Hi Folks,
Is it possible to setup some parameter on Dial command to
hangup a call if the customer press # ?
Thanks,
Isamar
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Somesh,
On 2/3/06, Somesh S Shanbhag [EMAIL PROTECTED] wrote:
I want to do a three-party conferencing using manager api.
But I found out from the asterisk-users list that I *MUST* use
the meeting room concept.
I wanted to know wheather meeting room can be configured dynamically?
on the
Remember that the *8 in your features.conf has nothing to do
with direct pickup. So in your case try replacing _86. with
_*8. but I don't know if that will cause problems.
Yes!!!
I thought that this was a feature too instead it's a dialplan application.
Asterisk is a bottomless sea.
Thanks
Have you seen that 3 Asterisk servers were running during this show ?
François,
I was there (had a coffee with Dave in fact) but was wondering, there
was no official asterisk presence, was there? Maybe we should have
helped organize this as * is a Linux Solution
Ok, I think I am getting somewhere. When I am ringing extension 200 I
do a show channel SIP/200 and this is what I get:
-- General --
Name: SIP/200-b699
Type: SIP
UniqueID: asterisk-2177-1138957721.175
Caller ID: s
Caller ID Name: (N/A)
DNID Digits: (N/A)
Hi Alex
I tried your exact example below and still the same thing. I am getting
403 Denied after I see the Pickup cmd in the CLI. If you do a show
channel SIP/XXX when the phone is ringing, do you get a value for
Extension:??
Kind Regards
Garth
Alex Barnes wrote:
-Original
Hi,
I'm having a problem calling international numbers with debian's
Asterisk 1.0.7 w/ zaptel 1.0.9 in Moscow. Russia doesn't seem to have
touchtone dialing, so pulsedial is enabled on my TDM400P interface.
Local numbers work fine, but when it comes to long distance or
international, I'm lost.
Hi,
I'm trying to write a small Visual Basic app to throw a popup with
CallerIDNum when a call center agent answers a queue call.
Does anyone know what is the right manager event to intercept?
Thanks
Mimmus
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Link event2006/2/3, Mimmus [EMAIL PROTECTED]:
Hi,I'm trying to write a small Visual Basic app to throw a popup withCallerIDNum when a call center agent answers a queue call.Does anyone know what is the right manager event to intercept?ThanksMimmus
It works. Thanks a lot.
With 15/20 users, is it better to use a manager proxy or to
connect directly to the Asterisk server?
Thanks
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giovanni
MianoSent: Friday, February 03, 2006 11:42 AMTo:
Asterisk Users Mailing
Bartosz Jozwiak wrote:
Check if rxfax actually receives anything...
How?
--
Best regards,
Bartosz Piec
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Dial event, in Asterisk 1.2:203-201Event: Dial Privilege: call,all Source: SIP/203-8467 Destination: SIP/201-45d9 CallerID: 203 CallerIDName: 203 SrcUniqueID: asterisk-1912-1138197095.3769 DestUniqueID: asterisk-1912-1138197095.3771Mimmus [EMAIL PROTECTED] ha scritto: Hi,I'm
Right. My original question was about making Asterisk wait a number or
rings (or amount of time) before picking up a Zap line. If the
rings/time were not reached while the line is still ringing, do nothing.
As someone must have already said, it's not a good idea to share lines
with asterisk.
Florian Overkamp wrote:
Hi Ronald,
Ronald Wiplinger wrote:
voipbuster/ 194.221.62.201 5060
UNREACHABLE
voipstunt/x 194.120.0.200 5060
a reload shows than:
voipbuster/ 80.239.235.200 5060
Pierre Burton wrote:
What's your cisco conf ? how did you transfert between Cisco and
asterisk ? A-law, U-law ??
This is part of my Cisco config:
voice-card 0
no dspfarm
!
!
!
voice service voip
sip
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec
There is an issue here in France with our Siemens DECT phones that
required a patch to change the ring _frequency_. It was given here
ages ago, but now I can't find it.
Shame on me for not coming back!
//{20,7,RING_OSC,0x7EF0}, // changed to
{20,7,RING_OSC,0x7E6C}, // new value for 25hz
for
I have been playing with the Signate switch. Official training starts
soon but just playing with it leaves me with the impression that it is
powerful but very complex. You need to RTFM to get anything working.
