Armand A. Verstappen wrote:
I think we should have these setups listed:
- home user with 1-2 telco lines and 2-5 phones
- small office with 4-8 telco lines and 8-16 phones
- small office with a fractional E1/T1 and 12-24 phones
- medium office with full E1/T1 and 24-48 phones
- medium office with
Tilghman Lesher wrote:
On Monday 06 October 2003 05:13 pm, Carlton J. O'Riley wrote:
Are there any plans to incorporate the running of Asterisk as a
non-root user into the current CVS? There is nothing in Asterisk
that requires root access as far as I know and this would solve the
vmail.cgi
At 10:42 PM -0500 10/3/03, Steven Critchfield wrote:
On Fri, 2003-10-03 at 20:11, Masakazu Nakano wrote:
I found it. but that webite is chinese BIG-5. take care.
http://www.mpn.com.tw/index-big5-PRODUCT.html
The WiFi600 described in the above URL is the device I currently
have, and on which
On Tuesday 07 October 2003 01:23, Olle E. Johansson wrote:
Tilghman Lesher wrote:
On Monday 06 October 2003 05:13 pm, Carlton J. O'Riley wrote:
Are there any plans to incorporate the running of Asterisk as a
non-root user into the current CVS? There is nothing in Asterisk
that requires root
It is that type of mechanism that enum uses and yes it was to solve a
similar goal, but in this case you need a 'route server' type system - in
particular as this is for IP routing of PSTN end points not on an IP
network.
A discussion about this came up a while ago. I suggested something
On Sunday 05 October 2003 16:41, Robert Hajime Lanning wrote:
I sent this earlier under Editting variable contents but no-one
has responded. So, the subject is now more to the problem, instead
of the solution I was trying to implement.
ChanIsAvail returns the channel ID plus -session.
How
quote who=John Todd
On Sunday 05 October 2003 16:41, Robert Hajime Lanning wrote:
I sent this earlier under Editting variable contents but no-one
has responded. So, the subject is now more to the problem, instead
of the solution I was trying to implement.
ChanIsAvail returns the channel
If you write code in C/C++, you'd better use sqlrelay :
http://sqlrelay.sourceforge.net/
SQL Relay is a persistent database connection pooling, proxying and load
balancing system for Unix and Linux supporting ODBC, Oracle, MySQL,
mSQL, PostgreSQL, Sybase, MS SQL Server, IBM DB2, Interbase,
Jaun,
In your sip.conf try changing the [EMAIL PROTECTED] to [EMAIL PROTECTED]
My MWI works rock solid now almost instantaneously coming on and off.
Babak
Juan J. Sierralta P. wrote:
On Fri, 2003-10-03 at 14:53, Babak Pasdar wrote:
This issue was resolved by adding the @context in the
Tilghman Lesher schrieb:
I'm trying to get a Snom100 configured with H.323. Right now, the
...
Hi,
I had Snoms (100 and 200) configured with H.323 working with
asterisk-0.4.0.
Since I upgraded to asterisk-0.5.0 and I had problems with H.323
I switch to sip.
(The problem was: when I
Hi Leif
Very good Idea Everything you have wrote is right!!!
Many thanks
Dimitri
On Tuesday 07 October 2003 04:59, Leif Madsen wrote:
Hey all,
I am in the middle of creating a new user wizard which will generate all
the .conf's the new Asterisk user will require to get themselves up and
On Mon, 2003-10-06 at 23:59, Leif Madsen wrote:
Hey all,
I am in the middle of creating a new user wizard which will generate all
the .conf's the new Asterisk user will require to get themselves up and
running in Asterisk without having to touch a single configuration file.
This is
Title: Message
has
anyone got a voicetronics openline4 card working ??
If so
do you have any notes etc.
thanks
in advance.
Regards Mick
Title: Leterhead
Hi
I am a newbie and just set up my first Asterisk box.
I have got 2 x Grandstream 101s working as extensions and am now
looking to get to the outside world.
Q.) Can you use a voice/fax modem as an FXO interface?
If yes, then how would I configure it.
Example: 456-7000 is your main number and you have 7001 to 7099 as DIDs:
exten = 7000,1,Goto(AutoAttendant|s|1)
exten = _7XXX,1,Macro(yourdialmacro|${EXTEN})
How are you dropping the 456 there? I thought extensions picked up what
either the SIP phone had dialled, or what DTMF detection
Title: Message
got
the voicetronics openline4 card sort of working
I keep
getting this error
-- Event
[7=[03] Record fifo overflow] on vpb/1-4
and
the auto attendant is not clear.
thanks
in advance
Regards Mick
The number of digits that your telco sends to you is a configurable figure
(at least it is here in Aus). The example assumes that the telco is sending
you the last 4 digits.
