Yuan LIU wrote:
From: Yuan LIU [EMAIL PROTECTED]
But I'm curious as to the approach others use. Is doing dialplan
coding in an AGI more efficient, or do people just do it that way
because it's easier than learning dialplan code? Or are there some
things that people think they can't do any other
Hello,
I have Elmeg IP290 phone and have problems with VM. I don't know how to
configure this IP phone, that it could call to [EMAIL PROTECTED] if I
pressed VMail button. Now if I press buttom VMail , ip phone dials:
sip:[EMAIL PROTECTED] (192.168.0.1 - Asterisk IP). So I don't
understand ,
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I upgraded to the newest 1.2 Zaptel release and this is still occurring. I
checked and the digium card is not sharing an IRQ with any other devices.
I also changed busycount=8, and set callprogress=no.
The call drops are still
Although I dont have an answer I would say to look at the defualt ports and
see if they are opend on all sides and if NAT is used that it is set
properly.
- Original Message -
From: Frederico Madeira [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sangoma A200 with echo can. has been real good for me. If you need 6 FXO's I
would go with one A200 with four fxo's and then a sip device for the other
2.
- Original Message -
From: Robert Augustyn [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, January 30, 2007
Have a look here: http://www.voip-info.org/wiki-Asterisk+variables
- Original Message -
From: Rafael Augusto
To: asterisk-users@lists.digium.com
Sent: Tuesday, January 30, 2007 8:48 PM
Subject: [asterisk-users] Record file name Agent
Hi people,
Hi All,
I'm using Asterisk 1.2.14 with Wildcard TDM400P. I need app_pickup.so
application so I can pickup channel-independent calls from any IP Phone
headset. How to compile and install only this application from bristufff
package?
Thank you in advance,
Dominik
On Tue, 30 Jan 2007 12:04:30 -0600
Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote:
While in 1.2.9 musiconhold
was playing continuous on sequential extensions after a
timeout, it is restarted for every extension in 1.2.14:
As I understand it, this is the way Native Music on Hold works.
Hi
I recently installed AsteriskNOW and I am trying to use its Call queue
feature. But after configuring the Queue whenever I place a call, no
phone in my Queue list rings.
I am not able to overcome this problem. I am using Snom360 as my
softphone.
Please help me In this regard.
With
PC When I set for Extern1/2 canreinvite=yes it works, but
PC Intern-2-Extern doesn't work because Asteisk gives out the
PC private IP-Adresses of Int1/2
Asterisk can't give out a public IP-address for Int1/2. Where
would it get one from?
Correct that it doesn't. But some
Many of you may have seen the recent announcement about Danny Windham
coming on as the new CEO of Digium. This is one of the most exciting
things to happen to Digium and to Asterisk at large. When Danny comes on
board, I will be transitioning to the role of Chief Technical Officer
(retaining
On Wednesday 31 January 2007 8:28 am, Conrad Wood wrote:
This assumes all sip phones are set to reinvite=yes.
I expect (one of) the options to dial (tTw or W) to force asterisk to
remain in the media path. This way *only* if it's int-int or ext-ext
will it send sip reinvite, right?
Yes, but
On Wed, 2007-01-31 at 08:42 -0500, Andrew Kohlsmith wrote:
On Wednesday 31 January 2007 8:28 am, Conrad Wood wrote:
This assumes all sip phones are set to reinvite=yes.
I expect (one of) the options to dial (tTw or W) to force asterisk to
remain in the media path. This way *only* if it's
Perfect. Here's a quick and hopefully doable followup question. We have
Polycom Soundpoint 501 phones. Is there a way to have a phone check 2
voicemail boxes? If we have a queue, and we want the MWI to show for say
that users's extension 1000 and the special billing vm box of 2000.
Either way,
Is your agent logged into that queue to receive the calls? You can
typically say show queues to list all queues and see who is not in
use vs unavailable. If they are all unavailable, are you getting a
successful Agent Logged In message when you log that guy in?
Rob
Manish Gupta02 wrote:
Hi
hello,
i want to make a dialplan where i can pickup calls to an extension when
there are internal and external calls.
i want to use only one prefix for pickup both situations so there is a
plan how to check if the incoming call is an internal call or an
extern???
regards rene
Please define pickup. Do you want to get parked calls are or you looking to
send all calls to a specific phone ?