They have also used IonCube to encode all PHP and HTML files so
customization is
Bartosz Piec wrote:
Hello,
I'm trying to receive faxes with asterisk. My configuration is like this:
Codec?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk
Matt Riddell (IT) wrote:
I'm trying to receive faxes with asterisk. My configuration is like this:
Codec?
In Asterisk or in Cisco?
--
Best regards,
Bartosz Piec
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hello!
has this made it into 1.2.3 already:
http://bugs.digium.com/view.php?id=6128 ?
i'm trying to set a variable that should be used as a dialstring in the
dial-command, including parameters seperated with the respective
delimiter, e.g. like:
exten =
How can I configure to call from the console by means of a sip phone,
any docs on this.
Regards,
Anthony.
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Title: Musiconhold in zapata.conf
I've been trying to change the musiconhold= in the zapata.conf to use something other than default. However it doesn't seem to do it. I know the other musiconhold source works but whatever I set it to in the zapata.conf file it always plays whatever is
Hello,
There're few POTS supporting touchtone, others - just pulse. In
Russia you need to dial 8, wait for tone and only then continue
dialing 10 (for intl. plan), country code, area code and number.
bkmc Hi,
bkmc I'm having a problem calling international numbers with debian's
bkmc Asterisk
On Fri, 2006-02-03 at 09:52 +0100, Wilson Pickett wrote:
Have you seen that 3 Asterisk servers were running during this show ?
François,
I was there (had a coffee with Dave in fact) but was wondering, there
was no official asterisk presence, was there? Maybe we should have
helped
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
The GPL primarily deals with linking to the libraries of a GPL project.
I am not aware of any changes they made directly to asterisk, php,
mysql etc that would bind them to the GPL. However, if they are
using/requiring mysql, then they may have to
i'm planning to migrate a call center to asterisk, i don't understand
if i can launch a resident application on the agent's client in
relation with the queue the agent's is answering.
For example:
I have
- queue A
- queue B
- queue C
Agent 100 (logged in A.B,C)
Agent 101 (logged in C)
When
Hi
i'm planning to migrate a callcenter to asterisk and VOIP, the call
center can have up to 25 cuncurrents agents logged in.
I'll have some simplty IVR business logic and the some queues.
Can a normal server with
1 GB ram
100 GB HDD
Pentium 4 3.6 Ghz CPU
Ethernet 10/100/1000
Support this?
I have been planning to do the same thing but never got around to it, I
actually did write a nice class to wrap the interface to the manager but
it isn't complete.
Would you be willing to share your work?
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Hi, ALL:
Can anyone tell me what *RT is ?
What is its full name? I think the * is asterisk but what is RT ?
2006/2/2, Rusty Shackleford [EMAIL PROTECTED]:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alistair Cunningham
Sent: Wednesday,
I think that the range of this question is too large.
You should tell us what your scenario is. And tell us more about your
configurations.
2006/2/2, Jack Wei [EMAIL PROTECTED]:
hi,
I'm trying to set up 2 SER and 2 Asterisks boxes using a bigip switch to do
load-balancing. I'm using Asterisk
Le Vendredi 3 Février 2006 13:54, Dave Cotton a écrit :
On Fri, 2006-02-03 at 09:52 +0100, Wilson Pickett wrote:
Have you seen that 3 Asterisk servers were running during this show ?
François,
I was there (had a coffee with Dave in fact) but was wondering, there
was no official
My remote 64kbit connection would only be used for VoIP and NOTHING else! No
email, no browsing. Besides, my remote 64kbit guaranteed-bandwidth
connection changed into a 256/512 ADSL connection from the same telco
provider (that's actualy wy cheeper then the other 64kbit guaranteed
connection)
I would be happy to share everything but actually I'm working only at a
feasibility study.
In addition, I'm a system admin and development job is made by someone else!
In principle, it's simple: open a socket to manager port, login and wait for
right event.
Ideal target is a small, traybar
OK with that being said how can you modify the phone to use the second line
button as a speed dial? Then you can label it has flash.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Thursday, February 02, 2006 11:28 PM
To:
I have been doing multipath between 2 cable modems for over 2 years now.
e-mail me off list and I will get you my configs for this.
Cheers,
/Zac
[EMAIL PROTECTED] wrote:
Yes, and, you will probably need a different method.