Paul
Example: 456-7000 is your main number and you have 7001 to 7099 as DIDs:
exten = 7000,1,Goto(AutoAttendant|s|1)
On Tue, Oct 07, 2003 at 04:40:36AM -0500, Steven Critchfield wrote:
What point do you feel that a user is too advanced to us your wizard, or
at what point do you think a user of your wizard will be more pissed at
being hindered by the product than helped?
I'm not trying to insult you, or
What about the following device: USB FXS device (S100U)?
This sounds like a nice thing. What language are you using?
I know this would be adding a burden to coding, but can you also make the app be data
driven so any future additions or new hardware would just be added to a a text file?
(I
The number of digits that your telco sends to you is a configurable
figure (at least it is here in Aus). The example assumes that the telco
is sending you the last 4 digits.
Hmmm ok so DIDs are not what is stuffed into the CNID field? Or rather
pieces of the DID make it into the CNID?
Is it possible to send an external hookflash
command to the Digium FXO card from the asterisk PBX?
This is a great idea. I have a good understanding of Asterisk but would use
this for initial setup if it were quick and easy, then go in and tweak the
settings.
This is especially good for a client that would like total ownership and
admin over the product but do not have the time or desire to
Why compress all your prompts to .gsm files? Isn't * going to have to
reformat them anyway based on the codec being used for the call? I have all
my voice prompts as 8khz/16bit .wav files (* can't seem to play back 8 bit
files). I recorded them through soundforge as a 48Khz/16bit mono .wav -
Hi all,
I'm about to build a basic browser based call management module with some
basic functions and was wondering if I can use AGI in the following way:
browser app -- calls perl script containing AGI stuff -- controls
asterisk.
The most important task I'm hoping to integrate is call transfer
I think that app_flash should do the trick :
astro*CLI
-= Info about application 'Flash' =-
[Synopsis]:
Flashes a Zap Trunk
[Description]:
Flash(): Sends a flash on a zap trunk. This is only a hack for
people who want to perform transfers and such via AGI and is generally
quite useless
Hi,
I would like to connect my * with another PBX.
Which card should I use ?? X100P, TDM400P, etc... Which signalling ???
Does Anyone knows any tutorial or samples config on web that focus this
problem...
Thanks In Advanced
Andre Lomonaco
On Tue, 2003-10-07 at 08:37, Dave Wilson wrote:
Hi all,
I'm about to build a basic browser based call management module with some
basic functions and was wondering if I can use AGI in the following way:
browser app -- calls perl script containing AGI stuff -- controls
asterisk.
agi is
On Tue, 2003-10-07 at 07:53, Andrew Kohlsmith wrote:
The number of digits that your telco sends to you is a configurable
figure (at least it is here in Aus). The example assumes that the telco
is sending you the last 4 digits.
Hmmm ok so DIDs are not what is stuffed into the CNID field?
I think someone here has perl manager libraries available. Use google.
--
Thanks Steven. Plenty of samples from google.
Dave
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On Mon, 2003-10-06 at 23:05, Brian West wrote:
use
mailbox=500
instead of [EMAIL PROTECTED]
[EMAIL PROTECTED]
Thanks guys, changing voicemail by mailbox did the trick !
--
Juanjo sin .sig
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I have an IconnectHere account
with a Inbound number and have setup the sip.conf to register and am recieving
the call but When I answer the call it disconnect. I have tried sending the call
to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon
as I accept the
Hi!
I have installed everything, asterisk, pwlib,openh323, chan_oh323. And now?
I want to install ophone to talk, but I don't see what is the asterisk role.
I mean, ophone lets us to talk with another phone,... why do we need
asterisk? What does ophone do and what dows asterisk do?
Thanks for
Steven Critchfield wrote:
On Mon, 2003-10-06 at 23:59, Leif Madsen wrote:
Hey all,
I am in the middle of creating a new user wizard which will generate all
the .conf's the new Asterisk user will require to get themselves up and
running in Asterisk without having to touch a single configuration
Hey all,
I'm having problems reliably dialling out my FXO card. About 30% of the time
I'll get a your call cannot be completed as dialed. I'm thinking it might be
the dialling speed, but I can't find any configs that change that setting.
