Is there no way to throw a timer chip in there(I suppose it's way too
late to put that suggestion forward now)?
Curiosity,
On Tue, Aug 01, 2006 at 08:07:01AM +0200, Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Check this for a detailed description:
http://en.wikipedia.org/wiki/Berkeley_DB
Copy/paste
Berkeley DB (DB) is a high-performance, embedded database library with
On 14:44, Mon 31 Jul 06, Tom wrote:
Any good suggestions on where to buy rack space in a country that is
not honoring stupid US patent law and has great and secure Internet
connections?
Easyspeedy (denmark)
Server4you (germany)
Those two are cheap and give you a lot of stuff.
Connection is
I have had a similar problem a few days ago, when i did a blindtransfer
i wanted to know which extension the transferer had.
i added a variable my self:
pbx_builtin_setvar_helper(chan, BLINDTRANSFERER,
transferee-cid.cid_num);
i see that this is not what YOU need, but maybe it helps to get
Is SRTP
available in asterisk? Or how
to implement it ? am using trixbox
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written
Hi
There is a problem in Asterisk 1.2.10 (at least). Even though in
theorie the source code of app_voicemail.c can be modifier to set up
the proper permission on the directories and file created for the
voicemail, this code can not work.
It doesn't take into account that the umask needs to be
Hello friends, does anyone know if there is a gui for asterisk provided with
the asterisk source or has to downloaded from somewhere else.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
try www.trixbox.orgasterisk source does not come with any GUIOn 8/1/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello friends, does anyone know if there is a gui for asterisk provided with the asterisk source or has to downloaded from somewhere else.With warm regards.Vivek J. Joshi.
[EMAIL
Trixbox is not a GUI, it's a package that includes the OS, Asterisk, a GUI, etc. FreePBX is the GUI included in Trixbox.AlexOn 8/1/06, Rajeev Natarajan
[EMAIL PROTECTED] wrote:
try www.trixbox.orgasterisk source does not come with any GUI
On 8/1/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I trying to get the softhangup option to work. I'm using the
Polycom_acd_functions branch of asterisk, so not sure if it works with this,
or I'm doing something wrong.
Below is what I have in the dial plan, using 444 and a mobile for testing,
as I would like to use this for emergency
On Monday 31 July 2006 16:32, kritikus Araklidas wrote:
Anyone know some idea if the Asterisk voicemail (WMI) can send the messages
to meridian for activate the light on meridian digital phones for voicemail
notification
Aside from using a Norstar ATA connected to an FXS port on Asterisk
If you could just post a link to your source after it is done, that would be
great.
My need would be tied to the voicemail and if I could use that instead of a
database (for the most part), I think it would be
preferred and more portable.
1. Be triggered by a script that monitors a VM folder
Are there any
problems with always having nat=yes and qualify=yes?
We just opened up
our server to be accessible to SIP from the internet. (used to require
VPN)
I had to set the SIP
setting for my test softphone to nat=yes and qualify=yes.
This makes
sense.
Some of these phone
will
true - i was meaning to say that it has a gui 'bundled' with it... (not to mention phpmyadmin, AGI to connect to high-level
application development tools such as PHP and Perl, integrated voicemail and fax-to-email support, contact
management, calling card billing and management software.
Well that made it sound like a much better system than I did ;-)AlexOn 8/1/06, Rajeev Natarajan [EMAIL PROTECTED]
wrote:true - i was meaning to say that it has a gui 'bundled' with it... (not to mention phpmyadmin, AGI to connect to high-level
application development tools such as PHP and Perl,
Please keep responses to the list, so this can help everyone.
On Tuesday 01 August 2006 09:26, you wrote:
Thak you for you response. My interconection between Asterisk (Voicemail)
and my meridian is througth PRI T1, so the only stuff that i can't activate
is the light in the meridian digital
Hi List,
I need a bit of advice please. I want to ban calls to expensive
destinations such as cell phones.
