Thanks for your response :-). (I'm in GMT+2, and I currently have no
internet at home, so that is why my response is so late)
>On second thought--- it would be silly to make this sort of application!
>When
>would you run it?
The "feature/wish/need" is for when using both a CTI software and a IP
I'm having a tough time figuring out how to do something. If I have an
operator (which could potentially be in their own context) and an
internal-only context, is it possible to make it so the operator can
call the internal-only context but *NOT* transfer calls to it?
The idea is that the inte
It can be done - I saw a tech do it the other day.
They used the 'local' dial option, with the D option (from memory)
If you want more info, I can grab it from them next week
PaulH
On Thu, 2007-07-12 at 12:54 +, lemmel lemmel wrote:
> So I'm back on this matter (I thus give enough time
Yes, you are right.
I know that I´m trying to use a queue with a strange functionality.
Life is hard!!
I try to combine a short timeout and a short maxlimit.
Thanks a lot!!
On 7/12/07, Anthony Francis <[EMAIL PROTECTED]> wrote:
Carlos Chavez wrote:
> On Thu, 2007-07-12 at 15:03 -0300, equis
[EMAIL PROTECTED] wrote:
> Hi,
>
> I'm playing around with the QUEUE_WAITING_COUNT function but it always
> seems to return zero? I've tried everything. I suspect that this feature
> is not implemented in 1.2.7 which I am running..
>
> Does anyone know in which version this function was added?
>
I have an application that uses the Asterisk Management Interface to bridge
two calls using the Originate command with Dial as the action.
Using one SIP server, there is no ringback on the second leg of the call.
The first person is called, answers, and hears silence until the second
person picks
Hi,
is there a way to limit an account to do simultaneous calls in sip and iax?
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>They used the 'local' dial option, with the D option (from memory)
--
Documentation :
D(digits): After the called party answers, send digits as a DTMF stream,
then connect the call to the originating channel. (You can also use 'w' to
produce .5 second pauses.)
--
On Fri, 2007-07-13 at 21:58 +0800, Mark Quitoriano wrote:
> is there a way to limit an account to do simultaneous calls in sip and
> iax?
You can use the GROUP and GROUP_COUNT dialplan functions to enforce
arbitrary limits as you see fit. There's an example on the wiki at
http://www.voip-info.org
I am having trouble getting asterisk-addons 1.4.2 to compile (after a
successful configure). Asterisk itself (and AsteriskGUI) compile fine.
I get:
cdr_addon_mysql.c: In function `handle_cdr_mysql_status':
cdr_addon_mysql.c:91: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c
On 7/12/07, Jared Smith <[EMAIL PROTECTED]> wrote:
It probably wouldn't hurt to open a bug for this... I've seen something
like this before, only it was manager events ending up inside of SIP
traffic. It definitely sounds like a pointer problem or maybe a locking
problem to me... which means it
I'm using realtime sip already!
To let you understant better my problem, I'll explain a bit more:
In a redundancy scheme, I have two asterisk servers, each running on
different machines although sharing the same MySQL DB for relatime sip.
Problem arises when the second server assumes the product
There already are x86 Solaris builds for codec_g729 -
ftp.digium.com/pub/telephony/codec_g729/unsupported/
- "Bruce McAlister" <[EMAIL PROTECTED]> wrote:
> Hi All,
>
> Does anyone know what the current status is of the G729 codec on
> Solaris? According to the following link:
>
> http://www
Hi,
I'm playing around with the QUEUE_WAITING_COUNT function but it always
seems to return zero? I've tried everything. I suspect that this feature
is not implemented in 1.2.7 which I am running..
Does anyone know in which version this function was added?
Regards,
Jan
__
Hello,
I'm trying to monitor asterisk with snmp.