- Original Message -
From: René Enskat
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, January 31, 2007 4:39 PM
Subject:
I was wondering when this would happen. A lot of successful and prospering
open source company like yours seems to do this.
Much like Google did. Once a company has grown to a point it's more
valuable to have someone focus on the business from a businessmans
perspective working with
Hello
We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application if we do not have E1
lines in the development environment?
Is there some kind of software tester to test IVR/Callcenter
applications virtually??
Why don´t you put the IVR in an extension...
and call it also from an extension of the same PBX.
On 1/31/07, fadi mujahid [EMAIL PROTECTED] wrote:
Hello
We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application if
Thanks for ur suggestion.
But the problem is that won't test the queuing of the outbound and inbound
calls of the callcenter
thanks again
On 1/31/07, Alejandro Lengua [EMAIL PROTECTED] wrote:
Why don´t you put the IVR in an extension...
and call it also from an extension of the same PBX.
On
Significant albeit insanely stupid Asstricks message:
2007-01-30 09:22:57 DEBUG[9946]: pbx.c:2300 __ast_pbx_run: Oooh, got
something to jump out with ('2')!
(Oooh how about creating errors we can figure out Digium!)
Any thoughts
What Error ? it says DEBUG
This just tell you that the user
Gordon Henderson wrote:
Just a general question on dialplan programming... I've implemented a
fairly full-featured system using dialplan code only. I've not used any
AGI for it, yet it ticks all the boxes I want it to tick (diverts,
follow-me, voicemail, dnd, outdialing restrictions, simple
What do you mean? Setup another box, make a bunch of calls (as if you were
clients) into the production box, use back to back E1 cards.
Bill
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of fadi mujahid
Sent: Wednesday, January 31, 2007
We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application if we do not have E1
lines in the development environment?
Is there some kind of software tester to test IVR/Callcenter
applications virtually??
Just use an
Hello to all,
I have a strange problem with my asterisk.
Line drops while i am in a call and without a reason.The line drops no
matter if it is a incoming or outgoing call and it happen while i am
talking on the phone (no silence detection problem).
I tried to debug the situation and the only
Lee Jenkins wrote:
Gordon Henderson wrote:
Just a general question on dialplan programming... I've implemented a
fairly full-featured system using dialplan code only. I've not used
any AGI for it, yet it ticks all the boxes I want it to tick (diverts,
follow-me, voicemail, dnd, outdialing
You can try us, http://www.voicemeup.com
TDM in most areas , others offloaded white routes to L3 mainly.
Cover most of usa , and canada.
you can ping www.voicemeup.com to get an idea on location , we are directly
on peer1,teleglobe,videotron with best quality bandwith only.
Per minute pricing
You can use a cross-over cable between Asterisk boxes to imitate the
functionality of a T/E-1.
Bill Gibbs wrote:
What do you mean? Setup another box, make a bunch of calls (as if you
were clients) into the production box, use back to back E1 cards.
Bill
That's interesting I use Voipjet cheap lines and I don't have a problem at
all.
Peter
On 1/30/07, Alejandro Lengua [EMAIL PROTECTED] wrote:
Hello, we have this problem with Trixbox 1.23
I have created an outgoing route where the 1st line
has Voipjet and the 2nd an 3rd have voipcheap accounts.
Seriously man.
I don't want to be testy here, but what part of *dynamic* didn't you
understand?
Adding a context to a flat file and reloading the server is NOT
dynamic.
And, as I explained in a previous post, realtime is not a solution I
can use for this issue because I'm updating proxy
hi every body,
i m new to this mail list, and also with asterisk IPBX,
i havr digium TE110P card, can someone till me if he has an particular
experience with this card, kind of bugs, problems...
kind regards
Younss
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Rob Schall wrote:
Perfect. Here's a quick and hopefully doable followup question. We have
Polycom Soundpoint 501 phones. Is there a way to have a phone check 2
voicemail boxes? If we have a queue, and we want the MWI to show for say
that users's extension 1000 and the special billing vm box of
Hello,
We usually use a crossover T1/E1 cable and a multi-port T1/E1 card and
call the server from itself or another Asterisk server. We have used
this method to do stress testing in VICIDIAL, which has a builtin set
of tools for stress testing outbound dialing.