This is fairly simple here in the UK because all cell phone numbers
begin with a 7 where as all geographic numbers begin 1 and 2
Elsewhere this is different, take Andorra
Insert your patterns in a database, have a field called expensive, and query your database before making a call!On 8/1/06, Chris Blunt
[EMAIL PROTECTED] wrote:
Hi List,
I need a bit of advice please. I want to ban calls to expensive
destinations such as cell phones.
This is
Hi,
I have one fairly basic question about AddQueueMember diaplan
application, which I'm sure you guys will know to help me with:
If I add Local channel to the queue using AddQueueMember (for example:
AddQueueMember(MyQueue,Local/[EMAIL PROTECTED]) ), the newly added queue
member will have
Hi,
try to list the blocked numbers first!
Then you should be able to use wildcards without a
problem. :-)
That was the solution for the same problem at our
dialplan.
hth
Martin
- Original Message -
From:
Chris Blunt
To: asterisk-users@lists.digium.com
Using the 1.2 branch of SVN, I've been experimenting with FastAGI.
I want to do something useful for the caller (e.g. play a message)
if the FastAGI server is not running, i.e. AGI gets connect refused.
What I have found is that when AGI gets connect refused, it returns -1,
and control is passed
May be you can build an application which controls the background
terminal of the Meridian. (This would be a serial connection to the M1)
This application sends background commands like: se mw 3000.
This could be a try.
Best regards
Hans
Andrew Kohlsmith schrieb:
Please keep responses to the
Hi,
I'm one of those types who want to know what the heck is wrong when
something is wrong.
I just installed a new server (see config below) and it all works fine
for a few hours. But after 3-5 hours asterisk starts behaving VERY
strangely for no apparent reason...
1) MoH stops playing
2) Some
Tony Mountifield wrote:
Using the 1.2 branch of SVN, I've been experimenting with FastAGI.
I want to do something useful for the caller (e.g. play a message)
if the FastAGI server is not running, i.e. AGI gets connect refused.
What I have found is that when AGI gets connect refused, it returns
HI
I want to route media directly to one Caller IAX Phone
to Called IAX phone
signaling
IAX Phone1-Asterisk---IAX Phone2
and media
IAX Phone1IAX Phone2
Is it possible ?
__
Do You Yahoo!?
Tired of spam?
Actually I found one error now after a reboot..Although I don't think it has
anything to do with the strange behaviour. Could someone please tell me what
this means?
Aug 1 16:59:25 DEBUG[6771] chan_zap.c: Failed to read gains: Invalid argument
Where is the invalid argument? I've set the gains
As far as i know qualify=yes will increase you network traffic, this will make asterisk to communicate with all sip friends every X seconds, not sure the default value.On 8/1/06,
BerkHolz, Steven [EMAIL PROTECTED] wrote:
Are there any
problems with always having nat=yes and qualify=yes?
We
In article [EMAIL PROTECTED],
Rich Adamson [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
Using the 1.2 branch of SVN, I've been experimenting with FastAGI.
I want to do something useful for the caller (e.g. play a message)
if the FastAGI server is not running, i.e. AGI gets connect
Hi, I have a general Park and Announce question I can't seem to find the answer to. I keep seeing example conf files for ParkAndAnnounce but I'm fairly new to asterisk and I am not sure whether Park and Announce is a replacement for Park or a compliment. I guess my question is, how do I use it?
I'm having a problem where the very first words of the Asterisk voicemail
system prompt are distorted into a loud ear-splitting beep. When I dial my
VoiceMailMain extension I get this loud beep followed by the rest of the
initial voicemail system prompt. After that everything works fine. I've
Actually that worked perfectly, now I have another issue. I don't know the parking system too well. I'm not sure whether I should hack res_features.c to include a ast_sendtext() call to peer to send the message or if I can do it from the conf file through SendText(). the issue is whether the conf
Fadjar
I cannot offer documentation as you request.
