On Mon, 07 Mar 2005 21:32:29 -0600, Kristian Kielhofner [EMAIL PROTECTED]
wrote:
The Sipura 841 goes for under $90. It works well, and has a nice web
interface. Once you get more advanced you can use their Sipura Profile
Compiler tools to work with multiple phones (or work through
On Tue, 2005-03-08 at 08:05 +0800, Steve Underwood wrote:
The problem mid call is due to a bug in the wctdm driver. It slips
frames. When that happens, the echo canceller's training is completely
wrong, and the echo comes back. It might even be rather worse than with
no canceller at all.
Hello all,
I am sorry for posting again, but when asked on irc, they told me that the
reason why I get this error is because I cannot even connect to bv. I used
sip show peers and it shows the server is unreachable. I have tried all the
proxy servers and it doesn't seem to work. I thought it
Dear list
I need to find a way to hook up 4 analog phones to *
was wondering if anyone had old hardware not being used and they are
willing to sell
or know of any places to buy that are cheaper then digium :/
Thanks
I need 2 ports but would perfer 4
let me know
Thanks
On Mon, 2005-03-07 at 13:30 -0800, beonice wrote:
--- Colin Anderson [EMAIL PROTECTED]
wrote:
The problem I suspect will arise is the number of
inodes allowed by the file system. I don't know the
exact size of the typical inode-max, but this will
also presumably become an issue when the user
I have A cisco 7960 and couple of Budget Tone's. I'm hearing more echo
on the BT's than on the Cisco, this is with AGGRESSIVE_SUPPRESSOR
defined
and calls coming in via my analouge lines to X101P clones (in the UK).
The Cisco echo/choppiness is not nearly as bad as the BT.
Is it likely some
At 04:43 AM 3/8/2005, you wrote:
it doesnt have to be a card
it can be a device.
long as it has 2-4 ports
Jer
Dear list
I need to find a way to hook up 4 analog phones to *
was wondering if anyone had old hardware not being used and they are
willing to sell
or know of any places to
The style of echo canceller in use in MEC2 cannot cope with both echoed sound
and 'new' sounds coming down the pipe at the same time (double talk).
Accordingly it has to stop certain parts of the processing during the so called
event. Once double talk has been 'detected' the echo can is frozen
This may be of some use to you on your quest:
http://bugs.digium.com/bug_view_page.php?bug_id=0002820
-Original Message-
From: Kris Boutilier
Sent: Tuesday, March 08, 2005 2:01 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
Discussion
Subject: RE:
Hi all,
I've note that the variable DIALEDPEERNUMBER is broken.
Now i want to know if exist another method to retreive the called number on *,
and, if it's possible, an example ;)
Regards.
___
Asterisk-Users mailing list
- Original Message -
From: Dinesh [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, March 08, 2005 3:39 AM
Subject: RE: [Asterisk-Users] BroadVoice configuration changes for
Outbound
Hello all,
I am sorry for
Hello roger,
Yes I can ping sip.broadvoice.com and I can also register to the sip server
at bv, but cannot do sip show peers, when I do, I get a
owl*CLI sip show peers
Name/username HostDyn Nat ACL Mask Port
Status
bv-in
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 =
Would IAX phone work instead??
[EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
03/07/2005 08:31 PM
Please respond to
Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To
Asterisk Users Mailing List - Non-Commercial Discussion
Quoted message 21:00 05/03/07 +0100 , from Wilson Pickett:
Farfon is in Pakistan, not China
Right (got that mixed up)... so 1 in Pakistan, 2 in China...
Quoted message 15:58 05/03/07 -0500, from C F:
Well let's try to figure this out.
1. The biggest telecommunications market in the world (at
- Original Message -
From: Dinesh [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, March 08, 2005 4:51 AM
Subject: RE: [Asterisk-Users] BroadVoice configuration changes for
Outbound
Hello roger,
Yes I can
I manage the PBX system for amedium
sizedcall center. Where all calls are distributed via a few call Queues.
