On Sun, Feb 12, 2006 at 03:35:56PM -0500, John Novack wrote:
That certainly is the way it SHOULD work. Blind and attended transfer
should be able to be initiated the same way. It certainly is the most
efficient logical way. Attended transfer should revert to blind simply
by the initiating
[EMAIL PROTECTED] wrote:
Maybe do a transfer to a dedicated extension, which calls the script
with the system() command to open the door? Or use the feature keys for
a blind transfer. Seems like it could work.
Btw, what kind of door phone opener do you have? I've been looking for
Hi,
In the Zap Channel Module's configuration file (zapata.conf), you can define groups of Zap channels that get treated as a single channel as far as the
Dial command is concerned. I want to know if it's possible to do same thing with SIP channels.
Because I have found some not too clear
Just a reminder, there is now a list dedicated to the 480i:
http://groups.google.com/group/Aastra-480i-Users
Come join in the fun ;-)
Also, remember that there is quite a bit of good info on the voip-info.org 480i page
Richard
On 2/12/06, Carlos Chavez [EMAIL PROTECTED] wrote:
Does anyone know
Hi guys. I've found a fair bit of information with regard to how to setup
Asterisk to send and receive calls through SER (SIP Express Router),
however I can't figure out what information is up-to-date. Many of the
examples suggest using insecure=very for the SER entry in sip.conf, but I
María Chóliz wrote:
Hi,
In the Zap Channel Module's configuration file (zapata.conf
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf),
you can define groups of Zap channels that get treated as a single
channel as far as the Dial
Hi,
Does anyone have any experience connecting asterisk to alcatel 4200
series pbx with bri cards?
Does it should work with asterisk bri in NT mode, and alcatel bri with
TE mode?
Cheers, Igor Neves.
___
--Bandwidth and Colocation provided by
On Mon, 13 Feb 2006, [EMAIL PROTECTED] wrote:
I am trying to figure out which way to go for a quad port fxo solution
with a good echo can on it. My options are the sangoma remora, a
mediatrix fxo, or something similar.
The issue is that I would need a good EC. This would be on about a 9000
Hi,
I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch.
When forwarding a call to Voicemail, here is somehow what the softswitch sends to Asterisk :In INVITE : Vm Phone Number ( to route the call )In To : Person who has been called !In From : Person who was calling !
Of course,
C F ha scritto:
Am I the only one having trouble with this list?
Since the begining of the week I have not been receiving mail from the
list like I used to, is this a gmail problem? or is it subscription
problem? or is something wrong with the list?
anybody else using gmail having any problems?
Hi,
I have good results with the new TDM2400P serie (with the hardware echocan,
of course).
May be you must check one TDM2401E to see if it's ok for you...
Good luck.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la
Dov Bigio ha scritto:
I found the problem.
Master.csv reached 2.0GB and since the moment this happened Asterisk
went crazy!
Since I am using cdr-mysql, how do I disable the use of csvs?
Thank you
Dov
Why don't you simply rotate the logs with logrotate ?
(no, I don't know how to
Title: Message
SIPGetHeader(var=headername)
Olivier
-Message d'origine-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
Jean-Marc SalsaEnvoyé: lundi 13 février 2006
11:40À:
asterisk-users@lists.digium.comObjet: [Asterisk-Users] How to
Get SIP Header
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Iuri Gomes Diniz
Sent: Friday, February 03, 2006 7:55 PM
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
] logger.c: -- Goto (macro-record-enable,s,4)
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Executing
AGI(SIP/100-611e, recordingcheck|20060213-061944|1139829584.46) in
new stack
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck
Feb 13
Hello every one.This is a question done by me, not yet answered. Please, help.I:1. Run install-pdf from linux to support faxes on my asterisk. 2. Made the configurations throuhg AMP in a.Setup-Inbound Routing-(the only route I have)-fax extension-System b. Setup-Inbound Routing-(the
Hello
all!
Does anyone know if there are more
ways to limit a sip users bandwidth usage ? I have seen the bandwidth=
setting in sip.conf, which can be set to low, medium and high and thereby only
allowing certain audio codecs...
In my case I am using video
endpoints and want to say
On Mon, Feb 13, 2006 at 05:25:37AM -0600, yrving rivas wrote:
Hello every one.