In answer to creating a central system. This is possible but requires some
level thought and time.
You may be better choosing one of the turnkey packages available, either
OpenSource or Commercial that if well put together would achieve what
I explained it backwards,
the thing is I need to make a call right when an event happens, for
example when the second link is down, or when I receive a particular call.
In the following sample, I get a call on the first span E1 (g1), and
transfer it to the second span (g0). IF the
I'm suddenly needing a way to extend an analog phone extension about
15 miles. One end need to be a phone, SIP or analog, don't care, the
other end needs to look like an analog phone to connect to a phone
jack on the office PBX. In between the 2 ends is the Internet. I've
spent some time
On 31 Jul 2006, at 22:11, Jerry Geis wrote:
Help please. I have two systems on the net.
one in indiana and one in georgia.
connected with IAX. local SIP phones in each office (10 each) are
cisco and running sip.
TDM04B card in each location has 4 local lines.
Incoming calls to each location
I remember seeing on a website instructions on how to add controls to
hold music (volume, change classes etc.)
I've been looking in all the usual places, (voip-info, asteriskguru,
asteriskdocs etc.) and I can't find this anywhere.
Does anyone know where I can find this?
I use a provider, that allows me to use IAX tunnelling.
If I forward a call that uses G.729 and they are configured to allow
G.729 and ulaw, then ulaw will be negotiated (and the call is transcoded).
If I forward a call that uses G.729 and they are only configured to use
G.729, then (as expected)
from
http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
qualify=xxx|no|yes
where XXX is the number of milliseconds used. If yes the default timeout is used, 2 seconds.
If you turn on qualify in the configuration of a SIP device in sip.conf,
On Tuesday, August 01, 2006 9:36 AM Kai Ober wrote:
when you park a call (asterisk feature defautl keys: #700 ...) at
your isdn phone and you forgot to catch the call on another phone,
the phone from where you parked the call, should ring after 45
seconds (default)
does this work for you?
Does the accountcode from a SIP user agent get passed to IAX when trunking a
call from one asterisk box to another? The SIP caller id, extension etc do get
passsed, so why not the account code? It's a standard field.
Doing a 'iax2 debug' doesn't even show the accountcode field.
Good grief. IAX2
Yeah is true.but we have to sincronize this console command with
Asterisk SIP MWI
Regards.
Cris.
From: Johann Steinwendtner [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List -
Douglas Garstang wrote:
Does the accountcode from a SIP user agent get passed to IAX when trunking a
call from one asterisk box to another? The SIP caller id, extension etc do
get passsed, so why not the account code? It's a standard field.
Doing a 'iax2 debug' doesn't even show the
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Tue, 01 Aug 2006 15:14:51 -0300
Subject: [asterisk-users] IAX and
Accountcode
Does the accountcode from a SIP user agent get
From voicemail.conf:
; If you need to have an external program, i.e. /usr/bin/myapp
; called when a voicemail is left, delivered, or your voicemailbox
; is checked, uncomment this:
;externnotify=/usr/bin/myapp
Maybe this approach can send the commands to the M1.
Best regards
Hans
kritikus
Ira wrote:
I'm suddenly needing a way to extend an analog phone extension about 15
miles. One end need to be a phone, SIP or analog, don't care, the other
end needs to look like an analog phone to connect to a phone jack on the
office PBX. In between the 2 ends is the Internet. I've spent
FK == FaberK [EMAIL PROTECTED] writes:
FK Hi, thanks Jean-Yves, but I've already found that page (googling),
FK but I asked because following those instruction I couldn't find
FK the SIP settings. Maybe are not present on my N70? Well I'll
FK investigate *## on my mobile says: V 2.0539.1.2
Hello all!
