exten = 9,1,NoOp(ISDN: Pickup outside line (early B3 connect) for: ${CALLERIDNUM})
exten = 9,2,SetCallerId(${THORCOM_MAIN}) exten = 9,3,Dial(CAPI/g1//b) exten = 9,4,Hangup
use this string with BT
extn = 9,3,Dial(CAPI/g1//bo)
Should provide correct progress
chan_capi registers fine: **
[chan_capi.so] = (Common ISDN API for Asterisk) == This box has 1 capi controller(s). == Reading config for BRI1 -- ast_capi_pvt BRI1-pseudo-D (MSN1,MSN2,capi-in,0,2) (1,4,128)
--
Make sure you have turned off VAD as asterisk does not support Silence supperssion.
Jason
On 9/21/05, Juan Jose Comellas [EMAIL PROTECTED] wrote:
Have you tried upgrading the firmware? I had several problems with theoutbound volume of these phones until I upgraded them.
On Tuesday 20 September
On 9/19/05, Frank Tarczynski [EMAIL PROTECTED] wrote:
I'm new to asterisk and need some help with ideas to handle thisconfiguration question.I am trying to establish a termination point/DID number in another
country.I am currently running Asterisk CVS-HEAD.My foreign provideruses SIP and
But when BT-100 calls 7960 the following is happening:
-- Executing Dial(SIP/3710-8f2b, SIP/1707|15) in new stack
-- Called 1707
-- SIP/1707-e96a is ringing
-- SIP/1707-e96a answered SIP/3710-8f2b
-- Attempting native bridge of SIP/3710-8f2b and SIP/1707-e96a
May 4
On 4/28/05, Eugenio De Vena [EMAIL PROTECTED] wrote:
Oh boy I am getting crazy...
I installed an asterisk with J4BRI ( 3 BRI point to point ) , Snom phones
and everything works fine. Where's
the problem? Well I can not get music on hold.. Well really MusicOnHold
works, works on Queue, works
On 4/26/05, Matthew Boehm [EMAIL PROTECTED] wrote:
Trying to make a call via our PRI: (CVS everything,
CVS-NHEAD-04/23/05-16:08:12)
-- Executing Dial(IAX2/[EMAIL PROTECTED],
Zap/R2/2815699900|30) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called R2/2815699900
On 4/26/05, snacktime [EMAIL PROTECTED] wrote:
On 4/26/05, Sean Kennedy [EMAIL PROTECTED] wrote:
Hi folks,
I'm curious; What does everyone do for failover? I have two servers,
same os/compilation. I designate one the master, the other the slave,
and I rsync the config files once an
On 4/26/05, Jeremy Koski [EMAIL PROTECTED] wrote:
Normally, when you speak into the receiver of a phone, you can hear
yourself in the earpiece at a very low volume. I have a Cisco 7960 phone
that I'm using with asterisk and I don't get that echo back on the
earpiece speaker. I only have
On 4/27/05, Andre Normandin [EMAIL PROTECTED] wrote:
Does anyone know what the [WARNING: . Changethread: Can't change device
'**Unknown**'] line means below..
I just set verbosity to level 5, and noticed that error everytime a
voicemail is left.. Everything seems to work ok, and I have
On 4/26/05, Callum McGillivray [EMAIL PROTECTED] wrote:
My problem is that this installation is most likely to occur prior to the
release of the new card (and definitely prior to it's vigorous testing in
the field).
If anyone can give me ideas at this point it would be appreciated.
There
On 4/26/05, Stefan Gofferje [EMAIL PROTECTED] wrote:
Stefan Helbing schrieb:
Hello,
the incomingmsn line in chan_capi's capi.conf is limited to 80 characters
(AST_MAX_EXTENSION default value).