However I am having an issue where reps are being distributed calls
regardlessof wether they are on a call.
I have looked into using SetGroup but I don't think
this works with Call
w fm3 schrieb:
I too am having the same problem with =VS from last night. From my
debugging, * never attempts to start MOH. Anyone else =ound this?
Me too
Music on hold - with SIP handsets at least - stopped working for me with
asterisk 1.0.6 and cvs.
If I downgraded to 1.0.5 works fine,
Hello list,
I am glad to announce that XC-AST version 0.8.0 is out today.
This version introduces an importante new feature: you can now listen -
using your browser - to calls that have been previously monitored. This
way you can click on a call and listen to the actual conversation, and see
Jer wrote:
Dear list
I need to find a way to hook up 4 analog phones to *
was wondering if anyone had old hardware not being used and they are
willing to sell
or know of any places to buy that are cheaper then digium :/
You could always get a couple of SIPURA 2000 (SIP devices that can talk
to
On March 8, 2005 04:32 am, Adam Goryachev wrote:
Is this the same thing that I might be getting on a TE405p? Most of the
time, everything works nicely, but sometimes, things go astray, and echo
comes into the conversation. Also seems to affect faxes with some lines
with errors during the fax.
Hi
I am having problems getting my card to hang up properly when a remote
party hangs up the line.
I know i have to use the busydetect stuff but it doesn't seem to be
working.
It is a BT line and my zapata.conf is as follows:
[channels]
language=en
rxgain=0.0
txgain=0.0
immediate=no
Until recently, I was using Broadvoice for my in/out calling thru
Asterisk. I was extremely pleased to see that Broadvoice was actually
passing the callerid info (number and text) that I had set up on each
device in my SIP.CONF file. I had PSTN users tell me that they were
actually seeing name
Roger,
Same thing.
Dinesh.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roger Hanson
Sent: Tuesday, March 08, 2005 8:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] BroadVoice configuration changes
Would IAX phone work instead??
SIP should work, but it's not easy to configure.
And YES, IAX would definitely work behind NAT. That's the main point
of IAX superiority over SIP
hth
___
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Asterisk-Users@lists.digium.com
Dinesh,
I've not been following your posts that close, but don't forget
that a fair number of changes to the *.conf files don't get used
unless you stop and restart asterisk. A simple reload will not pick
up many of the changes that you've been making.
Rich
Roger,
Hi List,
After some research, it seems the only reasonable thing to do in order
to get SIP phones behind NAT working reasonably well without fiddling
with the DSL router is to have some kind of far end nat traversal mechanism.
Is there any way to do this with open source tools? I've seen
Well this morning I'm running without AGGRESSIVE_SUPPRESSOR but with the MMX optimization. If I have more trouble, I might try the pentium4 instructions flag.
I've searched everywhere and found nothing on the wtcdm driver bug with it dropping frames. Never heard of this until Steve mentioned
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
I use queues here and if you have a multiple presence (i think that's the term, you know where you have call waiting on an extension) phone, it will do this. My polycoms do this and I actually like it because my queues are just single user queues. Try turning multiple presence off and see if
I might be misunderstanding the question but wouldn't ${EXTEN} work?
On Tue, 2005-03-08 at 04:31, Guy Decarpentrie wrote:
Hi all,
I've note that the variable DIALEDPEERNUMBER is broken.
Now i want to know if exist another method to retreive the called number on *,
and, if it's possible,
Hello,
i'm using ser+nathelper+rtpproxy in front of asterisk. It has been
terrific. The only problem i have is with some DSL modems that grab
port 5060 for themselves (why, i don't know, it's very annoying but
easily solvable). Other than that, no issues at all, in the NAT, in
the DMZ, between
use ${EXTEN} or ${MACRO_EXTEN} if in a macro, in some case this
doesn't do what you want (if you used a goto). but in most case this
will work.
On Tue, 8 Mar 2005 11:31:03 +0100, Guy Decarpentrie [EMAIL PROTECTED] wrote:
Hi all,
I've note that the variable DIALEDPEERNUMBER is broken.