This is a question done by me, not yet answered. Please, help.
How about a decent subject for your message?
I:
1. Run install-pdf from linux to support faxes on my asterisk.
Of what software
I am actually now working on massproducing door
openers that will work with asterisk. It will have an
rj45 port and then a port to plug the door opener in
to. Please contact me off list if you are interested.
Dovid
--- Thomas Artner [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
Maybe do a
[EMAIL PROTECTED] ha scritto:
Hi,
I'm stuck on a silly thing. I need to get the billsec CDR value after a
call. But I'm finding its always 0.
Here's my test code:
exten = *244*,1,Dial(Local/[EMAIL PROTECTED]/n,,g)
exten = *244*,n,Noop(after dial duration is ${CDR(duration)} billsec is
Hi list!
How to send a call directly to voicemail recording?
When I put this
exten = 313,n,VoiceMail,u221
Or this
exten = 313,n,VoiceMail,b221
In my dial plan, calling person first hears greeting message (busy or
unviable). I would like to avoid greeting message (I would play something with
Useful discussion on this. There are some other functions in this which need to
be addressed. For example if doing an attended transfer and the recipient phone
number goes to voicemail, you have to wait for the timeout to reconnect to the
original caller - unless someone know differently. There
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMailIt's all in thereOn 2/13/06,
Tomislav Parčina [EMAIL PROTECTED] wrote:
Hi list!How to send a call directly to voicemail recording?When I put thisexten = 313,n,VoiceMail,u221Or thisexten = 313,n,VoiceMail,b221In my dial plan,
Yes - in a traditional PBX environment the transferring station has the
ability to pull the call back by pressing a sequence of keys. In some
PBX's, pressing the transfer key twice, like a double-click of a mouse,
will pull the call back. In some analog environments, pressing the
flash key twice
GotoIf(SIP/100-611e, 0 0?2:4) in new stack
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Goto
(macro-record-enable,s,4)
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Executing
AGI(SIP/100-611e, recordingcheck|20060213-061944|1139829584.46) in
new stack
Feb 13 06:19:44 VERBOSE[5079] logger.c
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Michael Collins
Sent: 13 February 2006 00:58
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-
Commercial Discussion
Subject: RE: [Asterisk-Users] attended call transfer
You may not be able to disable the logs but if you do not care about the
information you cal always link to /dev/null
Ie
ln -s /path/to/log/file /dev/null
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Simone Cittadini
Sent: Monday, February
On 2/13/06, Michael Collins [EMAIL PROTECTED] wrote:
Nik,
Just curious - what is your telco setup? Do you have PRI with the
specified D channels? You need to make sure that your telco is set up
to have the D channels on 16 and 47. When you first start Asterisk, or
when you log on to the
In my expereince a channel bank with a digium single span card and a
Tellabs EC perfomed the best, but is too expensive (it gives you a
minimum of 8 ports). Next to that I use a mediatrix 1204, and compared
to all others I have tried works best. I have tried:
Sipura SPA3000
Digium TDM400 with 4
I would suggest the Pika Technologies Daytona cards. We have been beta-testing
them with Asterisk for a while now and the results have been very, very good.
The only drawback right now is that they only support 1.09. 1.2.x support
should be ready later this quarter. Contact me off list if you
Thanks Tzafrir: I am apologize because of my language problems writing the subject. I am not good at english. So thanks for telling me I am doing it the wrong way, and I will be more carefully next time. Help me if it is possible to you. The Asterisk version is 2.1 wich I downloaded trhough
When Asterisk first starts up, it will attempt to bring up the B
channels on any PRI circuits. If you are using [EMAIL PROTECTED] then you can
log on
to the Asterisk CLI (asterisk -r) and then do stop now to stop
asterisk. Start up Asterisk again by typing asterisk -cvvv at the Linux
command
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all,
I am running into some problems here with PrivacyManager. We used to
use it without any issue, but now there seems to be several problems.
We are currently running Asterisk 1.2.4.
