Hi,
I have several parts of my dialplan implemented wirh the group function
and now I need to decrease the group counter in some special case without
hanging up the channel.
Is this psossible ? So far I didn´t find something related in the
documentation...
Any hints will be appreciated...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Barry Porch
Sent: Friday, May 11, 2007 8:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] A couple of questions for the
Mitelgurus(phone-related - not
Hi,
I have written an asterisk application app_custom.c and I want to link it
with a third party library libthirdparty.so. Is there a way to do this with
the 1.4 build system. Also does anyone have any documentation on customizing
the 1.4 build system it's a lot different from the 1.2 build
Hi,
I'm trying to install a Junghanns quadbri for a few days but i stay with an
asterisk error. (Everyone is busy/congested )
Asterisk is working with a Fritz PCbut from one year and now i want to add
the quadbri.
The quadbri card has been configured in NT mode and with no 100 ohms S/T
Am Freitag, den 11.05.2007, 18:44 -0400 schrieb Jon Pounder:
just out of curiousity - anyone ever hijack pairs and get away with it ?
(do your own cross connects on the street and utilize some crossconnect
all within one branch of F1 cable out of the CO ?)
I've been tempted in the past, and
Jon Pounder wrote:
Quoting Stephen Bosch [EMAIL PROTECTED]:
C F wrote:
Stephen i disagree. growing up in new work city i can say its quite
easy to get away with it in the city. where i live now in new jersey
(population of around 6) i wouldnt be able to pull that off.
The world is a
hi list,
I'm looking for a way to execute commands in my dialplan specifically when a
caller has hung up. my curretn dialplan looks like this:
exten = s,1,Answer
exten = s,n(restart),BackGround(intro)
exten = s,n,Read(Enter,4,4)
exten = s,n,Voicemail(${Enter},u)
exten =
Per Jessen wrote:
Perhaps something along the lines of unauthorised tampering with a
telecomms installation?
More likely conspiracy to aid terrorists by destroying the
infrastructure.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK
from extensions.conf:
exten = _X.,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I basically try to lookup the CLIP and attach a name for each inbound
call. This works fine, except when I have just restarted asterisk - at
which time I've more than once seen the message from the subject.
In article [EMAIL PROTECTED],
Michael Kamleitner [EMAIL PROTECTED] wrote:
I'm looking for a way to execute commands in my dialplan specifically when a
caller has hung up. my curretn dialplan looks like this:
exten = s,1,Answer
exten = s,n(restart),BackGround(intro)
exten =
I have a queue defined which I use like this:
exten = _X.(reception),n,Ringing
exten = _X.,n,Queue(enidan,t,,,20)
exten = _X.,n,Voicemail(443,u)
exten = _X.,n,Hangup()
When I start asterisk, this queue doesn't work -
-- Executing [EMAIL PROTECTED]:3] Queue(mISDN/3-u0, enidan|t|||20)
in new
Open source...I wish...at least not to my knowledge yet. Likely
something to do with the licensing for Skype...someone correct me here
if appropriate.
I'll drop a second email with details on the configuration unless
someone else pipes up requesting it.
D.
-Original Message-
From:
Anybody
I am still waiting.
Kapil Dhawan wrote:
Hi
Can somebody brief me the working of RTP mixer from MeetMe perspective.
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Hi,
MoH volume is uncomfortably high and I want to bring it down. Its mpg123.
How can I do it?
--
Zeeshan A Zakaria
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thx Tony, but DeadAGI doesn't seem to fit my needs... the way I understand
its functioniality (
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI), DeadAGI
is ensureing that an executed AGI-script is finished, even if the caller
hung up _during_ execution.
in my case, I need to
Hmm, I tried this, but I get the following notice:
NOTICE[27486]: pbx_dundi.c:4695 set_config: Ignoring invalid EID entry
'*'
Do you perhaps know for any other option?
Thanks, Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Bagnall
Sent:
Hi There,
Good guide on setting up chanskype on trixbox
http://www.geek-pages.com/articles/asterisk/setting_up_trixbox/asterisk_to_use_skype.html
also:
http://www.chanskype.com/
working on my trixbox 2.0 :)
Best Regards,
Com os melhores cumprimentos,
Hugo Picão
Link Consulting -
On Sat, May 12, 2007 at 10:44:19AM +0530, Kapil Dhawan wrote:
Hi
Can somebody brief me the working of RTP mixer from MeetMe perspective.