My problem: I have to include several MSNs but NOT all. The interface is a
30 channel PRI
On 4/26/05, William C. Lohr Jr. [EMAIL PROTECTED] wrote:
Does anyone know if it is possible to resolve an IP from outside a small
LAN. I would like to be able to specify a SIP client that is outside my
office LAN. The problem is that the isp will not provide a static IP that's
affordable. I
On 4/26/05, Nathaniel Angelo A. Torres (247talk) [EMAIL PROTECTED] wrote:
Hi, I just wanted to know if Digium support ETSI ISDN?
Yes
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On 4/26/05, Michael Baird [EMAIL PROTECTED] wrote:
You might look into the Lucent TNT's, they do SIP/MGCP (with the Hash
codes, os 10.1.xx+), and also terminate modem calls. They are cheap
(check ebay, www.qualitek.net) and their are loads of them out there.
One TNT will handle your
On 4/22/05, Stefan Gofferje [EMAIL PROTECTED] wrote:
Hi folks,
I'm using a Fritz!PCI with chan_capi 0.3.5.
I found that chan_capi neither seems to signal Busy or Congestion to
callers from ISDN nor does it seem to set HANGUPCAUSE, CAUSECODE or
DIALSTATUS if an outgoing call fails. There is
On 4/3/05, Ronald Wiplinger [EMAIL PROTECTED] wrote:
You can't see the sweat, but ...
I would like tp post my improvements to ASTCC somewhere, ... but where???
Post them as patches to bugs.digium.com and then they can be
incorperated into the main code.
On 4/7/05, Thomas Andrews [EMAIL PROTECTED] wrote:
On Thu, Apr 07, 2005 at 10:15:09AM +0200, Thomas Andrews wrote:
I have a Fritz! card set up to use capi, however when incoming calls to
the card are answered, asterisk segfaults.
Have you tried a make clean then make install in the
On 4/20/05, Sean A. Newton [EMAIL PROTECTED] wrote:
Do the SetGroup and CheckGroup functions behavior differently in CVS-HEAD
vs CVS v1-0?
When I upgrade to CVS-HEAD my call waiting disable doesn't seem to work,
using:
exten = s,1,SetGroup(SIP${ARG1})
exten = s,2,CheckGroup(1)
exten =
On Mar 31, 2005 3:32 PM, Mimmus [EMAIL PROTECTED] wrote:
Hi,
I'm trying to compile channel_capi with current Asterisk CVS.
Asterisk compiled successfully but channel_capi (patched with all patches
needed, as suggested from some nice people on IRC #Asterisk) compilation
fails with:
On Mar 31, 2005 1:00 PM, Robson Ribeiro [EMAIL PROTECTED] wrote:
Hi,
If I want a user to, while waiting for a transfer after responding to an IVR,
to listen to music instead of a ring sound, what is the change should i do in
extensions.conf? Is it on the IVR menu or on the optional
On Mar 31, 2005 12:31 PM, Marc SCHAEFER [EMAIL PROTECTED] wrote:
Hi,
is there a configuration in iax.conf to specify that if a call goes to
that peer, a second call should not be allowed.
Specifically, I do this:
Dial(IAX2/iaxcomm) # in extensions.conf for a specific extension
in
On Mar 31, 2005 1:05 PM, Muhammad Haris [EMAIL PROTECTED] wrote:
I've setup * with TDM400P w/1 FXS, 1 FXO modules.
I've one analog phone connected to TDM400P FXS module, 1 PSTN line to
one of the FXO module(ZAP) , and IP phone connected to asterisk on
LAN.
The calls between SIPs and zap
On Tue, 22 Mar 2005 10:45:42 -0800, Trevor Peirce [EMAIL PROTECTED] wrote:
Mike Dent wrote:
Hi,
the topic says it all really.
Does the Sipura 3000 detect and report UK clid correctly?