Are these t1's to the same provider? Have you considered bonding
Hi,
Has anyone used the varion v400p card? what's its performance??
Rgds,
_
Express yourself instantly with MSN Messenger! Download today it's FREE!
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
On Thursday 02 February 2006 11:40, Arne Morten Johansen wrote:
How do I setup a Callback script?
This script does what I want to do. But how do I set it up?
http://www.junghanns.net/en/callback.html
I see it uses PHP for scriptlanguage. So where do I place it (the .agi)?
Imran Ahmed wrote:
may or may not work, try at your own risk:
1) Use a sip soft phone and set the dtmf mode = inband.
2) In asterisk set the dtmf mode for that soft phone to be rfc2833 or
info. (this is done so that asterisk ignores the inband dtmf on the
sip channel).
3) Design your dialplan
Hi Group, I
am sending my question again
why I dont have answer yet:
I am developing a application, this use
Manager API to connect with Asterisk. But when I call to an
external number (over a zap channel), I dont receive any event when the target
answer, Who can help me?, Which event
Hi,
do you require more information about this behaviour, I'd be more
than glad to provide it.
thanks,
kr,
Bart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bart van Daal
Sent: woensdag 1 februari 2006 11:03
To: 'Asterisk Users Mailing List -
Well in my setup I have a few IP phones connected to Asterisk as well as POTS
phones on my analog line. When a call for my daughter comes in on the analog
line (determined from callerID) I send it to her own voicemail after 20 seconds
of ringing. It all works quite well.
Here's a step-by-step
Fabrice a écrit :
Le Vendredi 3 Février 2006 13:54, Dave Cotton a écrit :
On Fri, 2006-02-03 at 09:52 +0100, Wilson Pickett wrote:
Have you seen that 3 Asterisk servers were running during this show ?
François,
I was there (had a coffee with Dave in fact) but was wondering,
Realtime.. As in pulling configs from a realtime database..
Or he's trying to link Asterisk to www.bestpracticals.com version of
Request Tracker (also known as RT)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Charles
Wang
Sent: February 3, 2006 8:13
Akpome Akpoguma wrote:
Hi,
Has anyone used the varion v400p card? what's its performance??
Rgds,
Its the Tormenta 2 card, just like the old T400P and E400P cards from
Digium. See www.zapatatelephony.org for details.
Steve
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You should write a proxy and not connect directly, the reasons are as follows:
1. You don't want asterisk to crash because of problems with the
manager app over the network, which Asterisk is known not to handle
very well (as per the wiki).
2. Security, if you have every computer connecting to
Hi Ronald,
Ronald Wiplinger wrote:
You could read out all the entries in the DNS zone and create your own
list of entries in /etc/hosts, and then create multiple asterisk
peers: voipbuster1, voipbuster2, etc... Then you can use regular
dialplan logic to cycle through all of them.
that is
You should create a secret dialing prefix like if you wanted to dial 1555333222 the user would actually have to dial 548261555333222. This way, even if they snatch the username/password but do not know the prefix, they won't be able to dial.
On 2/2/06, Miguel [EMAIL PROTECTED] wrote:
[EMAIL
Does anyone know of a special character used in pattern matching that
would match 0 or 1 digit?
I would just like to cut down on the number of extensions I have.
Current example:
exten = 2125551234,1,Dial(SIP/2125551234,15,rt)
exten = 12125551234,1,Dial(SIP/2125551234,15,rt)
I would like to do
Yes, it is possible. You need to track the queue log and channels via manager console or by tailing logs in real time and then match the destination of the caller by the callerid. Then make the decision which URL to redirect the caller too. None of this comes with Asterisk but it is possible to
I 'm developing something similar. It a perl script which tells you
who is calling but it do it by sendind a jabber message.
it's my first perl script so it's not finished yet.
i'll share it so you can contribute if you want...
2006/2/3, C F [EMAIL PROTECTED]:
You should write a proxy and not
I think I have the same issue...
In case usershave an IP Phone on their desks
and Softphones on their PCs and are configured with the same username
extensions, which phone will ring? The one that last sent the
REGISTER...
This can be conflicting...