(RTP? in MeetMe? I don't really know this well but AFAIK RTP doesn't get
mixed directly in MeetMe. MeetMe is a generic Asterisk channels mixer)
Aparantly
Did you have the IP specified in sip.conf?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Yaakov Menken
Sent: May 13, 2007 10:43 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Sudden appearance of SIP/2.0 401
Asterisk 1.2.17
I am starting to have problem with one of my queue. Everytime when I try to
login an agent with AgentCallBackLogin(), it will play periodic announcement
for the queue during this function call. Also when this agent answer the call,
during the conversation, the agent also hear
Remix your wav/mp3 files with a lower volume :)
On 5/14/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Hi,
MoH volume is uncomfortably high and I want to bring it down. Its mpg123.
How can I do it?
--
Zeeshan A Zakaria
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Zeeshan,
On Mon, 14 May 2007, Zeeshan Zakaria said something to this effect:
MoH volume is uncomfortably high and I want to bring it down. Its mpg123.
How can I do it?
There are some settings in musiconhold.conf that may yield the desired
effect:
[default]
mode=mp3
We use the Handset Gateways from Citel.
They convert SIP to Digital Handsets, so there is no hardware to add to the
server and you can still use your 2-wire phone lines.
--
--
Steven
http://www.glimasoutheast.org
bilal ghayyad [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Hi;
Not now that they have intoduced number portability.
The phone companies have to keep huge databases to keep track of which carrier
to send the call to.
--
--
Steven
http://www.glimasoutheast.org
Ritesh Agrawal [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Hi Folks,
Is
Hi,
I'm running Asterisk 1.4.2 on a Debian GNU/Linux and a Digium TE210P card. I
have one E1 with MFCR2 Signaling. I compiled asterisk + libunicall, and I
can make calls over E1, but can't receive.
Here the CLI when I make a call:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/23-081cbc40,
Is there any way to play a file on a channel from the Manager API
(other than from Originate)?
This question was asked by someone else on the ast-dev list and the only
advice given was that Redirect was the solution. I find myself with the
same problem now but I don't understand the response.
The Citel Handset Gateways were the best option for our scenario.
The cost per port for the number of buttons on our NEC DTerm/E phones was about
half.
Also, no network reengineering.
We connected new 66 blocks to the Citel units. And just cutover from the old to
the new.
When you configure
Having had various issues with local vendor (begins with V). am looking to
move to all wireless. Anyone know if current vendor can refuse to port the
current land line numbers to a wireless provider?
From what I've read, the Fed's seem to say no, they cannot refuse, or impede
this.
joe a.
Friends,
I have gotten a few questions lately on the status on the Codename
Pineapple project, the project
that hopefully will produce a more stable and SIP compliant SIP stack
for Asterisk.
Due to lack of funding, it's postponed until further notice.
I have a few sponsors, but not enough
Here the problem is that it is streaming audio from the Internet and I can't
lower its volume.
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On Mon, 14 May 2007, Joe acquisto said something to this effect:
Having had various issues with local vendor (begins with V). am looking
to move to all wireless. Anyone know if current vendor can refuse to
port the current land line numbers to a wireless provider?
LNP does provide for
Quoting Joe acquisto [EMAIL PROTECTED]:
Having had various issues with local vendor (begins with V). am
looking to move to all wireless. Anyone know if current vendor can
refuse to port the current land line numbers to a wireless provider?
From what I've read, the Fed's seem to say no,
I was going to port a number here in Ohio and Verizon said it would cost
$90 to do so as they can charge what it cost them.
Bob R
Joe acquisto wrote:
Having had various issues with local vendor (begins with V). am looking to
move to all wireless. Anyone know if current vendor can refuse to
Hello,
I had noticed strange crackling sound on my phone calls going through my
zaptel device (TDM400P), so i decided to check on possible timer issue,
and found lots of issues on forums concerning the sensibility of zaptel
with IRQs, and tried about everything: moving PCI slots, noapic and
Hi Zeeshan,
On 5/13/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
I've solved this problem. It was very easy (only if I knew how to do it
before). I changed the UDP ports, i.e.