I'm using asterisk 1.4.4 compiled with res_snmp on a debian stable:
*CLI> module show like snmp
Module Description Use Count
res_snmp.soSNMP [Sub]Agent for Asterisk 0
I've c
Hi,
I am sure someone has already asked this, but I am fairly new to the list, so I
haven't seen anything on this. Does anyone know of a good way to leave one
voicemail message, and the message be forwarded to multiple voicemail boxes at
once. I realize that we could leave one message in a g
Lee Jenkins wrote:
> I'd say that Micro is the "MS" of Restaurant POS. We replace their
> systems regularly ;)
I'm curious what with?
--
Chris Mason
Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670
International: (305) 704-7249 Fax: (815)301-9759
Yahoo IM only: [EMAIL PROTECTED
2007/7/12, Noah Miller <[EMAIL PROTECTED]>:
> Hi -
>
> > I have a strange comportment of the MOH system on my asterisk.
> > When i respond to a call and after fews second i set this call in hold
> > mode the correspondent listen the music fine.
> > When i re-take my correspondent at T0 instant the
Hello List
I am having some problems receiving RNDIS on a EuroISDN E1 in both Asterisk 1.2
and 1.4.
Im not receiving anything, and when I do a pri debug span, I get this message:
-- Making new call for cr 114
-- Processing Q.931 Call Setup
-- Processing IE 161 (cs0, Sending Complete)
-
Can't help you with the cause but I can tell you that you can use the
"soft hangup" command to kill those channels instead of restarting.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael J.
Liberatore
Sent: Thursday, July 12, 2007 3:56 AM
To: asterisk-users@lists.digium.co
Hi,
Am Donnerstag, den 05.07.2007, 14:58 -0400 schrieb Andrew Kohlsmith:
> On Thursday 05 July 2007 2:38 pm, Doug Lytle wrote:
> > Already did that. I use ASSP for filtering. Digium and associated
> > mailing lists are white listed. There was only 1 attempt for deliver
> > and there were no del
On Fri, 2007-07-13 at 08:20 -0500, Jadrien Wauthier wrote:
> Does anyone know of a good way to leave one voicemail message, and the
> message be forwarded to multiple voicemail boxes at once.
You can pass multiple mailboxes to the VoiceMail() dialplan application
separated by ampersands, like this
On Fri, Jul 13, 2007 at 10:06:13PM +0800, Jeremy Malcolm wrote:
> I am having trouble getting asterisk-addons 1.4.2 to compile (after a
> successful configure). Asterisk itself (and AsteriskGUI) compile fine.
Please provide the following:
* Version of Asterisk you have installed
* Linux distri
On Fri, 2007-07-13 at 08:20 -0500, Jadrien Wauthier wrote:
> Hi,
>
> I am sure someone has already asked this, but I am fairly new to the
> list, so I haven't seen anything on this. Does anyone know of a good
> way to leave one voicemail message, and the message be forwarded to
> multiple voicem
Hi List;
All we know that in voice, there are a type of
communications between endpoints, for example: in some
communications we do a proxy for media and signaling
while other communications we do a proxy for only
signaling.
Where I can determine these things in Asterisk if I am
using SIP and if
Jeremy Malcolm wrote on 7/13/07 10:06 AM:
> I am having trouble getting asterisk-addons 1.4.2 to compile (after a
> successful configure). Asterisk itself (and AsteriskGUI) compile fine.
> I get:
[snipped]
Sounds like you're missing the -devel package for MySQL would be my
first guess.
--
On Fri, 2007-07-13 at 09:16 -0700, bilal ghayyad wrote:
> I have this example for Macro and I am not able to
> understand some line, if any one can help me plz :)-
[snip]
> exten => _s-.,1,Goto(s-NOANSWER,1)
>
> Also, what does it mean _s-. ? It indicated for which
> dialing number?
The undersc
Hi List;
I have this example for Macro and I am not able to
understand some line, if any one can help me plz :)-
[macro-voicemail]
exten => s,1,Dial(${ARG1},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten => s-NOANSWER,2,Goto(incoming,s,1)
exten =>
Bilal,
Asterisk is an IP PBX and thus a back-to-back user agent; by default, it
will proxy media. The only way to disengage it from the media stream is
to use signaling protocol-specific mechanisms to coax the endpoints into
talking to each other directly; in SIP, this can be done via "re-I
On Fri, 13 Jul 2007, Mark Quitoriano wrote:
> is there a way to limit an account to do simultaneous calls in sip and iax?