- Original Message -
On Friday 03 February 2006 10:21, Facundo Ameal wrote:
I 'm developing something similar. It a perl script which tells you
who is calling but it do it by sendind a jabber message.
it's my first perl script so it's not finished yet.
i'll share it so you can contribute if you want...
not sure what you want, but for multiple returns i use
Set(AGI_STATUS=mystatus), so in the dialplan i just check for the
variable AGI_STATUS and do whatever i need depending on the status.
regardsOn 2/2/06, [EMAIL PROTECTED] [EMAIL PROTECTED]
wrote:Hello friends,Asterisk
applications like Dial
On Fri, 2006-02-03 at 08:49 +0100, [EMAIL PROTECTED] wrote:
From what I understand it means that the *hardware* in your computer
*acknowledges* the call as soon as it is recieved and then sends it to
asterisk dialplan for processing.
Hrm. Yes, that is what I got from it. But in my case the
Hi,
I have an IBM xSeries
206 and now looking at the Wildcard TE110P to connect to our ISDN30. Has
anyone any experience with this combination? Would the TE110P work
in this server? I've listed the PCI slots the machine has:
2 ( 2 ) x PCI-X / 66 MHz - full-length ¦ 3 ( 3 ) x PCI
- full-length
_N
On 2/3/06, Sum Ding Wong [EMAIL PROTECTED] wrote:
Does anyone know of a special character used in pattern matching that
would match 0 or 1 digit?
I would just like to cut down on the number of extensions I have.
Current example:
exten = 2125551234,1,Dial(SIP/2125551234,15,rt)
exten =
Hi,
We are ordering a bank of numbers from
our provider BT. We will have an ISDN30 with 8 channels enabled.
Is it possible to do this? Is this known as DDI? Can anyone
give tips on how to configure the Asterisk server so that users are available
on the extensions.
Hope this explains this
Ezequiel A. Sculli wrote:
Hi Group, I am sending my question again why I don’t have answer yet:
I am developing a application, this use “Manager API” to connect with
Asterisk. But when I call to an external number (over a zap channel), I
don’t receive any event when the target answer, Who
Hi,
The usual bottleneck is cpu.
i'm planning to migrate a callcenter to asterisk and VOIP,
the call center can have up to 25 cuncurrents agents
logged in.
Ie. max 25 concurrent calls.
I'll have some simplty IVR business logic and the
some queues.
Unknown number of concurrent calls
Christian Benke wrote:
hello!
has this made it into 1.2.3 already:
http://bugs.digium.com/view.php?id=6128 ?
i'm trying to set a variable that should be used as a dialstring in the
dial-command, including parameters seperated with the respective
delimiter, e.g. like:
exten =
You recive Link event when Channel Caller and Channel Called bridged, match it and good luck.cheers,Giovanni2006/2/3, Ezequiel A. Sculli
[EMAIL PROTECTED]:
Hi Group, I
am sending my question again
why I don't have answer yet:
I am developing a application, this use
"Manager API" to
Dan Journo wrote:
Ok, i feel like im getting somewhere but i need a little help.
Asterisk displays this when its loading:-
[res_musiconhold.so] = (Music On Hold Resource)
== Registered application 'MusicOnHold'
== Registered application 'WaitMusicOnHold'
== Registered application
I've been using it in a test environment with no problems. However, I
haven't used it in production yet. I'm doing some voice broadcasting
with a PRI and so far I'm content with the performance.
-MC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
N matches any digit from 2-9. Are there any other wildcards outside of
the ones listed below
http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns
X matches any digit from 0-9
Z matches any digit form 1-9
N matches any digit from 2-9
[1237-9]
On 2/3/06, Script Head [EMAIL PROTECTED] wrote:
Yes, it is possible. You need to track the queue log and channels via
manager console or by tailing logs in real time and then match the
destination of the caller by the callerid. Then make the decision which URL
to redirect the caller too. None
I've also purchased their GUI and hoped it would work for us, but the lack of
proper documentation, horribly garbled tech support lines (support seems to
come from Australia, and they apparently use very low quality voip
trunks),broken installer, and cryptic interface forced me to reconsider.
hi to all
i have a website powered by a c# CMS
i have an asterisk in our office
my need is that my customers could surf on my
website, click the phone button, a sip call is established between the
website (sip client) and my phone allowing me to talk with them
any idea where i can find the
About once an hour, my Sipura 2002 rings just once. I thought it might be
faulty, so I configured a second one, and it does the same thing. I
updated the firmware to 3.1.5 and still have the same problem.