MATT---
On 1/31/07, fadi
you can use a SIP based phone service to try it out
Hello
We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application if we do not have E1
lines in the development environment?
Is there some kind of
Hi,
We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application if we do not have E1
lines in the development environment?
Is there some kind of software tester to test IVR/Callcenter
applications virtually??
I
Outgoing calls to certain toll-fee (8XX) numbers fail -- we hear
ringing but the calls are never
answered. All other calls, and most toll-free numbers are not
affected. The numbers that are
affected are all travel related companies (United Airlines, American
Airlines, US Air, Starwood
Hotels,
Hello,
I'm currently facing a decision regards to the system I have to build.
Basically, I'm aiming for 2 Asterisk servers with 1 PRI line in each.
And each of them will record all calls in and out. I was wondering if
anyone had any suggestions in that regards ?
I'm currently thinking of
Hello all,
I think Lee has given me a head start, but I'm still running in a circle
(at least i'm in the lead).
The problem is with my queues. The phones go to their own voicemail
after 5 rings.
That's about the same time I allow the phone to ring before trying
another phone in the queue. Is
Lacy Moore - Aspendora wrote:
On 1/30/07, *Benko* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
Hello!
I've upgraded from 1.2.9 to 1.2.14 recently but experience an
unexpected behaviour with musiconhold: While in 1.2.9 musiconhold was
playing continuous on sequential
If you are having problems compiling NVFaxDetect (app_nv_faxdetect.c) or other
Newman Telecom applications on Asterisk 1.4, please look at Steve's comments at:
http://www.voip-info.org/wiki/view/NewmanTelOnAsterisk14
Several changes to Asterisk prevents NVFaxDetect and other apps from
yusuf wrote:
j wrote:
Greetings!
I've searched far and wide for an answer and have gotten no where, so I
was hoping one of you guys might have the answer;
Is it possible to dynamically add a context to the dialplan?
You can add extensions via the CLI, however if the context doesn't exist
I
How many simultaneous calls per account are you sending ?
On 1/31/07, Peter Halliday [EMAIL PROTECTED] wrote:
That's interesting I use Voipjet cheap lines and I don't have a problem at
all.
Peter
On 1/30/07, Alejandro Lengua [EMAIL PROTECTED] wrote:
Hello, we have this problem with
At 02:17 PM 1/30/2007, you wrote:
Yes exactly. I tried the 'add extension' command. With *and* without the
'replace' argument, if the context does not already exist the command
gives an error ;(
You could create a set of empty extensions to use and re-use as
needed. It's one of those tasks
Nelson, Clinton, Snowe REIntroduce Voice Over Internet E-911 Legislation
Bill Will Prevent Tragedies By Making Sure Calls for Help Made On
Internet-Based Telephone Service Connect to Local 911
Washington, DC - Senator Bill Nelson (D-FL); Senator Hillary Rodham
Clinton (D-NY), Co-Chair of the
In the queues that I've established, I've assigned a different number to
queue-agents than their normal extension. If their extension is 2120,
their roll-over (second extension) would be 3120 and their
queue-agent-id would be 4120. That way I can assign a different
dial-plan for 4120 that
This is a common issue with large inbound call center operations.
They like to cheat. They actually start sending prompts to the caller
without actually signalling their carrier that they have answered the
line. Typically they do not answer until a phone is ringing or you
are in a queue. I
I'm looking for some help from any Asterisk heavy who might be doing
something similar to what I'm trying to do...
Background:
I work for a research lab, testing telephony products and tools.
Historically, we used Ameritec Crescendos and Fortissimos to act as load
generators and call sinks
30 jan 2007 kl. 06.38 skrev Yuan LIU:
When Asterisk dials an IAX destination with no registration, it
very quickly comes to the conclusion that it can't make the call
-- Executing [EMAIL PROTECTED]:2] Dial(Zap/1-1, IAX2/
[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack
-- Called
Check what's going on in that branch. I also believe there's an open
issue in
the bug tracker for this, so you can see when it's ready for testing
again.