Yes it does
Jason
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do you really have
[specialized]
[specialized]
it is twice try removing one entry
Jason
On Wed, 23 Mar 2005 02:37:42 +, Scheda [EMAIL PROTECTED] wrote:
Hey, I'm currently using the GotoIf application to set it so if
certain caller ID's call my number, it will transfer it to my cell
On Mon, 21 Mar 2005 15:03:14 -0400, Francisco Moreno
[EMAIL PROTECTED] wrote:
Now, when I dial from any of the ext. to '0' It actually matches the
'0', plays the goodbye message, but doesn't hangup but gets directly to
the 'pasvalide' context. Same thing happens when I dial to the ext. 1002
On Thu, 24 Mar 2005 14:19:20 +, Gavin Hamill [EMAIL PROTECTED] wrote:
Hullo :) Can someone help me untangle a bit of a mess?
I'm trying to set up a demo * server to show off how useful it can be to our
business (as an IVR system and VoIP backup if our ISDN30s fail). I've not
been able to
On Fri, 18 Mar 2005 13:57:37 +0100, Reuben Grech
[EMAIL PROTECTED] wrote:
Dear All,
Would like to know what I should do to:: pickup call immediately and
simultaneously Ring a Group, so that caller is listening to message whilst
group phones are ringing and first one to pickup gets the call.
On Fri, 18 Mar 2005 16:59:36 +0100, Reuben Grech
[EMAIL PROTECTED] wrote:
Dear All,
I am listening to blips during conversations when I have an incoming call
from an X100P card. This does not happen on all conversations.
Any clues? :)
Check cat /proc/interrupts make sure the X100P
On Thu, 17 Mar 2005 02:42:27 -0800 (PST), Kamran Ahmad [EMAIL PROTECTED]
wrote:
hi
any one tell me how to make a dialplan
my extensions.conf
exten = _40,1,Dial(OH323/${EXTEN})
i want to dial to 40 number.
could be any number like 923335224005 or
On Tue, 15 Mar 2005 14:13:40 -0500, Kanuri, Seshu (Company IT)
[EMAIL PROTECTED] wrote:
atxfer = *2 ; Attended transfer
Remove attended transfer capability and then you will be able o enter *2XXX
Jason
___
Asterisk-Users mailing
On Sun, 13 Mar 2005 21:49:52 +0200, Dimitris Kounalakis
[EMAIL PROTECTED] wrote:
Hello *Martijn,
Thank you for your response.
*That was my opinion too, it looses the context due to a bug, and can anyone
confirm it also?
But I have no output from the command Show channels, and it happens so
On Fri, 11 Mar 2005 17:45:55 +0200, Calin Serbanescu [EMAIL PROTECTED] wrote:
hello list,
last night i upgraded my asterisk box from 1.0.5 to 1.0.6 and my music
on hold did not work anymore.
Download version 1.0.7 from Cvs this has the fixes in it
Modify the dialplan.xml on your tftp server to this
DIALTEMPLATE
TEMPLATE MATCH=* Timeout=4 User=Phone/
/DIALTEMPLATE
Jason
On Thu, 10 Mar 2005 10:31:34 -, Marshall, Ed
[EMAIL PROTECTED] wrote:
Hi There
I am currently having an issue with a Cisco 7960. The phone is
try with a CVS head before 03/03/05 as the channel structure was
changed then, or get an updated version of asterisk-oh323 if there is
one availiable
Jason
On Thu, 10 Mar 2005 06:25:04 +0330, mohammad [EMAIL PROTECTED] wrote:
Hi;
I use the following asterisk, openh323, pwlib:
asterisk
On Thu, 10 Mar 2005 16:22:39 +0100, Deti Fliegl [EMAIL PROTECTED] wrote:
Hi,
how can I setup asterisk to use the number presentation bits on the isdn
side to suppress the number presentation? We need to transmit the
subscriber number for billing purposes via ISDN whether or not the user
On Tue, 08 Mar 2005 17:34:31 -0500, Jerry Geis [EMAIL PROTECTED] wrote:
SEPDEFAULT.cnf is not required nor recommended. The 7912 only uses the
gkMAC file
and the software version CP7912XXX file
The gk file must be lower case..