Any suggestions for troubleshooting?
Thanks,
Brad
On Mon, 2003-10-06 at 20:47, Brian West wrote:
Works fine on my 7960 with 5.3 firmware.
bkw
Im having a similar problem with my 7960 when I receive two incoming
calls I cannot join them.
Hello,
I am trying to conference two or more calls on a Cisco 7940 phone. When I have
Hello folks,
I trying to get working my xten (X-lite V2) working with Asterisk !!!
It's working nice with my development server without nat but NO in my
production server with NAT=yes !!!
Below my client configuration for asterisk:
[general]
port = 5060 ; Port to bind to
Im having a similar problem with my 7960 when I receive two incoming
calls I cannot join them.
ya you can't join them. That sucks.. but you can park one call, go back
to call number 1. Press conf. Dial the parking orbit.. then press join!
bkw
Hello,
I've been playing around with * for quite a while now, and have run into a
problem that I just cannot seem to figure out.
When using * and any IAX client (I have tested with GnoPhone and both
clients from iaxclient.sourceforge.net) I have incredibly bad jitter on the
connection.
On Tue, 7 Oct 2003, Brad Waite wrote:
I'm having problems reliably dialling out my FXO card. About 30% of the time
I'll get a your call cannot be completed as dialed. I'm thinking it might be
the dialling speed, but I can't find any configs that change that setting.
We had the same
OK, I've got my script all set up and running, but now Asterisk crashes when the
digits are entered with the following error:
Ouch ... error while writing audio data: : Broken pipe
I just retrieved and compiled the latest CVS this morning, as well as the latest AGI
perl module. Why won't
On Tue, 2003-10-07 at 11:14, [EMAIL PROTECTED] wrote:
At this point I have extension 8500 setup to take me to voicemailmain. When
I connect (IAX only - I do not have any Digium cards in the server at all) I
^^
ding ding ding,
is it preferred that the T100P generate the T1 clock or that whatever it is
plugged in to (channel bank, PRI, whatever) generate the clock?
Regards,
Andrew
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Hello
Is it possible to make an agi script keep going after a Dial is exectued?
Example:
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$AGI-verbose(-- Hello);
$AGI-exec('Dial',IAX2/whatever); when this call ends the agi script
ends.
$AGI-verbose(-- Hello again); --- it never gets to here
Hello,
Is there any way to park a call on a specific park number?
If this is not possible, is there any way to create multiple park orbits?
Also, is there any way to invoke call parking of an active call coming
through a Zap channel from the manager interface?
MATT---
Hello
Is it possible to make an agi script keep going after a Dial
is exectued?
Example:
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$AGI-verbose(-- Hello);
$AGI-exec('Dial',IAX2/whatever); when this call ends
the agi script
ends.
$AGI-verbose(-- Hello again); --- it never
Thought I would just mention that I have a Pentium 150 with 64MB of RAM,
asterisk installed, 2 Budgetone 102's and an X100P. No problem with
jitter here or anything like that. I don't use mp3 music on hold because
I doubt the hardware would cope particularly well. Has anybody got
Asterisk
Not sure if it's possible to keep the script running after Dial but
perhaps
you could explain what you're attempting to achieve and there may be a
workaround.
I want to know how long the call lasted :)
Panny
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On Tue, 7 Oct 2003, Panny Malialis wrote:
Not sure if it's possible to keep the script running after Dial but
perhaps
you could explain what you're attempting to achieve and there may be a
workaround.
I want to know how long the call lasted :)
Your AGI will continue to run, but
Hi Leif
im not good programmer but if need some help mail to me for everything.
Thanks
Dimitri
On Tuesday 07 October 2003 04:59, Leif Madsen wrote:
Hey all,
I am in the middle of creating a new user wizard which will generate all
the .conf's the new Asterisk user will require to get
Thanks, that makes sense now :)
Panny
- Original Message -
From: James Golovich [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 07, 2003 7:18 PM
Subject: Re: [Asterisk-Users] agi exit problem
On Tue, 7 Oct 2003, Panny Malialis wrote:
Not sure if it's possible to
The way I worked around this is to log the uniqueid in a database when the
call is placed with the start time and then execute an agi script upon all
hangups:
exten = h,1,AGI(call_log.agi,${EXTEN})
That script queries the database for the uniqueid and if it exists in the
table it figures out the
Citeren Dave Wilson [EMAIL PROTECTED]:
Is it possible to make an agi script keep going after a Dial
is exectued?