Anybody able to shed some light on the random ringing problem?
Thanks.
Accursio Avona wrote:
but maybe also (i'm not sure, i had not the time to study well enough
the source, and over all i'm not a so good c programmer)
that this part of code prevents asterisk to broadcast the sound to other
channels when it is not inband.
MeetMe is not designed to pass DTMF
[EMAIL PROTECTED] wrote:
Hi,
We are ordering a bank of numbers from our provider BT. We will have an
ISDN30 with 8 channels enabled. Is it possible to do this? Is this
known as DDI? Can anyone give tips on how to configure the Asterisk
server so that users are available on the
On Fri, 2006-02-03 at 12:19 +0100, Wilson Pickett wrote:
As someone must have already said, it's not a good idea to share lines
with asterisk.
Well, yeah, ideally I have the phones on an FSX, but a) I don't have one
yet and b) I want to make sure I am happy with running a PBX before I
invest
Steve Totaro wrote:
Another question, If Signate is not using ABE, what are their
requirements for releasing source as far as the GUI?
The Asterisk GPL has no bearing on the external tools used to
manage/configure it, unless those tools require changes in Asterisk
itself or loadable modules
Hi,
I have a provider
sending me data through SIP, but with no registration. (there are
constraints that forces us to work like this). And, as far as I am
concerned, that's fine.
Here is the relevant
portion of my SIP.conf file.
On Fri, 2006-02-03 at 07:37 -0700, Bromont Quebec wrote:
Well in my setup I have a few IP phones connected to Asterisk as well as POTS
phones on my analog line.
Ahhh. So we share the latter at least.
When a call for my daughter comes in on the analog line (determined from
callerID) I send
hi,
I'm trying to get the latest chan_sccp. The links from
http://chan-sccp.berlios.de are all dead. Is it just me? Does anyone
know an alternate source to get chan_sccp?
Seems like chan_sccp is the way to go if I have a bunch of cisco
phones I don't want to use SIP on? Anyone
On 2/3/06, Jeremy Koski [EMAIL PROTECTED] wrote:
About once an hour, my Sipura 2002 rings just once. I thought it might be
faulty, so I configured a second one, and it does the same thing. I
updated the firmware to 3.1.5 and still have the same problem.
Anybody able to shed some light on
Matt Riddell
wrote:
Ezequiel A.
Sculli wrote:
Hi
Group, I am sending my question again why I don_t have answer yet:
I am
developing a application, this use _Manager API_ to connect with
Asterisk. But when I call to an external number (over a zap
channel),
I don_t
receive any
Hello, I want to know if someone made a script in
php(with agi) to call some voip number, and when the user answer the call, he
hears a message with an advertisement. I want to input the number directly from
cli or read the numbers from a file(ex.8021,8022,8023).
Thanks in advantage
Ever
One of our Telephony Server 5000 modules will throughput between 2,000 and
2,500 SIP calls with streams if it is doing no other work. One of these days
we will again announce the details of the ongoing benchmarks that we perform
with the help of system engineers from a major computer
Listen to your voicemail.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jeremy Koski
Sent: Friday, February 03, 2006 9:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Sipura SPA-2002 rings randomly
On Fri, 3 Feb 2006, Andy Webster wrote:
hi,
I'm trying to get the latest chan_sccp. The links from
http://chan-sccp.berlios.de are all dead. Is it just me? Does anyone
know an alternate source to get chan_sccp?
Seems like chan_sccp is the way to go if I have a bunch of cisco
Kevin P. Fleming wrote:
but maybe also (i'm not sure, i had not the time to study well enough
the source, and over all i'm not a so good c programmer)
that this part of code prevents asterisk to broadcast the sound to
other channels when it is not inband.
MeetMe is not designed to pass DTMF
Andy Webster ha scritto:
hi,
I'm trying to get the latest chan_sccp. The links from
http://chan-sccp.berlios.de are all dead. Is it just me? Does anyone
know an alternate source to get chan_sccp?
Just tested, all the links work
Sergio
As BJ mentionned, it could be your MWI of depending on your profiling, it
might be scheduled to download it's profile every hour, and therefore might
reboot and ring after each download
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Koski
Sent:
On Fri, 3 Feb 2006 11:41:53 +0100
Giovanni Miano [EMAIL PROTECTED] wrote:
Link event
For me, Link event only occurs when the called number pickup the call.