Among other solutions that have been proposed, you can always use global
variables or AstDB values to keep per-peer reference counts and increment
and decrement them as ca
> > > I have a strange comportment of the MOH system on my asterisk.
> > > When i respond to a call and after fews second i set this call in hold
> > > mode the correspondent listen the music fine.
> > > When i re-take my correspondent at T0 instant the music is paused. And
> > > when i re-hold him
Hi List;
Can asterisk hear (receive) calls on two IP addresses?
How?
If yes, then:
If I have a VPN router, and my Asterisk server
connected to two network cards, one has a private IP
address (192.168.0.2) connected to the VPN router
(192.168.0.1) and another network card has a private
IP address
Hi Mark -
> I'm having a tough time figuring out how to do something. If I have an
> operator (which could potentially be in their own context) and an
> internal-only context, is it possible to make it so the operator can
> call the internal-only context but *NOT* transfer calls to it?
Sort of.
2007/7/13, Noah Miller <[EMAIL PROTECTED]>:
> > > > I have a strange comportment of the MOH system on my asterisk.
> > > > When i respond to a call and after fews second i set this call in hold
> > > > mode the correspondent listen the music fine.
> > > > When i re-take my correspondent at T0 insta
At 12:20 PM 7/12/2007, you wrote:
>Ira wrote:
> > I decided to try 1.4 today. Zaptel builds, installs and seems to load
> > correctly and Asterisk builds, installs and works but no Zap lines.
> > When I typed "make menuselect" for Asterisk and went poking around I
> > discovered that chan_zap is ma
Bilal,
There is no technical difference, from Asterisk's point of view, between
bridging call legs from two different subnets that have local interfaces
versus bridging call legs from two foreign IP destinations. As long as
they are routable and reachable, they can be connected. So, I think th
On Fri, Jul 13, 2007 at 09:58:30AM -0700, Ira wrote:
> At 12:20 PM 7/12/2007, you wrote:
> >Ira wrote:
> > > I decided to try 1.4 today. Zaptel builds, installs and seems to load
> > > correctly and Asterisk builds, installs and works but no Zap lines.
> > > When I typed "make menuselect" for Aster
Noah Miller wrote:
>
> Sort of. You can create a special extension in the operator's context
> with a Goto() statement. Something like this:
>
> [operator]
> exten => 100,1,Goto(internal,prompt,1)
>
> Then in the internal context:
>
> [internal]
> exten => prompt,1,Background(who-do-you-want
Dear Jared;
Thanks for your kindly help.
But what do u mean by more characters? What that
pattern that will contain a character?
Also, what that pattern that will contain a dash (-)?
Regarding to the s-CONGESTION then what it means by
"CONGESTION" word? Why u used here "CONGESTION"?
Regards,
-
Sorry in advance if this is slightly off topic.
I'm selling a Digium TDM400P with 4 FXO ports for $300 plus $20 shipping
costs. This is an authentic Digium TDM400P. It's brand-new and was only
installed into a system once and then removed (we decided to go with T1
spans instead).
Please email me
I have the same problem as the following:
http://threebit.net/mail-archive/asterisk-users/msg36602.html
I am using a b410p (Wildcard) in TE mode on 4 channels of ISDN2e and it
totally fails to detect incoming DTMF. The newer mISDN drivers have a
parameter "dtmfthreshold" where you can tune the vo
Dear Alex;
Thanks for your kindly reply.
Please explain for me what do u mean exactly in "a la"
in the following sentence u wrote it below?
" in SIP, this can be done via
"re-INVITEs" a la the canreinvite= option for SIP
peers in sip.conf"
Another thing, do u mean that it is easier (better) if
Bilal,
On Fri, 13 Jul 2007, bilal ghayyad wrote:
Please explain for me what do u mean exactly in "a la"
in the following sentence u wrote it below?