Example:
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$AGI-verbose(-- Hello);
$AGI-exec('Dial',IAX2/whatever); when this call ends
the agi script
ends.
exten = h,1, will not work if you park a call then pick it back up. You
are flipping the call direction from what Mark told me. Whats wrong with
CDR data? is that not good enough to tell call lenght?
bkw
On Tue, 7 Oct 2003, mattf wrote:
The way I worked around this is to log the uniqueid in
Does anybody out there run * on an ATT broadband phone line? I'm not
seeing any callerid and I can't tell if its ATT doing something funky
or if its my setup. I do see CID on my normal phones
Thanks,
Chris
--
The face of a child can say it all, especially the mouth part of the face.
Hi,
It is posible to put a call in the parking lot with a SIP phone as a
Cisco 7960 ?
Anyway, how can I put a call park on a FXS line ? Is there any magic
digits ?
--
Juanjo sin .sig
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On Tue, 2003-10-07 at 12:09, Brian West wrote:
Im having a similar problem with my 7960 when I receive two incoming
calls I cannot join them.
ya you can't join them. That sucks.. but you can park one call, go back
to call number 1. Press conf. Dial the parking orbit.. then press
I am trying to compile * on SuSE 8.2. When doing the make install in
/usr/src/zaptel I get the following error.
**
/usr/src/linux/include/asm/system.h:189: warning: dereferencing
type-punned pointer will break strict-aliasing rules
freeIn
I'd like to float the idea of a VON panel discussion for large-scale
open-source deployments, specifically using Asterisk as an
application service and as a gateway service. In order to do that,
I'd need to get a list of panelists together who might be interested
in speaking. If you manage
Hello,
Is there any way to park a call on a specific park number?
Not to my knowledge.
If this is not possible, is there any way to create multiple park orbits?
Not to my knowledge.
This seems to be lagging behind some of the other features within
Asterisk which allow compartmentalization of
On Tue, 2003-10-07 at 14:10, rnc Info Lists wrote:
I am trying to compile * on SuSE 8.2. When doing the make install in
/usr/src/zaptel I get the following error.
**
/usr/src/linux/include/asm/system.h:189: warning: dereferencing
On Sat, 2003-10-04 at 15:09, Jan Rychter wrote:
Any chance you could describe the hardware? Was it a Via-based board?
I have a setup where I use two *'s, both on Via boards. One is a
Mini-ITX and the other is a full-form motherboard.
Would interrupt-sharing between the X100P and another
I'm coming at this thing from an Operational standpoint rather than a development
standpoint. Viewing your problem from that angle, I wonder how well your network is
performing. Could you have a cable problem that the Asterisk server hasn't reported
(Layer 1); or perhaps your * Server is
I dont see it as a bug.. I see why it don't work.. and why people think it
should. Enable # transfers.. and setup call parking to get around this.
Also if you conf very much look at app_meetme.
bkw
On Tue, 7 Oct 2003, Juan J. Sierralta P. wrote:
On Tue, 2003-10-07 at 12:09, Brian West wrote:
Not yet.. but I sure wish we could... :)
On Tue, 7 Oct 2003, Juan J. Sierralta P. wrote:
Hi,
It is posible to put a call in the parking lot with a SIP phone as a
Cisco 7960 ?
Anyway, how can I put a call park on a FXS line ? Is there any magic
digits ?
--
Juanjo sin .sig
Dimitri Bellini wrote:
Hi Leif
im not good programmer but if need some help mail to me for everything.
Yah... me niether :)
At this stage it is simply going to be figuring out the logic so that I
know I have asked all the questions that need to be asked. If you can
think of things
is it preferred that the T100P generate the T1 clock or that whatever it is
plugged in to (channel bank, PRI, whatever) generate the clock?
That depends on the environment
This is what i have read b4 about t1 timing srcs
1. If the T1 is point to point where both ends terminate on a different
*
I have a problem that sometimes lines will go into what I call never never land. The
Asterisk system will put a line with Zombi on it when you type show channels it will
make the analog phone line dead. And on the CLI it says:
astsvr*CLIZap/1-2ZOMBIE(macro-twoline-exten,s,1)Up Dial
On Tue, 2003-10-07 at 14:51, Ariel Batista wrote:
I have a problem that sometimes lines will go into what I call never
never land. The Asterisk system will put a line with Zombi on it
when you type show channels it will make the analog phone line dead.