First, it seems that if the user does not press the pound
Hello,
I've problems with following -
- --- ---
PSTN | --- isdn --- | A | - iax2 -- | B |
- --- ---
On [B], there is unconditional call forwarding set back via [A]
(dialparties.agi is used)
Hi
Asterisk died this morning with this message
safe_asterisk: line 83: 6828 Segmentation fault (core dumped)
asterisk ${CLIARGS} ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Any idea what is the problem ?
here is a show channels before the crash
SIP/131-f5ad (None)
Hello,
Is it possible to start an asterisk application
from the command prompt?
This application has to dial to a number.
When the calling party picks up the phone,
the asterisk application had to play certain voicefiles.
Kind Regards,
Arjan Kroon
Mobillion B.V.
You
can do this via the Manager API
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Arjan
KroonSent: Monday, February 13, 2006 10:10 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users]
automatically start application from the
Hi All
I am using RHEL , kernel 2.6.9-5, asterisk
1.2.4 , zaptel 1.2.3 installed , when I give modprobe
for zaptel and ztdummy , I do not any error message
my iax.conf contains the entry for trunking
as
[hoportal]
type=friend
host=192.168.20.32
secret=mysecret
context=local
Hello everyone, I just ran into one big ugly problem: one of the FXO ports
on my TDM400P does something that causes my telco to mark my line as
deffective! Before installing the * I only ran into this situation when I
managed to short the two wires from the telco. Only this time the lines
are not
On Mon, 13 Feb 2006 10:02:28 -0500, Patrick Fortin [EMAIL PROTECTED]
wrote:
Hi
Asterisk died this morning with this message
safe_asterisk: line 83: 6828 Segmentation fault (core dumped)
asterisk ${CLIARGS} ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Any idea what is the problem ?
You
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Mon, 13 Feb 2006, Simone Cittadini wrote:
Dov Bigio ha scritto:
I found the problem.
Master.csv reached 2.0GB and since the moment this happened Asterisk went
crazy!
Since I am using cdr-mysql, how do I disable the use of csvs?
In
Hello There,
Yes I have a tellabs installed, in fact I may have been one of those who helped
you out :)
What I need though is only 4 ports, that's a bit overkill. I also did the spa
and tdm400 with little luck.
Greg
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL
Hi,
I've been fighting with a sip configuration for a few days, and I just
realized why it wasn't working.
In my sip.conf, I have the following
[someprovider]
Bla
Bla
Bla
Bla
And in my extensions.conf file, I have this
Exten = 555-555-,1,Noop(test)
Sure enough, when I dial 555-555-,
Hello,
Could someone tell me how to play an announcement (gsm file) to all of the members of a meetme
conference?
Thank you!
Barry
___
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Asterisk-Users mailing list
To
Hi allI have a problem to register a cisco 7960 to an asterisk 1.2.2I defined in sip.conf the next : ["phonenumber"]type=friendusername="username"secret="password"host=dynamiccontext=workI am trying to catch the register requests with sip debug with no success (empty screen).I
Hi all,
I am running 1.0.10 and calls over a SIP channel more than
often become zombies (call is still there but no more sound. I get the
following:
Unable to parse INFO message from
[EMAIL PROTECTED] Content
And then the call continues but no more sound (my zombie
analogy).
I
Jeremy G. Gault wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all,
I am running into some problems here with PrivacyManager. We used to use it
without any issue, but now there seems to be several problems.
We are currently running Asterisk 1.2.4.
First, it seems that if the
In the server's (or the phone's) cfg file, put in this line:
directory 1: company_directory.cfg
Then add a file of that name to your server. I'm currently using a
basic csv format file that consists of:
Name,1234
Thanks,
Bob McDowell
Original Message-
From: [EMAIL PROTECTED]
On Fri, 2006-02-10 at 16:05 -0600, Matthew Fredrickson wrote:
On Feb 10, 2006, at 1:21 PM, Gerard Saraber wrote:
Found it, going to go test it right now :) thanks!
So far reports have been positive on the echo, but its a slow day ;)
We're using cisco 7960 phones, they're pricy, but they
I likethe 9133ibetter than the Polycom 301 for
a similar price. It is very similar to the Norstar equipment we have
now. The e-mail support has been great for me.