" in SIP, this can be done via
"re-INVITEs" a la the canreinvite= option for SIP
peers in sip.conf"
It is an English colloquialism that enjoy
Ira wrote:
> So I need to build and install Zaptel 1.4 before the configure script
> will allow me to build Zap support into Asterisk?
>
> Seems a bit draconian, but whatever. Thanks for the hint, I'll try it
> and see how it goes.
Well, what would you expect? Should the configure script say "
At 01:00 PM 7/13/2007, you wrote:
>Ira wrote:
> > So I need to build and install Zaptel 1.4 before the configure script
> > will allow me to build Zap support into Asterisk?
> >
> > Seems a bit draconian, but whatever. Thanks for the hint, I'll try it
> > and see how it goes.
>
>Well, what would yo
Looks like this isn't possible. I wonder if there's a bug open on
this?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Ma
Sent: Thursday, July 12, 2007 1:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [a
On Fri, Jul 13, 2007 at 01:42:06PM -0700, Ira wrote:
> At 01:00 PM 7/13/2007, you wrote:
> >Ira wrote:
> > > So I need to build and install Zaptel 1.4 before the configure script
> > > will allow me to build Zap support into Asterisk?
> > >
> > > Seems a bit draconian, but whatever. Thanks for the
Ira wrote:
> Well, in that case, having the dependency for Zaptel indicate you
> need to build Zaptel and then run Configure would be useful instead
> of including a list of dependencies running off the edge of the
> screen that I can't read. In my case, Zaptel 1.4 was installed and
> running
Sorry for the very delayed response but wanted to add - the older I get
the more I realize that it very often has little to do with Technology.
Packaging something into an easy to understand easy to market 'product'
is more often than not the key to successful sales.
GrandCentral have won busines
After the Dial application completes, the variable ${DIALSTATUS} will
contain something like BUSY or NOANSWER or CONGESTION (CONGESTION here
means like no free phone lines or no route to destination for example)
Then, immediately after the Dial line is the Goto line
Goto(s-${DIALSTATUS},1)
th
On Fri, Jul 13, 2007 at 12:00:11PM -0500, [EMAIL PROTECTED] wrote:
> > I am having trouble getting asterisk-addons 1.4.2 to compile (after a
> > successful configure). Asterisk itself (and AsteriskGUI) compile fine.
>
> Please provide the following:
>
> * Version of Asterisk you have installed
To everyone on the list
I put a site on line the URL is
*http://bochterservices.com/phpbb/
*This is for any information on Good or Bad ITSP
You can post any problems you had with the provider
You can Vote on the provider
This is for allowing multiple viewpoints to be heard.
If a provider rece
Chris Mason (Lists) wrote:
> Lee Jenkins wrote:
>
>> I'd say that Micro is the "MS" of Restaurant POS. We replace their
>> systems regularly ;)
> I'm curious what with?
>
www.datatrakpos.com
Notice that I didn't say en masse but yes, we do replace a few Micros
systems a year. Same thing wit
Tomislav Parcina wrote:
> There is hotel application weary popular in Croatia - Micros-Fidelio.
> Now I need to connect Asterisk with this application for purpose of
> billing. Thing is that hotel would like to give customer one bill for
> every service that he used while he was in hotel.
>
> H
I was wondering if any of you guys are aware of ability to call customers by
click on customer's phone number in ACT?
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At 02:16 PM 7/13/2007, you wrote:
>What version of Zaptel do you have installed?
I discovered the issue when I rebooted the machine and the latest
version of Zap 1.4 installed and ran.
> > In 1.2 Zaptel was always built or maybe I just did
> > it right by accident the first time. I'm not stup
Hi! I am new here. Well I'm doing a call center using asterisk and I'm
looking for an open source screen pop software to pop the caller's
information, its call history and others things. i was looking around and
find the U-rang2 the problem is that it isn't open source. if someone knows
about an
This email set a record--taking 12 day to reach me. If it's something in my
control, I'd love to know what so I can fix it.
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