This phone 192.168.255.250 is requesting SEPXXX
It is
Could you do something with the h (Calling party Hangup)
eg
exten = h,1,DoSomething
On Sun, 6 Mar 2005 15:00:23 -0500, George Burt [EMAIL PROTECTED] wrote:
I am trying to run a macro at the beginning of call and after the call is
terminated.
exten =
On Tue, 08 Mar 2005 13:04:48 +, Dan Goscomb [EMAIL PROTECTED] wrote:
Hi
I am having problems getting my card to hang up properly when a remote
party hangs up the line.
With BT you do not need to use, Busy detect the power inversions will
disconnect for you however when the far end
Patch your chan_capi with this and you will be able to compile CVS
HEAD http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2
Jason
On Thu, 03 Mar 2005 18:13:19 +0100, Massimo [EMAIL PROTECTED] wrote:
Hi,
I successfully installed asterisk 1.0 with Capi 0.35. In my pbx system I
,
I'm trying to make work chan_capi with last asterisk CVS.
I installed last zaptel,libpri,last cvs ana patched chan_capi 0.35 with the
patch kindly suggested me by Jason Williams:
http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2
First I received error 127 that I resolved
Try using the url
http://ip-of-machine/phpconfig/phpconfig.php
On Thu, 3 Mar 2005 12:45:03 +0300 (EAT), Julius Kidubuka
[EMAIL PROTECTED] wrote:
Hi,
I have just tried to get phpconfig to work but to no avail. In my browser
I type; http://ip-of-machine/phpconfig/ and this returns the
Contexts can be used to partition Asterisk, but the administration is
not multitenanted
On Thu, 3 Mar 2005 10:47:03 +0100, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi all,
Has any one tested or know if Asterisk support multitenant PBX, ie the
Asterisk
support either multiinstances on the
Try adding an exten = h,1,DoSomething
in the context
Jason
On Thu, 03 Mar 2005 14:40:10 +0100, administrator tootai
[EMAIL PROTECTED] wrote:
Hi everybody,
I'm running an IVR menu with different languages setted up by user when
they enter this menu. What I want is when they hangup,
On Mon, 10 Jan 2005 19:38:23 +, John Middleton
[EMAIL PROTECTED] wrote:
Not an enterprise level system, but anyone used the www.intertex.se IX66?
Yes they work great no nat traversal issues,
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On Fri, 03 Dec 2004 18:49:32 -0500, Nick Bachmann [EMAIL PROTECTED] wrote:
Noah Miller wrote:
I've told lots of people about the Flash Operator Panel before, but
I've never actually used it myself. I've got it all set up nicely,
but I can't seem to authenticate to the asterisk
On Thu, 2 Dec 2004 22:30:17 +, Jon Lawrence [EMAIL PROTECTED] wrote:
Hi all,
I'm trying to sort out my dial plan.
What I'm wanting is something like the following - a bit simplified but
hopefully you'll get the idea.
1) match internal extensions: dial them
2) anything else: send out zap
On Wed, 1 Dec 2004 18:44:43 -0500, Christopher Jacob
[EMAIL PROTECTED] wrote:
After it downloads the files, I do a
make clean
make
make install
When I connect to the console on one machine I get...
Connected to Asterisk CVS-v1-0-10/15/04-14:48:10 currently running on bell (
Try
On Sat, 27 Nov 2004 12:17:45 +, Robbie Hughes [EMAIL PROTECTED] wrote:
-- started pbx on channel (callgroup=0)!
== Starting CAPI[contr1/368466]/33 at isdn,368466,1 failed so falling
back to exten 's'
-- Called 101
-- Called 102
-- SIP/101-1b74 is ringing
--
On Tue, 23 Nov 2004 17:14:30 -0700, Chris Modesitt [EMAIL PROTECTED] wrote:
Is there a way to check if an extension exists?
This is the problem I ran into, I have exceeded the number of extensions you
can attempt to match in one pass (1500+ Extensions).