I had some issues getting it set up, and am still a little
concerned about the echo and side tone. I personally can tune them to be
On 2/13/06, Michael Collins [EMAIL PROTECTED] wrote:
When Asterisk first starts up, it will attempt to bring up the B
channels on any PRI circuits. If you are using [EMAIL PROTECTED] then you
can log on
to the Asterisk CLI (asterisk -r) and then do stop now to stop
asterisk. Start up
On 2/13/06, nik600 [EMAIL PROTECTED] wrote:
On 2/13/06, Michael Collins [EMAIL PROTECTED] wrote:
When Asterisk first starts up, it will attempt to bring up the B
channels on any PRI circuits. If you are using [EMAIL PROTECTED] then you
can log on
to the Asterisk CLI (asterisk -r) and
I believe many users would benefit from such an improvement. That's the
kind of annoyance that is usually not present in a decent PBX and that
people have the worst time getting used to (at least in my experience).
l.
On Mon, 13 Feb 2006 14:46:35 +0100, Alex Barnes
[EMAIL PROTECTED] wrote:
Does anyone on the list have a recommendation for a TAPI interface to
Asterisk? I have tried all of the ones that Google produced, but have
still not yet found a solution that I can move into production. My
favorite to date is AstTapi, but it dies after five-or-so calls.
Thank you very much in
We have (had) two identical Asterisk servers for our outbound call
center. Both were running Linux 2.4 kernel, Asterisk 1.0.7, Libpri
1.0.7 and Zaptel 1.2.1. Each server has a TE410P card with two PRIs.
Last week, we upgraded one of them to Asterisk 1.2.4, Zaptel 1.2.3,
Libpri 1.2.2.
The
I tried to define the iax junction via AMP itself, instead that manually.
Doing this work, and the incorrect device 567 is replaced with what I
want to see 567
The only problem is the need to redefine all the iax connection between the
various * boxes, moving them form iax_custom to
the amp
Using some scripts
that have been posted, we have been able to get paging to phones working quite
nicely. However, with a few [EMAIL PROTECTED] 2.5 installs, (Aserisk 1.2.4) the
phones ring but never pick up. Any ideas on why or how to tweak the scripts to
get the phone paging working
On 2/13/06, Bob McDowell [EMAIL PROTECTED] wrote:
Does anyone on the list have a recommendation for a TAPI interface to
Asterisk? I have tried all of the ones that Google produced, but have
still not yet found a solution that I can move into production. My
favorite to date is AstTapi, but
The issue appears to be something on the XP desktop side. I can end-task
and restore TAPI functionality about 75% of the time. Otherwise, a
reboot always clears it up.
I'm unfamiliar with astmanproxy. I'll look it up.
I removed siptapi just after I determined that I couldn't get it
working...
On 2/13/06, Bob McDowell [EMAIL PROTECTED] wrote:
The issue appears to be something on the XP desktop side. I can end-task
and restore TAPI functionality about 75% of the time. Otherwise, a
reboot always clears it up.
I'm unfamiliar with astmanproxy. I'll look it up.
I removed siptapi
Hi all,
Well, i need h323 and asterisk working together.
I have asterisk with oh323 working
I have opengk installed on the same server (working too).
I have a h323 handset (swissvoice ip10s)
The swissvoice register with opengk (don't ask me how)..
I need opengk register with asterisk to have
Do you also have a SIP phone you are dialing from?
This is what I would have setup:
sip.conf:
[sipphone]
Bla
Bla
Bla
context=local-phones
[someprovider]
Bla
bla
bla
context=someprovider-in
extensions.conf
[local-phones]
exten = 55,1,Noop(test)
[someprovider-in]
exten =
I have installed AstManProxy, and it seems a bit better from within
Outlook, but things are the same from my application. The error the
application gives me is 'LINE ERR NOT OWNER' or something similar. I
would have copy-pasted, but I went into a hard lock after closing and
re-opening
My hardware configuration looks like this:
public network ---other PBX ---analog line (ext 100) --- FXO,Asterisk
|
|--- extension
(200)
by calling extension 100 from public network I can call Asterisk. Now I
would like to
Richard Perini wrote:
On Sun, Feb 12, 2006 at 03:35:56PM -0500, John Novack wrote:
That certainly is the way it SHOULD work. Blind and attended transfer
should be able to be initiated the same way. It certainly is the most
efficient logical way. Attended transfer should revert to blind simply
[EMAIL PROTECTED] wrote:
Hello All,
I am trying to figure out which way to go for a quad port fxo solution
with a good echo can on it. My options are the sangoma remora, a
mediatrix fxo, or something similar.