Thanks,
/Olle
31 jan 2007 kl. 07.04 skrev Tim Ferguson:
Is there any chance you could contact me or give me a website to
monitor the
Time Bandit wrote:
Significant albeit insanely stupid Asstricks message:
2007-01-30 09:22:57 DEBUG[9946]: pbx.c:2300 __ast_pbx_run: Oooh, got
something to jump out with ('2')!
(Oooh how about creating errors we can figure out Digium!)
Any thoughts
What Error ? it says DEBUG
This just tell
Thanks for this discussion! I've gotten a few ideas for better NAT
handling in chan_sip3.
The current way is implemented in so many installations, so it would
be hard to turn it
around, but in pineapple I can freely break backwards compatibility.
Let me think about it for a few days, then
Hi Rob,
put your call centre stuff in a context that is separate from all other
extensions (like internal, long distance, etc) and have it contain it's
own, dedicated dial() code.
[incoming-to-callcentre]
; Incoming calls to Call Centre arrive in this context
; IVR stuff.
;
; If Q1
Seriously? You want serious! You can't handle the serious!
I would assume that editing a file and refreshing a system by means of
a program or self intervention which causes no interruption in service
could be concidered dynamic. How does asterisk realtime handle this
thats so radically
Hi Friends,
This is Chandra from India. I have installed and configured Asterisk in our
company. I want to provide Polycom IP 501 model phones to our employees. I am
unable to find the dealer for these phones in India. Where can I buy these
phones in India? If anybody knows, please tell me the
Rob Schall wrote:
Hello all,
Lee recommended QUEUESTATUS, but that seems to return if anyone is in a
specific queue, and not if the current call came from a queue. I
probably just misunderstand how it all works. :)
Thanks all!
Rob
Hi Rob,
Remember that I am pretty new to Asterisk myself.
Hello
I was wondering if anybody has some practical experience with JastAGI to
share with me?
thanks for all the help
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That's an interesting idea. Do you know if its possible to just check
and see where the call came from. So if it came from the queue, do one
thing, otherwise, do another?
Rob
Joe Dennick wrote:
In the queues that I've established, I've assigned a different number
to queue-agents than their
This is a common issue with large inbound call center operations.
They like to cheat. They actually start sending prompts to
the caller
without actually signalling their carrier that they have
answered the
line. Typically they do not answer until a phone is ringing or you
are in
Asterisk version 1.4
Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI])
The optional AGI parameter will setup an AGI script to be executed on the
calling party's channel once they are connected to a queue member.
Ceará
Kind of - you could link that to the Local/xxx channel
also trixbox stop registering randomly on all versions..
confirmed with over 200 client accounts over here...
all using trxibox.. asterisk vanilla is ok
On 1/31/07, Alejandro Lengua [EMAIL PROTECTED] wrote:
How many simultaneous calls per account are you sending ?
On 1/31/07, Peter
Your dinner's in the oven.
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
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Lee Jenkins wrote:
Rob Schall wrote:
Hello all,
Lee recommended QUEUESTATUS, but that seems to return if anyone is in a
specific queue, and not if the current call came from a queue. I
probably just misunderstand how it all works. :)
Thanks all!
Rob
Hi Rob,
Remember that I am pretty new
I thought it was If it ain't broke fix it till it is!?
C F wrote:
Change log can help you a lot. I would stick to my grandmothers
advice, if it aint broken don't fix it.
On 1/14/07, Yuan LIU [EMAIL PROTECTED] wrote:
I don't have a particular reason to upgrade, but I'm installing a new
box,
so
At 09:20 AM 1/31/2007, you wrote:
I was once working on tracking down a particularly elusive bug in
one of our products and put a small piece of code showing a message
when testing a value that said Sh*t still doesn't work if a
certain value was true.
I guess you already know that I forgot
Anyone found a solution to this problem?
Remi
Damon Estep wrote:
I have considered opening a bug report on this, but wanted to get some
feedback and make sure I am not missing something in the way of a
simple work around. What is the scenario in which this impacts your
implementation?
On Wed, 2007-01-31 at 12:14 -0500, Lee Jenkins wrote:
yusuf wrote:
j wrote:
Greetings!
I've searched far and wide for an answer and have gotten no where, so I
was hoping one of you guys might have the answer;
Is it possible to dynamically add a context to the dialplan?