I prefer 'Newchannel' event when the 'State' are equal to 'Ringing'
--
Iuri Gomes Diniz adm.iuri (at) digi.com.br
Network Admin and
I'd change your definition to something
like
[providerX]
context=providerX-inbound
host=11.222.222.23
in your providerX-inbound context you can match the
different extensions
[providerX-inbound]
exten =
514907,1,NoOp(514907)
exten =
55,1,NoOp(55)
Now a question
Benjamin,
Thanks a lot for the answer. Sometimes the obvious escapes me, and this was
the case here.
Regards,
Mike
I'd change your definition to something like
[providerX]
context=providerX-inbound
host=11.222.222.23
in your providerX-inbound context you can match the different
Hi,
Sorry to ask a slightly off topic question here, but I've been stuck
on this for a while.
My SIP ATA's are displaying callerID without problems. The problem is
when a 2nd call comes in during a conversation, callwaiting callerID
dosen't show up. I can only hear the callwaiting alert tones,
At 07:34 AM 02/03/2006, you wrote:
I am pressuming that since I can use functions like Wait(), then
Answer() in dialplan to actually delay answering (for the Wait()
time) that Asterisk actually acknowledges the call.
I think you need to use dial instead of answer. You can put a timeout
in
Thanks, that was it.
On Fri, 3 Feb 2006, Benjamin Lawetz wrote:
As BJ mentionned, it could be your MWI of depending on your profiling, it
might be scheduled to download it's profile every hour, and therefore might
reboot and ring after each download
-Original Message-
From: [EMAIL
On Fri, 2006-02-03 at 10:59 -0800, Ira wrote:
I think you need to use dial instead of answer. You can put a timeout
in dial and if the call is hung up dial will exit.
Hung up? By whom? Assume this: while Dial() is working (and waiting
for the timeout) somebody has picked up a phone that
My Sipura 1001 does that when I have a message waiting. You can turn the
reminder half-ring off in the configuration settings.
About once an hour, my Sipura 2002 rings just once. I thought it might be
faulty, so I configured a second one, and it does the same thing. I
updated the firmware to
The way to first test the ATA is with a phone or caller ID display that
supports caller ID with call waiting. Some devices work that way by
default and some might require you to set the option. That test should
have nothing to do with zapata.conf at all. I assume the reason you
mention
Curious about this device as well. Seems
almost too good to be true?
The built-in switch feature would be much handier
than having to get a router for the extra network connection;
Also, the 'life line passthru' thing seems
interesting -- although I have no idea what a life line passthru
Yeah I want both my POTS phones and SIP phones to ring at the same time, that
way I have the choice to answer whatever one is most convenient. If a POTS
phone picks up, the Zap channel closes and Asterisk does nothing more, if a SIP
phone picks up, Asterisk connects the Zap channel to
On Fri, 2006-02-03 at 13:00 -0700, Bromont Quebec wrote:
Yeah I want both my POTS phones and SIP phones to ring at the same time,
that way I have the choice to answer whatever one is most convenient. If a
POTS phone picks up, the Zap channel closes and Asterisk does nothing more,
if a SIP
There you go. if it is doing no other work is key phrase. A lot of PC can do
that these days if all it has to do is re-route packets to different
destinations, and guess what, if you make sure silence compression is turned on
at the endpoints, you can claim even more streams can be passed
At 08:07 AM 02/03/2006, you wrote:
exten = 907,1,Set(DESTINATION1=Zap/G1/4989123456789|10|gh)
exten = 907,n,Set(DIALSTRING=${DESTINATION1})
exten = 907,n,Dial(${DIALSTRING})
Set multiple variables? One for each option maybe?
Or call a macro instead and have the macro split it apart, then
At 11:38 AM 02/03/2006, you wrote:
Hung up? By whom? Assume this: while Dial() is working (and waiting
for the timeout) somebody has picked up a phone that shares the POTS
line with Asterisk. Will that second pick up of the POTS line look like
a hangup on the POTS line to Asterisk while it is
Hi,
After installing mysql, mysql-devel mysql cdr add
on, I get the following error when I start Asterisk:
[res_config_mysql.so]2006-02-03 18:41:16
WARNING[24786]: loader.c:325 __load_resource:
/usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol:
_intel_fast_memcpy
My server
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