And on the CLI it says:
Brian,
Would you be kind enough to give me a brief overview of why it doesnt work. I also
appreciate the work aorund. This is something I will have to educate my soon to be
users on. We do a lot of conferencing of calls as a matter of facilitating clients'
immediate needs.
For now I will
I had a similar problem with redhat 9 stock kernel sources.
I had to enter the kernel sources dir,
do a make mrproper
then a make menuconfig
save the conf do make dep.
after that I was able to build zaptel without issues ;)
matteo.
Il mar, 2003-10-07 alle 21:10, rnc Info Lists ha scritto:
I
On Fri, 3 Oct 2003, Richard Lyman wrote:
you'll find that the context is being overwritten.
look in chan_iax.c line 1628 and chan_iax2.c line 1645 (or within
3 lines of each)
there is a sprintf that is stuff the context, if you comment
those out, it should work again.
Disclaimer: i have NO
Leif Madsen wrote:
think of things I've missed, feel free to chime in!
Leif, why don't you put the script up on the wiki so that we all can
edit it and add on line with versioning?
As a newborn Asterisk user, I had severe problems configuring an ISDN card.
I believe a lot of new users start with a
I would love to have separate callparking contexts available, it's omission
the reason I have not been using it up to this point.
As for redirect I haven't tried it yet, I just want to use the manager
interface to send a call on a zap channel to a parkedcall extension(ext.
700){or a specific park
Hi guys,
Thanks for your answers on my two questions yesterday. That's exactly
what I was looking for, sorry for not noticing it myself, but I'm still
getting acclimated to Asterisk and even Linux--from what I see so far, I
love it.
I've got another one now. Since my Asterisk install and
Why does having a call go through call parking make the h not work?
I currently don't use call parking for other reasons so I've never run into
that.
What is the event to run an agi script after a parked call is hung up?
I don't use CDR data because I have a custom perl/TK interface that grabs
On Tue, 2003-10-07 at 15:38, Brian West wrote:
I dont see it as a bug.. I see why it don't work.. and why people think it
should. Enable # transfers.. and setup call parking to get around this.
Also if you conf very much look at app_meetme.
I did it, problem that I have now is the
On Tue, 2003-10-07 at 15:13, Brancaleoni Matteo wrote:
I had a similar problem with redhat 9 stock kernel sources.
I had to enter the kernel sources dir,
do a make mrproper
then a make menuconfig
This was probably not a good idea as you have configured your kernel
source differently than
-- Original Message --
From: Steven Critchfield [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Tue, 07 Oct 2003 15:06:12 -0500
On Tue, 2003-10-07 at 14:51, Ariel Batista wrote:
I have a problem that sometimes lines will go into what I call never
cd /usr/src/asterisk; make config; cd /usr/src/zaptel; make config
regards
Martin
On Tue, 7 Oct 2003, john lawler wrote:
Hi guys,
Thanks for your answers on my two questions yesterday. That's exactly
what I was looking for, sorry for not noticing it myself, but I'm still
getting
On Tue, 7 Oct 2003, john lawler wrote:
But, when I come back from a restart, it appears that the Asterisk
startup failed, and I think it's b/c the wct1xxp module is not loaded.
What is the recommended way to ensure this happens? I've been reading
and found that modprobe (on startup, it
On Tue, 2003-10-07 at 15:30, john lawler wrote:
Hi guys,
Thanks for your answers on my two questions yesterday. That's exactly
what I was looking for, sorry for not noticing it myself, but I'm still
getting acclimated to Asterisk and even Linux--from what I see so far, I
love it.
Olle E. Johansson wrote:
Leif Madsen wrote:
think of things I've missed, feel free to chime in!
Leif, why don't you put the script up on the wiki so that we all can
edit it and add on line with versioning?
It's an interesting idea.. but I'm not sure if a wiki is the best place
for code...
john lawler wrote:
Hi guys,
Thanks for your answers on my two questions yesterday. That's exactly
what I was looking for, sorry for not noticing it myself, but I'm still
getting acclimated to Asterisk and even Linux--from what I see so far, I
love it.