Now i
Do you have a SIPDefault.cnf ? if yes check there. The phone is
unprovisioned b/c you didn't give any configuration settings.
On Tue, 08 Mar 2005 08:08:42 +0100, Thomas Trepper
[EMAIL PROTECTED] wrote:
Hi all,
i have a problem with some cisco 7960. Yesterday i did a
firmware-upgrade from
I have added the three lines to the sip.conf file based on the latest
changes
from broadvoice. I can receive incoming calls but cannot place any
outgoing calls.
The error I get is:
*CLI -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569
-- Attempting call on
On March 8, 2005 08:52 am, Dennis Webb wrote:
Well this morning I'm running without AGGRESSIVE_SUPPRESSOR but with the
MMX optimization. If I have more trouble, I might try the pentium4
instructions flag.
Note that it's only a good idea if your system has MMX capabilities (and a
P4),
hello all
i am having a problem in compiling openh323.
[EMAIL PROTECTED] openh323]# ./configure
checking for g++... g++
checking for C++ compiler default output... a.out
checking whether the C++ compiler works... yes
checking whether we are cross compiling... no
checking for suffix of
Hi List, Im having problems compiling chan_misdn:
asterisk:/usr/src/chan_misdn-beta-0.0.3-rc4 # make install
cc -ggdb -Wall -D_GNU_SOURCE -Wno-missing-prototypes
-Wno-missing-declarations -fPIC -I/usr/src/asterisk/include
-DAST_CONFIG_DIR=\/etc/asterisk/\ -I/usr/src/mISDNuser/include
Hi All,
I have a question about call routing. I currently have a phone number
provided by Voicepulse that connects directly to my Asterisk box and
another phone number provided by Verizon that I have Remote Call
Forwarded to the Voicepulse number. What I'm wondering is if the
information
On Mar 7, 2005, at 9:32 PM, Ryan Burke wrote:
Hello everyone, I've been watching this list for a while, but it is
the first time I've posted. I'ved decided to setup a * server for my
house and will need 3 phones (one main, one for my wife, and one for
my office). I was wondering if there was a
On Mar 8, 2005, at 8:15 AM, Adam Robins wrote:
Last week, due to numerous user quality complaints with Broadvoice, I
started a VoicePulse Connect account. I've tried both SIP and IAX2
setups, but can't get VoicePulse to pass the callerid from SIP.CONF in
similar fashion. I can issue a
The IAX protocol gets around all NATn?
I was thinking of using ser. Do you run it on the same box? Can you share
you ser.cfg?
Yair Hakak [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
03/08/2005 09:00 AM
Please respond to
[EMAIL PROTECTED]; Please respond to
Asterisk Users Mailing List -
I have an Asterisk box with TE110P PRI connected in net mode to a PBX.
Both are PRI EuroISDN.
The connection seems to work OK but when calling from Asterisk to the
PBX through an Xten, the Xten client does not get a ringing tone when
the PBX phone rings.
Is it possible to set this up?
Is there
-Original Message-
On Thu, 2005-03-03 at 12:23 -0800, Dustin Moore wrote:
I've been trying to figure out if it's possible to connect
Asterisk to a parent Inter Tel Axxess system through the
MGCP protocol. The archives for this list aren't searchable
and I'm wondering if anyone has a
This is correct behavior. Basically the Queue system will take the next
person in the queue and try to contact every agent. It is the phone's
responsibility to send back 486 Busy here responses. Turn off Call
Waiting.
-Matthew
- Original Message -
From: James Murray [EMAIL PROTECTED]
Yair Hakak wrote:
Hello,
i'm using ser+nathelper+rtpproxy in front of asterisk. It has been
terrific.
This sounds pretty cool... could you share some config files, maybe
stick them on the wiki somewhere? Or if you want you can send them to me
privately.
If I manage to get it to work thanks to
I'm a neewbie in Linux, so please bear with me.
I have a school assignment to make communication between 10 SIP softphones (kphone).