You can add
On Tue, 2007-01-30 at 20:17 -0800, Ira wrote:
At 02:17 PM 1/30/2007, you wrote:
Yes exactly. I tried the 'add extension' command. With *and* without the
'replace' argument, if the context does not already exist the command
gives an error ;(
You could create a set of empty extensions to
On Wed, 2007-01-31 at 09:28 -0900, Shane Spencer wrote:
Seriously? You want serious! You can't handle the serious!
Heh. Sorry man, it's been a bad day :(
I would assume that editing a file and refreshing a system by means of
a program or self intervention which causes no interruption in
Cool. My first attempt would have been to find out how to use
asterisk variables in the dialplan since I can set those like crazy
mad via an AGI. Then i would have cried and become horribly
demotivated.
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From:Mitch Thompson [EMAIL PROTECTED]I'm looking for some help from any Asterisk "heavy" who might be doing something similar to what I'm trying to do...Background:I work for a research lab, testing telephony products and tools.Historically, we used Ameritec Crescendos and Fortissimos to act as
lol.
On Wed, 2007-01-31 at 13:00 -0900, Shane Spencer wrote:
Cool. My first attempt would have been to find out how to use
asterisk variables in the dialplan since I can set those like crazy
mad via an AGI. Then i would have cried and become horribly
demotivated.
From:j [EMAIL PROTECTED]On Wed, 2007-01-31 at 12:14 -0500, Lee Jenkins wrote: I've searched far and wide for an answer and have gotten no where, so I was hoping one of you guys might have the answer; Is it possible to dynamically add a context to the dialplan? You can add extensions
Reload.. Reload.. Reload..!
/me ducks
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Shane Spencer wrote:
Reload.. Reload.. Reload..!
LOL.
--
Warm Regards,
Lee
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i am creating call files, and catching successfully the ones that don't
connect in a 'failed' extension. can anyone tell me how to find out the
reason for the failure (ie busy, no answer).
${DIALSTATUS} doesn't appear to get set (presumably because Dial() isn't
used) and channel_status doesn't
On 01/02/07, Yuan LIU [EMAIL PROTECTED] wrote:
What Lee suggested is to have the AGI script to actually parse, insert a new
context in extensions.conf, or deleting from it, then reload
extensions.conf. This would at least achieve what you wanted to do.
Or alternatively, to avoid complete
Rich Doughty wrote:
i am creating call files, and catching successfully the ones that don't
connect in a 'failed' extension. can anyone tell me how to find out the
reason for the failure (ie busy, no answer).
${DIALSTATUS} doesn't appear to get set (presumably because Dial() isn't
used) and
Hahaha,, I think thats a freaking SWEET suggestion :)
On 1/31/07, Andrew Furey [EMAIL PROTECTED] wrote:
On 01/02/07, Yuan LIU [EMAIL PROTECTED] wrote:
What Lee suggested is to have the AGI script to actually parse, insert a new
context in extensions.conf, or deleting from it, then reload
Jerry Jones wrote:
From asterisk, you do not hear anything other than ringing as it does
not cut the audio path through until it receives the answer from the
far end, hence the steady ringing.
So instead of Dial(Zap/g1/1800xxx,,r) just do
Dial(Zap/g1/1800xxx,,) so early audio can make
OK I need some help. Looking for comparisons for a large customer
wishing to provide voip service over a region. We are up against
Metaswitch who is claiming they can do anything Asterisk can do. I do
not have too much information on Metaswitch so am looking for any
information, preferably
When I was looking for a Java FastAGI interface for Asterisk I came
across asterisk-java first and didn't realize there was more than one
out there. It seems to work fine and I've got my first project working
with it, but I was wondering which Java FastAGI implementation is the
most popular
I used the FreePBX on Debian HowTo at
http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles
to initiate calls to my SIP carrier. They get my registration, but they
see that my call is interrupted before they can complete the connection.
My Asterisk log shows that the
Hi, folks:
Can the loop drop detection threshold (normally defined in milliseconds)
be set on the Digium TDM-400 cards? Most PBXs let you set this value.
-Stephen-
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Sheepishly, that was the magic bullet. Thanks Trevor!!