The solution is not to fix
On Tue, 23 Nov 2004 20:53:57 -0700, Ryan Courtnage [EMAIL PROTECTED] wrote:
If someone has both IAX and SIP clients, would you please attempt to
duplicate the below problem? I don't want to submit a bug unless the
problem can be verified.
The SIP client must support attended transfers (ie:
On Tue, 23 Nov 2004 11:11:02 +1100, Paul Hales [EMAIL PROTECTED] wrote:
Also - the mobile phone plans we are using get very expensive after approx
1500 minutes, so we have to make sure that none of the lines go over that!
But the zap/r1 options should be OK for a start at least.
show
On Tue, 23 Nov 2004 13:17:57 +, WipeOut
[EMAIL PROTECTED] wrote:
Please can someone look at my last two posts and try and shed some light
onto why my system is dropping calls..
If I don't get it right we will be forced to drop Asterisk which I
really don't want to do..
I Have not read
On Thu, 18 Nov 2004 18:52:21 -0600, Jeb Campbell [EMAIL PROTECTED] wrote:
I'm replacing a Merlin for a client and they have a PagePal Intercom
that I would like to reuse.
Here is what I know about it:
It has a screw-down wires that goto rj-11 (This was told to me over the
phone) that went
On Sun, 21 Nov 2004 18:54:25 +0500, khurram bhatti
[EMAIL PROTECTED] wrote:
I'm trying to connect * server from diax 0.9.8c client and * outputs this
errors on CLI
Nov 21 18:59:59 NOTICE[7316]: chan_iax2.c:5742 socket_read: Rejected connect
attempt from 192.168.0.4, requested/capability
On Sun, 21 Nov 2004 19:50:36 -, Kevin Brennan
[EMAIL PROTECTED] wrote:
I am planning to configure * box A with PSTN interface to route faxes to *
box B (running spandsp) over TDMoE. I am using 2xGb bonded NIC's for
connection between servers.
Was wondering
- does anybody have experience
On Tue, 16 Nov 2004 11:35:32 -, Victor Alvarez
[EMAIL PROTECTED] wrote:
Hi,
Trying to configure a voicemail system on FreeBSD 4.10 + asterisk 0.9.0, I
found the following problems:
Download the latest asterisk versions from cvs (try a make update in
the asterisk src directory)
On Fri, 12 Nov 2004 15:50:13 -0500 (EST), Doug Eubanks [EMAIL PROTECTED]
wrote:
Can someone tell me why Asterisk is sending 404 instead of passing this call
to the demo? I have replaced the IPs with descriptions
This is the actual asterisk debug,
Non-codec capabilities: us -
On Fri, 12 Nov 2004 18:32:55 -0700, Paul Fielding [EMAIL PROTECTED] wrote:
I want to authenticate to the phone system, then be able to call an
extension or dial an outside line. My preferred method would be to use
DISA, because a) it's non-verbal - ie. it doesn't talk, just provides
On Sun, 14 Nov 2004 16:44:12 -, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Is this quite simple to set up and can I attach asterix to my landline via a
standard modem?
Yes no go to http://www.voip-info.org/wiki-Asterisk
and read learn try and read try agin
Jason
On Mon, 15 Nov 2004 11:25:55 -0500, Chris TenHarmsel [EMAIL PROTECTED] wrote:
I don't believe the phone has the ability to transfer calls, I
remember looking for this and not finding anything.
You need to use # transfer check wiki
Jason
___
On Fri, 12 Nov 2004 13:39:02 +0100, Nicklas Bondesson
[EMAIL PROTECTED] wrote:
I see alot of these messages after the line is hung up. Why is that?