The issue is that I would need a good EC. This would be on about a 9000
foot loop,
I had the same problem when I did an SVN of the latest version of the
1.2 branch this past Friday (2/10) and re-made zaptel, libpri and
asterisk. I don't if someone goofed when updating the supposed stable
Asterisk 1.2 branch or what. But here's what I did to fix it...
I'm using Asterisk at
Hi there,I plan to use Aterisk in our small office. Until now we used a Panasonic D-1232 SuperHybrid System. The figure is representing the future configuration I where thinking about to have in the office.Question 1: We need only 4 lines and I thought to buy a TDM04B or a TDM2401E
Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kurt x
Sent: Friday, February 10, 2006 10:51 AM
To: Asterisk
Subject: RE: [Asterisk-Users] Cisco 2620 as PRI gateway
debug ccsip message
Kurt
___
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling
Sent: 13 February 2006 18:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] attended call transfer
As I understand
If you send it to a different context, you still have to have the
appropriate extension, i.e.:
[test]
Exten = 555-555-,1,NoOp(test)
I've also noticed that with providers that I don't register with, who
just blindly send the call to the same address (i.e., IPKall), context
seems to be
Here's a debug---
Longish, but I'm not sure what info in this might be useful to anyone--
I have a zyxel SIP phone configured as ext '6351' on Asterisk--
I can successfully call the SIP phone from an Sjphone client on my PC
and talk between the two-
so the SIP phone is in fact registered with *
I am trying to figure out which way to go for a quad port fxo solution
with a good echo can on it. My options are the sangoma remora, a
mediatrix fxo, or something similar.
The issue is that I would need a good EC. This would be on about a 9000
foot loop, and the lines don't
Do you also have a SIP phone you are dialing from?
This is what I would have setup:
sip.conf:
[sipphone]
Bla
Bla
Bla
context=local-phones
[someprovider]
Bla
bla
bla
context=someprovider-in
extensions.conf
[local-phones]
exten = 55,1,Noop(test)
On Feb 13, 2006, at 10:20 AM, Eric ManxPower Wieling wrote:
snip
The nearest CO my POTS line goes to is 11 miles away.
snip
i take it you aren't a DSL customer?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing
On Feb 13, 2006, at 2:45 AM, Simone Cittadini wrote:
C F ha scritto:
Am I the only one having trouble with this list?
Since the begining of the week I have not been receiving mail from the
list like I used to, is this a gmail problem? or is it subscription
problem? or is something wrong with
Pfff,
What for an answer :(
I use gmail and have no problems.
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Martin
Joseph
Envoyé : lundi 13 février 2006 20:36
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re:
Sorry for the late response, but the w for wait ONLY works with DTMF.
Not well documented, but asterisk doesn't detect dialtone, therefore it
can start to dial numbers before the CO is ready, and I don't know how
you can wait for a second dialtone if it doesn't even wait for the first
one!.
May be some truth to it though :(
Personally I use gmail, but use a different email address that is
forwarded to my gmail account. With this setup, I haven't had any
issues. I use gmail because it's easily accessible from any PC, and I
like how it groups conversations (probably why you see a
Hi There,
we are developing a dialer application using the java lib
to interface with the asterisk manager protocol. It works
fine so far. The only problem we have is that if we use
the originate command the user is required to pick up
the fone _bevore_ asterisk will originate the call to
the
This can also be done with the use of call
files. Check this out:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
-MC
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arjan Kroon
Sent: Monday, February 13, 2006
7:10 AM
To:
Is it possible to detect if a specific agent is on a call? I can check
${AGENTBYCALLERID_${AGENTID}} to see if they are logged in(ie if set they are),
but I want to be able to detect if they are actually on a call. The
ChanIsAvail() doesn't seem to work for Agent channels. I want to do this
I have experienced similar problems using gmail.
Gmail certainly had some problems with emails from asterisk lists.
I donot know if it was only restricted to asterisk lists.
As not all emails were being delayed (or dropped), some of you might be under
the impression that theres no problem.
Please
[someprovider-in]
exten = s,1,Dial(SIP/sipphone)
That is what I have. Unfortunately, the context=someprovider-in is being
ignored. I am running asterisk-1.2.4...