So far I got trouble installing Asterisk. The information in asterisk web site seems to be a bit outdated because it's mentioned only kernel 2.4.
Since Mandrake
On Tue, 8 Mar 2005, Rob Scott wrote:
I have an Asterisk box with TE110P PRI connected in net mode to a PBX.
Both are PRI EuroISDN.
The connection seems to work OK but when calling from Asterisk to the
PBX through an Xten, the Xten client does not get a ringing tone when
the PBX phone
In ReiserFS3, the performance loss for a directory containing 10's of
thousands of files is negligible. I've personally had directories with
70,000+ files in them, and the performance has been stellar. Most
traditional unix file systems break down around 5k-10k files, but I'd
trust ReiserFS
[EMAIL PROTECTED] wrote:
The IAX protocol gets around all NATn?
It's much more NAT friendly than SIP.
You know I can't believe that something that such a new, big standard as
SIP doesn't account for all environments.
Roughly, at the moment, this is the situation *to my knowledge*. If I'm
wrong
Hi
I'm just subscribed to this list because I'd like to try Asterisk
I'd like some soggestions for getting started with it
I tried to compile source tarball on my mandrake 10.1 but I got compilation
error, I think because mdk 10.1 has a too newer gcc compiler
I can use older mdk or red
Here's a good one for the group, I have 2 Ast servers behind a NAT
(Sonicwall :-( ) connecting to the same server outside the NAT. Each of
the 2 boxes behind register to the outside server. What I am wondering
is, would there be a problem if both servers behind the NAT were
listening on port 4569,
On Tue, 8 Mar 2005 07:45:42 -0800 (PST), Victoria Alexandru
[EMAIL PROTECTED] wrote:
I'm a neewbie in Linux, so please bear with me.
I have a school assignment to make communication between 10 SIP softphones
(kphone).
So far I got trouble installing Asterisk. The information in asterisk web
Hi!
Imagine I have a SIP phone named A. I have a target phone named B. And I
have an asterisk server called C.
No I call the target B from my phone A. Is the ip traffic routed
directly from A to B or goes it from A to C to B?
Background for this question: I want to buy a DSL-router with
I'd love to see config also !
Jean-Michel Hiver [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
03/08/2005 10:43 AM
Please respond to
[EMAIL PROTECTED]; Please respond to
Asterisk Users Mailing List - Non-CommercialDiscussion
asterisk-users@lists.digium.com
To
[EMAIL PROTECTED]
cc
[EMAIL
On Tue, 8 Mar 2005 16:55:30 +0100, Luca Bariani
[EMAIL PROTECTED] wrote:
Hi
I'm just subscribed to this list because I'd like to try Asterisk
I'd like some soggestions for getting started with it
I tried to compile source tarball on my mandrake 10.1 but I got compilation
error, I think
Victoria Alexandru wrote:
Since Mandrake cooker (10.2beta3) witch I'm using is using kernel
2.6.10, I wander if I have a chance to install it properly.
All I have are the 3 installation CDs and it looks I'm missing some
packages (bison, and associated -devel, zlib, and associated -devel).
I tihnk you may be risking the ire of the list. :)
My suggestion would be for you to use something like [EMAIL PROTECTED], a
pre-built install for Asterisk that requires minimal work on your part
to get linux up and running. See
http://asteriskathome.sourceforge.net/ for more information.
Luca Bariani wrote:
I tried to compile source tarball on my mandrake 10.1 but I got compilation
error, I think because mdk 10.1 has a too newer gcc compiler
Luca,
I run Mandrake 10.1 Official with current updates and compile without issue.
Doug
___
On Tue, 8 Mar 2005 16:55:30 +0100, Luca Bariani
[EMAIL PROTECTED] wrote:
I tried to compile source tarball on my mandrake 10.1 but I got compilation
error, I think because mdk 10.1 has a too newer gcc compiler
It compiles perfectly on MDK 10.0/10.1/Cooker, if you posted the
compilation
You should consider installing AMP (Asterisk Management Portal.)