Tim
Trevor Peirce [EMAIL PROTECTED] wrote:
Jerry Jones wrote:
From asterisk, you do not hear anything other than ringing as it does
not cut the audio path through until it receives the answer from the
far end, hence the steady ringing.
I pulled a working TE405P from one box and put it in another box. I
compiled the zaptel modules, modprobe zaptel, modprobe wct4xxp, but no
lights on the card come on.
I do an lspci and the card shows up there.
I ran ztcfg -vv and got the error message Unable to open master device
Stephen Bosch wrote:
Hi, folks:
Can the loop drop detection threshold (normally defined in milliseconds)
be set on the Digium TDM-400 cards? Most PBXs let you set this value.
Good question. Anyone knows if the TDM-400 actually detect loop drops?
Leo
Is udev running?
On 1/31/07, Wayne Jensen [EMAIL PROTECTED] wrote:
I pulled a working TE405P from one box and put it in another box. I
compiled the zaptel modules, modprobe zaptel, modprobe wct4xxp, but no
lights on the card come on.
I do an lspci and the card shows up there.
I ran ztcfg -vv
yes
On 1/31/07, C F [EMAIL PROTECTED] wrote:
Is udev running?
On 1/31/07, Wayne Jensen [EMAIL PROTECTED] wrote:
I pulled a working TE405P from one box and put it in another box. I
compiled the zaptel modules, modprobe zaptel, modprobe wct4xxp, but no
lights on the card come on.
I do an
On Wed, Jan 31, 2007 at 08:40:59PM -0700, Wayne Jensen wrote:
I pulled a working TE405P from one box and put it in another box. I
compiled the zaptel modules, modprobe zaptel, modprobe wct4xxp, but no
lights on the card come on.
I do an lspci and the card shows up there.
I ran ztcfg -vv
On Wed, Jan 31, 2007 at 07:57:54PM -0700, Stephen Bosch wrote:
Hi, folks:
Can the loop drop detection threshold (normally defined in milliseconds)
be set on the Digium TDM-400 cards? Most PBXs let you set this value.
What exactly do you need it for?
On the FXO module (detecting) or on the
If you already havent seen this:
http://dir.indiamart.com/impcat/video-telephone.html
cheerz
- Ben.
Crazy Boy wrote:
Hi Friends,
This is Chandra from India. I have installed and configured Asterisk
in our company. I want to provide Polycom IP 501 model phones to our
employees. I am unable
Tzafrir, I'm assuming FXO module, since that's where one can usualy
(on other PBXs) set it.
On 1/31/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Jan 31, 2007 at 07:57:54PM -0700, Stephen Bosch wrote:
Hi, folks:
Can the loop drop detection threshold (normally defined in milliseconds)
Yuan LIU wrote:
From: /Mitch Thompson [EMAIL PROTECTED]/
I'm looking for some help from any Asterisk heavy who might be
doing something similar to what I'm trying to do...
Background:
I work for a research lab, testing telephony products and tools.
Historically, we used Ameritec Crescendos
On 1/31/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Jan 31, 2007 at 08:40:59PM -0700, Wayne Jensen wrote:
I pulled a working TE405P from one box and put it in another box. I
compiled the zaptel modules, modprobe zaptel, modprobe wct4xxp, but no
lights on the card come on.
I do an
Hi
My agents are logged into the queue through the UI but when ever I run
show queue on the console, it shows
Members
Agent 6001 (Unavailable)
Agent 6002 (Unavailable)
NO callers
I have followed all the steps in Queue creation.
And whenever I place a call at Queue, I
Hello all,
Can I send fax from PAP2 through a zap channel to PSTN? I have tried but
it is not successful.
Thank you for your help!
Lai
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Anyone know of a softphone for the Palm OS ?
Thanks.
Dovid___
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Hi all,
I am using asterisk1.2.14,realtime and I find there is a strange
case in the receiver's display. I have a dial plan to route a call
to the destination. I haven't set the callerid(num) for the caller.
In the receive ends, it's display shows asterisk when I make a call
to the receiver.
Yeah, your waittime parameter in your call file is set to 45 seconds.
db
On Wed, 2007-01-31 at 21:52 -0500, Matthew Rubenstein wrote:
I used the FreePBX on Debian HowTo at
http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles
to initiate calls to my SIP carrier. They
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