Urgent handler
-- Executing Hangup(SIP/200-9493, ) in new stack
Urgent handler
-- Executing Dial(SIP/200-9493, SIP/[EMAIL
You could try adding the line insecure=very to the relevant section of
the sip.conf this would force asterisk to only validate the IP address
and not the user name (possibly but it is woth a shot)
Jason
On Mon, 8 Nov 2004 10:28:03 -0800, Randy Bush [EMAIL PROTECTED] wrote:
You could maybe
On Mon, 8 Nov 2004 20:19:42 -1000, Richard [EMAIL PROTECTED] wrote:
I have a question here. If both companies use 200 as their extension, how
can * tell which context a received sip call uses?
The received sip call will be placed in the context specified buy its
definintion in sip.conf
Jason
Yes look at Ebay for x100P compatible cards
On Thu, 04 Nov 2004 10:10:50 -0500, Mike Shultz [EMAIL PROTECTED] wrote:
Quick Question that I hope someone can answer. Will Asterisk work with
basic PCI FaxModems instead of those expensive cards listed on the hardware
page?
Could someone please post the url for the conf? also mute your mic so
everyone can hear!!!
IAX2/[EMAIL PROTECTED]/4569
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To UNSUBSCRIBE or update
On Thu, 9 Sep 2004 12:26:56 -0400, Chris HARIGA [EMAIL PROTECTED] wrote:
Hi,
I have Asterisk w/ 192.168.1.1 and I setup IX66 to be 192.168.1.2 (I'm using
pppoe client and dyndns.org on IX66)
I setup on Local DNS Server my * box and after that I was able to register
my phones from the
On Thu, 16 Sep 2004 20:22:57 +0900 (JST), Isamar Maia
[EMAIL PROTECTED] wrote:
Hi FOlks,
I am trying to setup remotely an AudioCodes Mediant 2000 MG Module 2 to
work with Asterisk through SIP or H323.
But since I don't the product manual, it's being a little hard.
Anybody would the manual
On Thu, 26 Aug 2004 09:25:55 -0600, Andrew Elchuk
[EMAIL PROTECTED] wrote:
Is a sound card needed in order to playback some of the asterisk sounds
in /var/lib/asterisk/sounds when dialing out with an X100P? Thanks.
No Sound card is requied
___
On Thu, 26 Aug 2004 17:31:46 +0200, Alessio Focardi
[EMAIL PROTECTED] wrote:
Also dialing out works like a charm, the only problem is that calling
out asterisk is displayed on the called phone instead of the sip address of the
asterisk
box.
In the general section of sip.conf use the
On Fri, 27 Aug 2004 09:50:37 +0100, Jon Fautley [EMAIL PROTECTED] wrote:
Heh, good old BT. I've never tested voice over Business Highway, as
every BT engineer/support/sales person I've spoken to swore blind that
it wouldn't work - and in BT's eyes, if they say it won't work, it's
unsupported,
On Fri, 27 Aug 2004 11:15:07 +0100, David Gurr
[EMAIL PROTECTED] wrote:
What FXO interface methods are folks using successfully in the UK?
Ditch FXO completely and use a BRI Solution much better quality.
or use Digium TDM400P card with two FXO modules, and apply UK CallerID patches,
In my
On Wed, 25 Aug 2004 14:03:36 +0100, Jon Fautley [EMAIL PROTECTED] wrote:
On 25 Aug 2004, at 13:42, Benjamin Johnson wrote:
Thanks for that Jon,
can anyone confirm whether Asterisk can pick up which MSN has been
dialed and route the call depending on this - or does this
functionality
On Wed, 25 Aug 2004 19:38:34 +0100 (BST), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I suspect it's the POTS end since I haven't been able to reproduce it
by dialling IAXComm from a SIP client connected to Asterisk 1, but I can't
confirm it. What would cause the X100P to randomly drop a call
On Tue, 17 Aug 2004 01:16:21 +0200, Patrick [EMAIL PROTECTED] wrote:
On Mon, 2004-08-16 at 22:13, Markus Engelbrecht wrote:
Hello,
so I decided to update to the latest CVS version of asterisk and of
chan_capi.