The local-phone context is working properly though. I can't see why one is
behaving as I expect and the other isn't.
Pessoal,
alguem gostou das TE411P, parece que elas nao fazem o deveriam fazer :-) ?!?!
alguem que usa sangoma com EC poderia nos dar um feedback?
Obrigado!
--
Bruno de Assumpção Loureiro
msn: [EMAIL PROTECTED]
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Reverse your call order.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Arnd
Vehling
Sent: Monday, February 13, 2006 3:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Manager cmd: originate without picking up
Sorry I sent it to the wrong list!!!
Regards.
-- Forwarded message --
From: Bruno de Assumpção Loureiro [EMAIL PROTECTED]
Date: Feb 13, 2006 8:41 PM
Subject: Sagoma w/EC x TE411
To: Asterisk Users Mailing List - Nont -Commercial Discussion
asterisk-users@lists.digium.com
Hello all,
I've started implementing iLBC on some of the ATAs we have floating around
clients' homes, but I'm coming against this error message with most of them:
codec_ilbc.c: Huh? An ilbc frame that isn't a multiple of 50 bytes long
from RTP (38)?
The ATAs in question are various
Found solution:
exten = s,1,Wait(10)
exten = s,2,Answer()
exten = s,3,Wait(5)
exten = s,4,Flash()
exten = s,5,Wait(2)
exten = s,6,SendDTMF(200)
exten = s,7,Wait(5)
exten = s,8,Hangup()
--- Ursprüngliche Nachricht ---
Von: Hans-Juergen Brand [EMAIL PROTECTED]
An:
I am getting messages on the console about
Registered SIP ... expires 60
How do I increase that 60 to 3 minutes???
I have tried in [general] of sip.conf
to set
expirey=300
defaultexpirey=300
nothing seems to affect it.
Thanks,
Jerry
___
Hi Jerry --
Have you tried adjusting the settings in the SIP device itself? That's where
you can adjust how frequently the device will try to register.
Regards,
Mike.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Tuesday, 14 February
We're got a T1 from Sprint that we use for internet. During VIOP calls,
if you download something, the VOIP calls break up.
I found some info at Sprint for adding 'class of service', and I also
have some information on configuring our Cisco routers.
I've read the relevent pages on the
Hi,
I have the folowing setup
SPA-2002 (exten 22) with a Motorola ME4052 dect
|
Asterisk 1.2
|
SPA-2002 (exten 58) with a Motorola ME4052 dect
All connected to a simple switched lan.
When making calls, sometimes the incoming (58) side does not seem to be
capable to send
I am trying to trunk calls from an old televantage system
into asterisk. The version of televantage being used does not support SIP, so
H.323 trunks must be used. Has anyone had any experience with this? Would I
have to use opengk and get both asterisk and televantage to register as
So, I'm noticing that when Asterisk executes an AGI script, that the AGI script
keeps running until the call is complete.
Is there any way to have the script terminate when the call is answered?
Also noticed that when user makes a call to user B, if user B hangs up the
call, then Asterisk
Curious: Why did you need the wait times to be so long - was it because of your
PBX or is that simply what you wanted?
-MC
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans-Juergen
Brand
Sent: Monday, February 13, 2006 2:12 PM
To: Asterisk Users
Hi all,
I was just on the phone with a B2C company in Australia who
are looking to integrate an Asterisk solution with their Salesforce.com CRM
platform.
They are looking for a consultant/team to provide the
following functionality
Complex IVR
Eg can interface via API into
Nik,
I'm not sure that NOP is correct, but I'm in the states so I'll to
defer to someone who knows E1/PRI. When I run zttool I have OK under
the alarms. Is there a way you can call the telco and confirm the
settings? Make sure that framing, coding and D channels are set up on
their end the
Hi Dean,
We at ATP have a range of resellers/integrators on our system to provide
solutions around Australia. Get them to contact us, and we'll put them
in touch with the nearest integrator with the correct skillset.
Cheers,
Paul
www.austechpartnerships.com
t) +61 (0)3 9221 0888
SIP) [EMAIL
Huh? I don't understand.. If the operator can't pick up the call you need
more operators to compensate.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ira
Sent: Sunday, February 12, 2006 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial
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