1. In sip.conf, bindport=5070
2. In my IP Phone server settings, www.myserver.com:5070
Now it seems to be working
Joe acquisto wrote:
Having had various issues with local vendor (begins with V). am looking to
move to all wireless. Anyone know if current vendor can refuse to port the current land
line numbers to a wireless provider?
From what I've read, the Fed's seem to say no, they cannot refuse, or
On May 12, 2007, at 4:11 PM, Atlanticnynex wrote:
Thanks Alex, some great ideas.
I think, however, I'm leaning towards Asterisk at this point- since
I have quite a bit of experience there, and very little with SER.
At this point, I'm wondering from a dimensioning standpoint, what
kind of
Hi,
I want to ask you if asterisk, when I use the command park(), gives me for
example a variable that contains the slot position where it parks the call
or if it only tells me (audio) in the channel this position number? In
other words, is there a way to obtain and use the value of the slot
Please provide us with your config in musiconhold.conf so I/we can see how
you are streaming. There may be a way to lower the volume, but it depends
on how you are performing the streaming.
On 5/14/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Here the problem is that it is streaming audio
Having had various issues with local vendor (begins with V). am looking to
move to all wireless. Anyone know if current vendor can refuse to port the
current land line numbers to a wireless provider?
From what I've read, the Fed's seem to say no, they cannot refuse, or
impede this.
Hi Guys,
Does anyone know if is it possible to put one channel in two different
spygroups?
Thanks! Alex
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On Mon, 14 May 2007, Daryl Jurbala said something to this effect:
That being said, I don't meant to trash Asterisk at all. It's a
fantastic feature server, and a great PBX, both of which things I use it
for very successfully.
Agreed. And, it's worth pointing out, that's what Asterisk is
When trasferring a call, how is the context determined?
When using a zap device, and the DTMF code for blind or attended transfer is
entered, does the tranfer originate at the context the zap device is set to
be in, or does it originate from where the outside call being transferred
originated in,
I think Joe's analysis is unreasonably negative regarding the landline
companies' willingness to port. The link he provides,
http://www.fcc.gov/cgb/consumerfacts/numbport.html, reflects my experience.
A couple cautions, however:
Landline companies may take two to three weeks to actually complete
François,
I too had a similar problem and found the information on this page helpful:
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
What ended up working for me was changing the UDMA to mode 2 for the hard
drive. Once I did that, this card has worked perfectly for me.
I was reading an article on RTP Mixer so started studying about the
mixing done by Asterisk in MeetMe. Read that CC should contain the no
of participants ifupto 15 and CSRC should come, but not getting any by
asterisk.
Tzafrir Cohen wrote:
On Sat, May 12, 2007 at 10:44:19AM +0530, Kapil
Hi List
I want to try Asterisk with 10 PRI on a single Xeon machine with g711.
Is it feasible.
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On Monday 14 May 2007 10:41 am, [EMAIL PROTECTED] wrote:
I want to ask you if asterisk, when I use the command park(), gives me for
example a variable that contains the slot position where it parks the call
or if it only tells me (audio) in the channel this position number? In
other words, is
When I make a regular call from my SIP phone connected to my Asterisk
server I have no issues, however when I make a call using Originate :
'Channel'=SIP/[EMAIL PROTECTED],
'Context'='mycontext',
'Exten'='899',
'Priority'=1,
'Callerid'='whatever'));
It creates a screech sound when the first
On Mon, 14 May 2007, Kapil Dhawan said something to this effect:
I want to try Asterisk with 10 PRI on a single Xeon machine with g711. Is
it feasible.
In truth, it is very unlikely.
How are you planning to pick up the PRIs, anyway? 3 quad-span T1 cards?
--
Alex Balashov [EMAIL
Here's my instructions...based off Tim Hunt's great script...needs
cleanup but the gist is hear to get someone going...you may think I'm
reboot happy as there's more than a couple here but past experience
found that reloads didn't do it...reboot seem to get things
going...probably something
Per Jessen wrote:
Jon Pounder wrote:
Quoting Stephen Bosch [EMAIL PROTECTED]:
C F wrote:
Stephen i disagree. growing up in new work city i can say its quite
easy to get away with it in the city. where i live now in new jersey
(population of around 6) i wouldnt be able to pull that
Kapil Dhawan wrote:
I was reading an article on RTP Mixer so started studying about the
mixing done by Asterisk in MeetMe. Read that CC should contain the no
of participants ifupto 15 and CSRC should come, but not getting any by
asterisk.