You are compiling the wrong version of chan_capi to get chan_capi to
work
On Thu, 05 Aug 2004 10:09:37 -0400, Mike Cathey
[EMAIL PROTECTED] wrote:
How did you get CID to work? We have a Definity and both an FXO and PRI
(T100P) link to *. We can't seem to get CID to pass at all. We're
running v9{something} on the Definity.
It won't work on the FXO
However it
On Thu, 5 Aug 2004 16:07:01 +0100, John Howard [EMAIL PROTECTED] wrote:
And my dialstrings look like this:
;Internal lines
exten = 2001,1,Dial(SIP/2001,20,tr)
;outgoing calls
exten = _9XX.,1,Dial(Zap/1/WW${EXTEN:1},60,Tr)
The dial string I am interested in does not seem to be here,
On Mon, 2 Aug 2004 12:32:38 -0400, AJ Grinnell [EMAIL PROTECTED] wrote:
Can someone tell me where I can get just app.c from. Mine somehow got
corrupted, and no updates or anything else will fix it. I just need the one
file from the latest cvs. 8-1-04. Please help
Delete your corrupted app.c
On Mon, 2 Aug 2004 12:54:59 -0700, Alain Bautista
[EMAIL PROTECTED] wrote:
Anyone had experience 'marrying' the two?
We had setup * to front end Artisoft's Televantage.
It works with some issues need to be resolved:
- Inbound calls could not properly handled and routed by Televantage's
Call
On Mon, 02 Aug 2004 20:23:24 -0700, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
* -- SIP -- CISCO -- PRI -- PSTN
The PSTN sees no callerid.
*--- PRI[zaptel]-- PSTN
Callerid is there... which makes me think it's the cisco, not the
PRI/PSTN/telco.
CISCO PRI-- * PRI [zaptel]
Callerid IS
On Tue, 3 Aug 2004 11:40:28 +0200, Maurizio Marini
[EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 02 August 2004 18:39, Deti Fliegl wrote:
Your Extension has to match your MSNs. You have to configure all MSNs
you have in a comma separated list like
On Sun, 1 Aug 2004 14:50:26 -0400 (EDT), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On Sun, 1 Aug 2004, Trevor Peirce wrote:
Hello,
I know not too long ago I saw /something/ _somewhere_ about an
adjustment to call parking that allowed blind transfers from SIP phones
to park a call and
On Mon, 2 Aug 2004 04:50:08 -0400 (EDT), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On Mon, 2 Aug 2004, Jason Williams wrote:
On Sun, 1 Aug 2004 14:50:26 -0400 (EDT), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I use Brian's Valet Parking on our system.
exten = 700,1
On Fri, 30 Jul 2004 08:56:03 -0400, Deon Rodden [EMAIL PROTECTED] wrote:
We put a VWIC and a DSP in a Cisco 1720. The purpose will be for a customer
to use a T1 Crossover cable to connect the 1720 into their existing PBX
system. It'll be a Virtual T1 PRI type of thing. The Cisco 1720 will make
I do that. But when I play MusicOnHold the music is played slowly! I don´t know
why... but is how the bitrate is playing with a different number.
Make sure you are running mpg123 0.59r and no other version
Jason
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I've tried setting nat=yes in places, externip, et al with no success ..
even though the code I was running from back then worked without that.
Some of the options in sip.conf have changed look at the samples in
src/asterisk/configs/sip.conf.samples
Regards
Jason
On Fri, 23 Jul 2004 02:26:26 -0400, Jeremy Kenney [EMAIL PROTECTED] wrote:
Hello I am new to asterisk I want to setup the call queues where it will
ring multiple devices at the same time and send the call to the first one
that is picked up. There doesn't need to be an agent login for this I
- Original Message -
From: Steve McMahon [EMAIL PROTECTED]
Date: Fri, 23 Jul 2004 01:12:26 -0700
Subject: [Asterisk-Users] Cisco 12sp firmware... Anyone got it???