I've got another one now. Since my
On Tue, 2003-10-07 at 15:30, Ariel Batista wrote:
I have tried to release it with soft hangup Zap/1
also soft hangup Zap/1-2. If I use the last one is say trying to
hang up but it never does. I have to shut the system down then back
up! I am not able to run the GASTMAN due to I have no
On Tue, 7 Oct 2003 16:30:52 -0400, Ariel Batista [EMAIL PROTECTED]
wrote:
astsvr*CLIZap/1-2ZOMBIE(macro-twoline-exten,s,1)Up Dial
Zap/1-2|20|r
I have tried to release it with soft hangup Zap/1
also soft hangup Zap/1-2. If I use the last one is say trying to
hang up but it never does.
[...]
I
Ok I have the following on the Asterisk every minutes.
Got SIP response 481 Call Leg/Transaction Does Not Exist back from 2XX.2XX.133.1XX.
The user of this Sip phone is able to make calls and get calls. A Windows 2000 pro
using MS Messenger! I loaded it on my PC as well and it does the same
There's been a few replies but thought I'd elaborate on my initial reply..
How are you dropping the 456 there? I thought extensions picked
up what either the SIP phone had dialled, or what DTMF detection
picked up when * answered the line...?
No.. if you have a PRI, the signalling is digital,
I am wondering if it's possible to use a bunch of cards in a PCI
backplane instead of going out to the extensions with T1 and then and
adapter.
How are people connecting to large amounts of extensions?
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Wouldn't it be cool if you could take the call back, into
the IVR script after the remote party hangs up ?
Yeah, for the exact reasons you suggested!
We had an IVR product that would allow you to zero out to the call centre,
get some help, top up your account, whatever, then when the agent hung
You are wanting to use a PCI backplane and put a bunch of TDM400P FXS
cards instead of a T1 and a channel bank? If that's what you're asking...
A T1 card and a channel bank yield 24 extensions. If you figure the
TDM400P is $305 for 4 extensions, it would cost $1830 to get enough FXS
ports (not
Each card needs it's own IRQ, not shared with any other device (not even
shared with other Digium cards). Adding a PCI backplane gives you more
slots, but not more IRQs.
On Tue, 2003-10-07 at 16:23, Dennis Gearon wrote:
I am wondering if it's possible to use a bunch of cards in a PCI
On Tue, 2003-10-07 at 16:09, Babak Pasdar wrote:
Brian,
Would you be kind enough to give me a brief overview of why it doesnt work. I also
appreciate the work aorund. This is something I will have to educate my soon to be
users on. We do a lot of conferencing of calls as a matter of
Three days after launching our * system with 20 GS phones, I have
finally had to give up on dynamic registration. The phones keep
dissappearing from the sip peers list, even if just sitting idle.
Either I spend half my time re-booting phones to get them registered, or
the extension appears
Three days after launching our * system with 20 GS phones, I have
finally had to give up on dynamic registration. The phones keep
dissappearing from the sip peers list, even if just sitting idle.
Either I spend half my time re-booting phones to get them registered, or
the extension appears
Hi.
Since a customer requested us that feature, I wrote this
little patch for app_dial to allow to play an
announcement to the called party, as soon he answers.
you can define the file to play in the dial() option,
using A(filename).
for example:
exten =
Hey all..in trying to futher troubleshoot my caller id problem I'm
looking at some past troubleshooting tips and this struck me as strange:
If I cat /dev/zap/1 I *always* see data...no matter if the line is in
use or not...is that typical? Just curious...
Thanks,
Chris
--
People are not
How are you transfering to 700? You dial # while in a call and then it
says transfer and you then dial 700, or are you using a different
method?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Juan J. Sierralta P.
Sent: Tuesday,
He he ... too early
Thanks to a quick info from Mark on irc,
I've added the autoservice stuff on the other
channel, that's doing nothing meanwhile.
So here's the correct patch. discard the previous one.
Matteo
Il mer, 2003-10-08 alle 00:11, Brancaleoni Matteo ha scritto:
Hi.
Since a
I'm debugging SIP registration too. My next step is to
install an Ethernet sniffer to log everything that goes ovr the
wire using ports 5060 and 8000~8020. I'll soon know what's up.
You would be able to see if the phones are forgetting to register,
or if they are and Astrisk is dropping the
-- Original Message --
From: Ryan Tucker [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Tue, 07 Oct 2003 17:08:22 -0400
On Tue, 7 Oct 2003 16:30:52 -0400, Ariel Batista [EMAIL PROTECTED]
wrote:
astsvr*CLIZap/1-2ZOMBIE(macro-twoline-exten,s,1)Up
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