I'll just leave it at this so we can all move on
Alex Balashov wrote:
Zeeshan,
On Mon, 14 May 2007, Zeeshan Zakaria said something to this effect:
MoH volume is uncomfortably high and I want to bring it down. Its
mpg123. How can I do it?
There are some settings in musiconhold.conf that may yield the desired
effect:
[default]
Thanks Michael,
I've already been through all that unfortunately, and I have a SATA
drive, so no UDMA mode 2 as far as I know. I'm currently trying
everything again anyway, but i doubt it will work if nothing worked the
first time.
Anyone would know of issues with XEN or SMP (or both)
Steven wrote:
The Citel Handset Gateways were the best option for our scenario.
The cost per port for the number of buttons on our NEC DTerm/E phones
was about half.
Also, no network reengineering.
I've noticed that all the people who have good things to say about them
are using East Asian
Hello again gurus.
I have been using Asterisk with great results going on a couple of
years now.
My primary box is running asterisk 1.42 built from a tar ball on Mac
OSX 10.4.9.
I have a very odd issue that I cannot seem to nail down, which is
related to my Nokia E60 SIP phone.
I use
On May 14, 2007, at 4:53 AM, Arun Kumar wrote:
Im trying to install my TC400B trans coder card when I do:
modprobe wctc4xxp
tail -f /var/log/messages says:
May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder'
with 92 transcoders (srcs=000c, dsts=0101)
May 13
Hi, Francois:
François Delawarde wrote:
Hello,
I had noticed strange crackling sound on my phone calls going through my
zaptel device (TDM400P), so i decided to check on possible timer issue,
and found lots of issues on forums concerning the sensibility of zaptel
with IRQs, and tried about
With our current setup, we have an older avaya system which is linked
with our asterisk system via a em wink connection. When you press 2 on
the avaya network, it will jump to our asterisk box and then sends DTMF
digits. Asterisk listens for those numbers and then responses as soon as
it has a
On Mon, 14 May 2007, Stephen Bosch said something to this effect:
Is there a way to do it for voice mail messages? I have a user who has
trouble hearing the voice messages, saying they are too quiet.
From a cursory glance at the voicemail settings, I can't see a way.
The voicemail messages
Just a quick brief
I have a requirement of running 10 PRI's (300 Channels). I still have to
decide on hardware and cards. Can you suggest some. As per my
understanding it will be tough to go beyond 150.
Alex Balashov wrote:
On Mon, 14 May 2007, Kapil Dhawan said something to this effect:
I
Several people do use it for handling 50k minutes a day. (I'm one of
them).
Yes, you need to know what you are doing, and have a nice design, but it
is possible.Our code is only slightly altered. (mainly for billing
purposes).
Zoa
Daryl Jurbala wrote:
On May 12, 2007, at 4:11 PM,
Is there a way to do it for voice mail messages? I have a user who has
trouble hearing the voice messages, saying they are too quiet.
Just one user? Sounds like a user problem... however, with that said, you
can try increasing your zaptel volumes.
Perfect Josh...but if i am running an application which has a capability
of showing number or participants depending upon CC value, that doesn't
work. Secondly, Asterisk can show on CLI about current talking users
where it is maintaining talking status but not sending it down the line
to be
This belongs in the asterisk-gui mailing list.
However, I will see what I can do.
-bkruse
FYI. It is just a javascript pattern matching function, its super easy
to change.
Tom Lobato wrote:
Hi all!
Is there a way to asterisk-gui to allow underline (as such cpd_tom)
in Names? It
Stephen Bosch wrote:
Per Jessen wrote:
Jon Pounder wrote:
Quoting Stephen Bosch [EMAIL PROTECTED]:
C F wrote:
Stephen i disagree. growing up in new work city i can say its quite
easy to get away with it in the city. where i live now in new jersey
(population of
On Sun, 2007-05-13 at 20:54 +0300, Dovid B wrote:
I am actually getting DTMF over SIP when people call in to a clients system
that is running a2billing. They are using RFC2833.