To: [EMAIL PROTECTED]
Looking for firmware (anything) for the 12sp model phones. Anyone got
it drop me a line!
On Fri, 23 Jul 2004 12:00:22 -0400, mattf [EMAIL PROTECTED] wrote:
That means that you need to hit the pound key twice to initiate a
transfer instead of once. Because of our inbound call center we need to do
transfers and we also need to be able to hit the pound key once without
On Wed, 21 Jul 2004 15:43:17 +0200, Maurizio Marini
[EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi
i've installed asterisk by last cvs and i note
res_parking.c
is not anymore there; chan_capi-0.3.4b INSTALL file require:
in /etc/asterisk/modules.conf insert
On Thu, 22 Jul 2004 12:10:25 +0200, Diego Ercolani
[EMAIL PROTECTED] wrote:
Il 10:03, giovedì 22 luglio 2004, Jason Williams ha scritto:
ensure you have the following in the [global] section
[global]
chan_modem.so=yes
chan_capi.so=yes
sorry, why do you need chan_modem? I don't
On Thu, 22 Jul 2004 14:54:29 +0100, Steve Hanselman
[EMAIL PROTECTED] wrote:
Yes, you'd have a dialplan entry that set a value in the database, then
acted upon that.
You'd probably want some nice voice prompts
The system is currently in [Day/Night/Holiday] mode, press 1 to set to day,
2
On Tue, 20 Jul 2004 23:50:05 +0200, Andy Powell
[EMAIL PROTECTED] wrote:
Hi,
I've just (earlier today) updated from CVS so that I can apply the dtmf caller id
patches. Unfortunately this has had an undesired effect.
I'm using * with an IX66 and no issues, with CVS head I suggest you
have a
On Tue, 20 Jul 2004 10:49:51 -0400, Seth Remington
[EMAIL PROTECTED] wrote:
You are probably having a problem with parking being renamed to
features. Try a make clean then a make install. If that doesn't work
then delete the res_parking.so module from /usr/lib/asterisk/modules/.
You may need
On Mon, 12 Jul 2004 11:42:57 -0700, Dameon D. Welch-Abernathy
[EMAIL PROTECTED] wrote:
Through my Asterisk server, I am trying to use IAXTel to dial 800-type
numbers (yes, I know I can do the same thing with FWD and others via
SIP, but I wanted to play with IAX a little). It appears I'm running
On Fri, 16 Jul 2004 01:23:54 +1000, Andrew Yager [EMAIL PROTECTED] wrote:
Does anyone have the call hold feature working? If you do... how did
you make it work? The instructions say to press the left button to
place the call on hold, and the right button to take it off - except
when I am in a
On Mon, 19 Jul 2004 20:13:09 +0800, Nick Cobley [EMAIL PROTECTED] wrote:
Hopefully someone here can save my sanity. I have been trying to solve
this problem for days now, but just cant put my finger on it. Im new to
* so I have probably done something stupid!
Only a config issue I'm sure
On Sun, 11 Jul 2004 23:02:56 +0100, Richard Airlie [EMAIL PROTECTED] wrote:
On Sat, Jul 10, 2004 at 05:55:21PM +0100, Kevin Walsh wrote:
Richard Airlie [EMAIL PROTECTED] wrote:
First things first. Scrap the ports and build from the latest
CVS source. 0.9 is far to old and buggy, and
On Tue, 6 Jul 2004 11:58:58 -0500, Paul Concepcion [EMAIL PROTECTED] wrote:
Loopback should always make your status LEDs glow steady green. If that's not
working then you've got other problems.
It seems I may have those other problems you talked about. I made a
loopback cable and tested
On Tue, 6 Jul 2004 13:37:33 -0500, McInnis, JP [EMAIL PROTECTED] wrote:
We have a Mediatrix 1102 hooked into the network. Both of the attached
analog phones and all of their features work, but in the CLI we keep
getting -- Got SIP response 481 Transaction Does Not Exist back from
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