I've searched high and low and cannot find any documentation or even
examples of the mysql addon to Asterisk being used with stored
procedures/functions in MySQL 5.0+ situations. Anyone tried it? I've been
able to do a call to a simple procedure that returns only one column in one
row,
Hey all experiencing a quirky problem:
1) call comes in on line 1 welcome too foobar
2) another call comes in on another line (line 2)
3) make transfer on line 1... while line 2 rings
3) line 2 drops after line 1 connects via transfer
--
=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J.
Yes this is what I want.
I guess the question is what is the best way to do it?
Use a Queue? or something else?
On 25 Jul 2006 13:25:45 +0200, Benny Amorsen [EMAIL PROTECTED] wrote:
J == Jones [EMAIL PROTECTED] writes:
J Hi Folks, We're migrating from a conventional KSU/PBX to Asterisk
J
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 01, 2006 8:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX and Accountcode
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL
Or let me rephrase my question:
Why is Local/[EMAIL PROTECTED] of status Unknown as you can see from this CLI
snapshot (that includes add queue member CLI instruction as well)?
What do I have to do to make it available to the callers that call in
the queue testQ:
asterisk*CLI add queue
Hi all,
Is a way to force Asterisk to send DSS1 PROGRESS message to PSTN with
indicator: Inband information now available, before call is established (even
before ALERTING phase)?
I also think that this indicator can be contained in CALL PROCEEDING message.
My idea is to play not billed
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Tue, 01 Aug 2006 17:08:15 -0300
Subject: RE: [asterisk-users] IAX and
Accountcode
What about this scenario?
User A calls
hi,
rx_fax fails to get fax on a bit noisy lines
but real fax devices can do that on the same line
with no problem!
what's the problem?
thanks
___
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asterisk-users mailing list
To UNSUBSCRIBE or
Ciao Chris,
So if I try the following dial plan my pattern always matches the
first wild card
Exten = _00376.,1,Dial(my iax terminiator)
Exten = _003763.,1,Congestion
Exten = _003764.,1,Congestion
Exten = _003765.,1,Congestion
This is a common pitfall in Asterisk dialplans: Asterisk
Dundi question:
Is there a way to pass dial arguments to switch = DUNDi as if you were
dialing using Dial(${DUNDILOOKUP(${EXTEN})},,tTwW)?
We were going to impliment DUNDi, but realized we lost the ability to
use the Dial features.
I could just use the DUNDILOOKUP function, but that keeps
On 1 Aug 2006, at 21:08, Douglas Garstang wrote:
What about this scenario?
User A calls User B. User A and User B are registered on the same
Asterisk system.
User B does an attended transfer, and transfers the call to user C,
who is registered on a different asterisk system.
You set the
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 01, 2006 10:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] IAX and Accountcode
- Original Message -
From: Douglas Garstang
Are you using mpg123 for MoH or native? What's in your musiconhold.conf?
[EMAIL PROTECTED] wrote:
Hi,
I'm one of those types who want to know what the heck is wrong when
something is wrong.
I just installed a new server (see config below) and it all works fine
for a few hours. But after
Hi,the problem is that I have not the sip choice into my N70 menu.Today I've made an update of the system, now I have:V 5..0609.2.0.1but still no sip.I think is because my mobile has been customized by my telephone company, H3G.
I'll investigate.Thanks01 Aug 2006 20:54:53 +0200, Benny Amorsen
Last night I started compiling all the components of the Unicall stack.
So far I've been able to successfully do a testcall.
A couple of questions:
1) If you download the snapshot libraries, a funcion that used to be
called dtmf_put now has been changed to dtmf_tx_put, however the
client code
-Original Message-
From: Mitch Sharp [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 01, 2006 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dundi and Dial Arguments
Dundi question:
Is there a way to pass dial arguments to switch = DUNDi as
if you
On 15:39, Tue 01 Aug 06, Douglas Garstang wrote:
-Original Message-
From: Mitch Sharp [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 01, 2006 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dundi and Dial Arguments
Dundi question:
Is there a
Hello,
What is the best utility to convert GSM files into G729 files for batch
processing.