If you are using a Cisco router anywhere in the loop, there is a known
bug that causes rfc2833 and inband
On Mon, 14 May 2007, Rob Schall said something to this effect:
The problem is with having a send to voicemail option. Right now, a
user can press *5053 and they will be sent directly to that user's
voicemail box, rather than their phone. But when you press 2*5053, it
appears the * is ignored or
François Delawarde wrote:
Thanks Michael,
I've already been through all that unfortunately, and I have a SATA
drive, so no UDMA mode 2 as far as I know. I'm currently trying
everything again anyway, but i doubt it will work if nothing worked the
first time.
Anyone would know of issues
Alex Balashov wrote:
On Mon, 14 May 2007, Stephen Bosch said something to this effect:
Is there a way to do it for voice mail messages? I have a user who has
trouble hearing the voice messages, saying they are too quiet.
From a cursory glance at the voicemail settings, I can't see a way.
Hi folks,
I was wondering what happened to Areski CDR viewer that came before
with Freepbx. I've noticed that the live-CD contains Areski but the
repositories don't have it. Will you fix that? or shall I install
Areski from sources?
Regards,
--
Diego Quintana a.k.a. RouterMaN
Ingeniero de las
Stephen Bosch wrote:
# cat /proc/interrupts
CPU0 CPU1
1: 1626 0Phys-irq i8042
6: 3 0Phys-irq floppy
8: 0 0Phys-irq rtc
9: 0 0Phys-irq acpi
14: 63
try using testcall with 255 as debug level and report back results in
order to be able to help you.
http://www.moythreads.com/unicall/mfcr2-asterisk-unicall-0.2-english.pdf
On 5/14/07, Joca Loco [EMAIL PROTECTED] wrote:
Hi,
I'm running Asterisk 1.4.2 on a Debian GNU/Linux and a Digium TE210P
Daryl Jurbala wrote:
There is some light IVR type usage for reporting account balances and
the like. With anything more than 80 or 90 calls on the box, the IVR
prompts start to break up. Ben through replacing hardware, more
memory, different Asterisk builds, etc.
Zoa wrote:
Several people
The situation is one of my asterisk servers is behind a NAT firewall and one
is not. Both servers have multiple IAX peers. The NAT firewall has port 4569
mapped through to the asterisk server behind. But, the natted server is
almost permanently unreachable from this non-natted server, even though,
Stephen i disagree. growing up in new work city i can say its quite
easy to get away with it in the city. where i live now in new jersey
(population of around 6) i wouldnt be able to pull that off.
The world is a big place, and I suppose there's room for all kinds.
In these parts, the
Try switching to a Sangoma card. You wont have anymore IRQ issues once you
abandon Digium hardware.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV
I have a requirement of running 10 PRI's (300 Channels). I still have to
decide on hardware and cards. Can you suggest some. As per my
understanding it will be tough to go beyond 150.
Alex Balashov wrote:
On Mon, 14 May 2007, Kapil Dhawan said something to this effect:
I want to try Asterisk
On May 14, 2007, at 1:29 PM, Zoa wrote:
Several people do use it for handling 50k minutes a day. (I'm one
of them).
Yes, you need to know what you are doing, and have a nice design,
but it is possible.Our code is only slightly altered. (mainly for
billing purposes).
That's great if
Actually now I am getting so many other weird problems. First of all, choppy
sound on the receiving end on the test server. I don't understand why all of
a sudden voice will go choppy, when bandwidth and Internet upload and
download speeds are good. On the production server, it registers but
On 14 May 2007, at 17:50, Martin Joseph wrote:
Hello again gurus.
I have been using Asterisk with great results going on a couple of
years now.
My primary box is running asterisk 1.42 built from a tar ball on
Mac OSX 10.4.9.
I have a very odd issue that I cannot seem to nail down,
Hello,
We have finally upgraded to Asterisk 1.4, however we've run into two issues
that weren't occurring before the upgrade.
Issue #1: We're an outgoing call center and need to record all calls. We use
the uniqueid field in the CDR to match with the recording, which we labeled
with
I really need help.
We have a Junghanns DuoBRI and we plugged it into an Asterisk 1.4 box. We
keep on getting the error layer 1 deactivated (F3)!
We then contacted Junghanns, who told us that there is no driver for the
card for 1.4 and that we should try 1.2.
We tried 1.2 with their driver
I really need help.
We have a Junghanns DuoBRI and we plugged it into an Asterisk 1.4 box. We
keep on getting the error layer 1 deactivated (F3)!