Thanks
WAzb
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
-Original Message-
From: Michiel van Baak [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 01, 2006 3:57 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dundi and Dial Arguments
On 15:39, Tue 01 Aug 06, Douglas Garstang wrote:
-Original Message-
Hello,
The latest Polycom firmware (1.6.x series) supports HTTP(s) provisioning
that I have been trying to setup.
The admin guide mentions that in the boot settings for the configuration
server, URLs of this format can be used -
http://user:[EMAIL PROTECTED]/dir/config.cfg
But when I use that,
Ok Ok, the figure doesnt help.Here we go again - -- --- --| SIP | - | ASTERISK | -- | PANASONIC | --- | PSTN | - -- --- -- | | Ext1 Ext2Here is my dialplan[incoming]exten = s,1,Answerexten = s,2,Background(prueba-pbx)exten =
Again you are not saying how asterisk is connected to the panasonic,
stop using pictures.
On 8/1/06, Pablo Mora [EMAIL PROTECTED] wrote:
Ok Ok, the figure doesn't help.
Here we go again…
- -- --- --
| SIP | - | ASTERISK | -- |
On Tue, 2006-08-01 at 18:23 -0400, Wasif wrote:
What is the best utility to convert GSM files into G729 files for batch
processing.
I don't think sox supports G729. However, you can actually use Asterisk
to do this for you if you use the trunk, or upcoming 1.4 release. In
the trunk, there is
Hi,
I could found out why the phone received '404 Not Found'.
The reason was this part is not parsed and not Added extensions
after that.
Because there was not at least one space after ; in front of the
line of exten = 0033,1,Meetme(|qM).
Regards,
Zen
From: Zen Kato [EMAIL PROTECTED]
Subject:
Hello,
Is is possible to setup an asterisk server with out buying Digium card.
I mean can we do this type of setup.
We all know that X-Lite can be used as a soft phone to have an IP
extension.
Is it possible to take a service from another VoIP service provider, and
get the IP phone number.
On Aug 1, 2006, at 3:13 PM, Michal Doležel wrote:
Is a way to force Asterisk to send DSS1 PROGRESS message to PSTN with
indicator: Inband information now available, before call is
established (even before ALERTING phase)?
I also think that this indicator can be contained in CALL PROCEEDING
I've searched
through the newsgroup and online and haven't found an answer for my question...
maybe I am looking for the wrong terms, I am not sure...
I have a client that
would like a phone that is like a "typical" receptionists
phone.
Requirements:
- Ability for
their3 lines to
Title: RE: [asterisk-users] VOIP phone for Receptionist use
Doesn't [EMAIL PROTECTED] need the DB flag for call waiting disabled? I believe it is *70 to enable call waiting and *71 to disable.
Bill
-Original Message-
From: [EMAIL PROTECTED] on behalf of Jeff Busch
Sent: Tue
(Andrew Kohlsmith) wrote:
Re: MWI from Asterisk to Meridian
So, as I said, you are stuck using a Nortel ATA and an FXS port on Asterisk
and using a hookflash *1 sequence to toggle it. Unfortunately the VM
callback # will be the ATA's DN, so only one person at a time can access
voicemail.
Hey all! For the past year I have been working on and off on an SS7
implementation here at Digium called libss7. I have it to the point
where it can pass phone calls, so I figured it would be a good time to
release it and let people begin testing it. It's still somewhat bare
bones in
On 8/1/06, Barzilai [EMAIL PROTECTED] wrote:
Last night I started compiling all the components of the Unicall stack.
So far I've been able to successfully do a testcall.
Congratulations! :)
1) If you download the snapshot libraries, a funcion that used to be
called dtmf_put now has been
Pablo, according to description I assume that you have an FXO at *
connected to an FXS port at Panasonic. If this is correct, could you
replace Asterisk by a telephone and see if it is possible to make call
to Ext1?
Jorge
Pablo Mora wrote:
/Ok Ok, the figure doesn’t help./
/ /
/Here we go
By default, CW is turned off in AAH. You need to turn it on. I use
the 301, 500, 501, 600, and 601. CW works w/ AAH and Trixbox.
You should visit http://www.trixbox.org/index.php if you are using AAH
or Trixbox.
On 8/1/06, Jeff Busch [EMAIL PROTECTED] wrote:
I've searched through the
Barzilai wrote:
Last night I started compiling all the components of the Unicall stack.
So far I've been able to successfully do a testcall.
A couple of questions:
1) If you download the snapshot libraries, a funcion that used to be
called dtmf_put now has been changed to dtmf_tx_put,
Ok,
Im going to stop pictures
I have a Digium 4 FXO Card in my asterisk, and
connect to Panasonic through 2 extensions (configured in a pool)
This means when you dial 200 (example) in Panasonic,
the call goes to asterisk and it answers.
In this sense, the answer is yes
How does the Meridian turn on the MWI? does it use simple DTMF?
On 7/31/06, kritikus Araklidas [EMAIL PROTECTED] wrote:
Hi everyone:
Anyone know some idea if the Asterisk voicemail (WMI) can send the messages
to meridian for activate the light on meridian digital phones for voicemail
Caros,
I installed the A2Billing - v1.2.2 with Asterisk 1.2.10. All works ok, but when
I try callout got a message saying the number in not available.
Can you help with a step-by-step to make a card autenticate and dial a number?
Thank you
Luc Moreira
Mais VoIP
-- Accepting
You need to add a ww or 2 like this:
exten = 101,1,Dial(Zap/g1/ww${EXTEN})
or like this:
exten = 9,1,Dial(Zap/g1/ww9)
Hope this helps.
On 8/1/06, Pablo Mora [EMAIL PROTECTED] wrote:
Ok,
I'm going to stop pictures
I have a Digium 4 FXO Card in my asterisk, and connect to Panasonic
Douglas--
Just to let you know--
- Douglas Garstang dgarstang at oneeighty.com wrote:
context new_pbx_betty_start {
_X. = {
for (x=0; ${x} 3; x=${x} + 1) {
Verbose(x is ${x} !);
}
};
}
Here's the output.
The var x
Thank you Steve.
About the configs in Asterisk... I confess that I'm new to the code so I
still need to read more. I didn't know about ast_config()
About the hardcodedness of the countries... that seems to be the
problem. Everything is too oriented to my country works like this
with this
I have been trying to test out softhangup(). Every time I use it in a macro,
it doesn't seem to hang up any call/s on the trunk. I have used:
exten = s,1,SoftHangup(SIP/trunk-sx)
exten = s,1,SoftHangup(SIP/trunk-sx|a)
exten = s,1,SoftHangup(SIP/trunk-sx-1)
exten =
Koopmann, Jan-Peter wrote:
On Friday, July 28, 2006 3:12 PM Kai Ober wrote:
What about DIAL ( |M(macro-name))
and set the userfield in cdr during execution, ...
Set the userfield to what? That is the entire problem. ${CHANNEL} will give me
something like Zap/10-1. ${BRIDGEPEER} is empty. I
Hi,
Would just like to ask, I have an SER SIP Proxy and I setup an Asterisk,
i used an SER local as a trunk for the Asterisk.
When the Asterisk box register to SER it will have this URI
sip:[EMAIL PROTECTED], instead of sip:[EMAIL PROTECTED]
Anyone has encountered this problem? Because I'm
Hi
Last week i was working on the same
i had same problem
later after struggling lot, i have found solution by trying some options
iam able to succeeded for the same
may be this config should help you
sip_additional.cfg
register=account:[EMAIL PROTECTED]/account
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