We then contacted Junghanns, who told us that there is no driver for the
card for 1.4 and that we should try 1.2.
We tried 1.2 with their driver
Hey all,
We're starting to see all circuits are busy and a few dropped calls.
When these happen, in the messages log, I see the following error.
May 14 14:42:13 WARNING[5604] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL or
TOK_LP or
We have a Junghanns DuoBRI and we plugged it into an Asterisk 1.4 box. We
keep on getting the error layer 1 deactivated (F3)! The card sees no ISDN
device connected to it, neither in NT or TE modes alike.
We then contacted Junghanns, who told us that there is no driver for the
card for 1.4 and
Thanks for all the input guys.
This is what I had originally expected.
Does anyone have any recommendations for other software configurations? I've
thought about using OpenSER + rtpproxy(or media proxy), but it seems that
OpenSER is not designed
to do this sort of thing and would require some
Is the queue enidan configured at all in queues.conf? and how is it
defined?
l.
In data Mon, 14 May 2007 13:56:25 +0200, Per Jessen [EMAIL PROTECTED] ha
scritto:
I have a queue defined which I use like this:
exten = _X.(reception),n,Ringing
exten = _X.,n,Queue(enidan,t,,,20)
exten =
You didn't even read the thread before replying. And for what it is
worth, we at Digium are very anxious to solve any sort of IRQ problems
that you (or others) might have.
Matthew Fredrickson
On May 14, 2007, at 1:43 PM, Salvatore Giudice wrote:
Try switching to a Sangoma card. You won’t
Actually, OpenSER is just the you will need to scale Asterisk.
We have perform a number of OpenSER to Asterisk implementation for 50k plus
users
-E
On 5/14/07, Atlanticnynex [EMAIL PROTECTED] wrote:
Thanks for all the input guys.
This is what I had originally expected.
Does anyone have any
remco, et al,
could I use dundi where I could use an area code to determine the
connecting server or dial string? just like we would use 88XXX to dial
a 3 digit extension on another server at location 88? or dial 84XXX for
a 3 digit extension on a server located at 84?...
thanks,
daveC
Hi,
We have a simple AGI script that provides some IVR functionality. It connects
to a MySQL database in order to create a call record and capture IVR data.
During the IVR process, we need to store the time the call started, so
basically we INSERT a new MySQL row with callstart = NOW(),
Sorry, just to make sure this is clear, in #2 below, when I said We would like
for the AGI script to stay running for the life of the call..., I also meant
after the call is transfered to the customer service queue. This is so because
we need to note that the call ended (update callend = NOW())
In article [EMAIL PROTECTED],
Michael Kamleitner [EMAIL PROTECTED] wrote:
thx Tony, but DeadAGI doesn't seem to fit my needs... the way I understand
its functioniality (
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI), DeadAGI
is ensureing that an executed AGI-script is
thx a lot Tony, I didn't know about using the h-extension (I'm new to
Asterisk)!
this way it works:
...
exten = s,n,Voicemail(${Enter},u)
exten = s,n,AGI(foneboxx.php|${Enter})
exten = h,1,DeadAGI(foneboxx.php|${Enter})
greetings,
michael
On 5/14/07, Tony Mountifield [EMAIL PROTECTED]
On Mon, 2007-05-14 at 14:52 -0500, Rob Schall wrote:
Hey all,
We're starting to see all circuits are busy and a few dropped calls.
When these happen, in the messages log, I see the following error.
May 14 14:42:13 WARNING[5604] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error,
Can someone tell me what is included in this distro?
Does it have voicemail, meetme, panel, and IVR?
Thanks,
Wiley E. Siler
Director of Information Technology
4110 N. Scottsdale Rd. Ste 110
Scottsdale, Arizona 85251
(480) 296.0260
(866) 320.2083 ext. 1003
mailto:[EMAIL PROTECTED]
Hi all,
Is there a way to tell which extension transferred a call in a blind
transfer?
Sorry if it's a basic question, but I haven't seen an answer.
${CALLERID(num)} still holds the outside party caller id (which it
should), but I'd like to the extension number of the extension that
How about forking the process when the AGI launches, and pass the PID back
to Asterisk in a variable. When the call ends (caught at the h), call
another AGI script to kill/stop that pid.
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From: [EMAIL PROTECTED]
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