They have a full walkthrough on installing it and Asterisk on their
website. - http://amp.coalescentsystems.ca/
We used Redhat Fedora Core 3. The install walkthrough will tell you what
you need to install, dependancies, etc. From
Thanks for the quick response but I'm afraid this is not an option for me...
I could use a live cd with asterisk preinstalled and declare where are my config files, but it seems I have to deal with mdk cooker, not even early releases or debian which I know they have an rpm, or deb.
In fact, at
On Tue, 2005-03-08 at 11:16 -0500, Doug Lytle wrote:
Victoria Alexandru wrote:
Since Mandrake cooker (10.2beta3) witch I'm using is using kernel
2.6.10, I wander if I have a chance to install it properly.
All I have are the 3 installation CDs and it looks I'm missing some
packages
hello
i have tried
http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en
but failed same error while compiling openh323
---
g++: Internal error: Terminated (program cc1plus)
Please submit a full bug report.
See
Phone unprovisioned means just that. You need to set up the config file so
the lines will register with asterisk. If you got to the unprovisioned
part, the phone loaded the os ok. make sure you have the SIPMAC.cnf file
and the SIPDefault.cnf file on your tftp server or manually program the
Hi all,
I am trying to write an application to monitor queues using the
Asterisk Management API.
So far I have had some level of sucess, basically reverse engineering
the protocol and the event messages using ethereal etc.
I know there are a couple of pages on the Wiki that attempt (no
On Tue, Mar 08, 2005 at 08:35:47AM -0800, Victoria Alexandru wrote:
Thanks for the quick response but I'm afraid this is not an option for me...
I could use a live cd with asterisk preinstalled and declare where are my
config files, but it seems I have to deal with mdk cooker, not even early
On Tue, Mar 08, 2005 at 04:55:30PM +0100, Luca Bariani wrote:
Hi
I'm just subscribed to this list because I'd like to try Asterisk
I'd like some soggestions for getting started with it
I tried to compile source tarball on my mandrake 10.1 but I got compilation
error, I think because mdk
The best way to figure out the manager protocols is through looking at the
manager.c source code and trial and error.
Some things just don't behave the way you think they should, some things are
not fully documented and some actions do not work in certain cercumstances
while others will.
And
I have some problems starting asterisk with a hfc card using zaphfc:
[chan_zap.so]Mar 8 17:53:06 WARNING[2447]: loader.c:258 ast_load_resource:
/usr/lib/asterisk/modules/chan_zap.so: undefined symbol:
ast_retrieve_call_to_death
Mar 8 17:53:06 WARNING[2447]: loader.c:391 load_modules: Loading
Better yet, ditch the Mandrake box and try [EMAIL PROTECTED] for you test
machine.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Tuesday, March 08, 2005 10:00 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users]
Hello,
after reloading Asterisk, I always receive
Mar 8 18:07:47 NOTICE[2707]: indications.c:397
ast_unregister_indication_country: Removed default indication country 'us'
I guess it has something to do with a setting on my BT-102 SIP phone
connected. How can I omit this notice message, what
Hello everyone,
what is the secret about using the i extension? I tried
exten = i,1,Playback(invalid)
but nothing happens when I dial a nonassigned number, it is completely
ignored.
Thanks
Florian
___
Asterisk-Users mailing list
On Tue, 8 Mar 2005 14:17:23 -0300
Alejandro G [EMAIL PROTECTED] wrote:
I have 2 asterisk box in different locations. When I
received a call in one
location and want to transfer it to an extension in the
other location the
external call is hanged up when the person who is
transfering the call
Don't you need 1 in front of the number?
Attempting call on SIP/Broadvoice/5068012
It should be Attempting call on SIP/Broadvoice/1(area code)5068012
Try it and see if you can place outgoing.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
On Tue, 2005-03-08 at 18:19 +0100, Florian Effenberger wrote:
Hello everyone,
what is the secret about using the i extension? I tried
exten = i,1,Playback(invalid)
but nothing happens when I dial a nonassigned number, it is completely
ignored.
Do you have wildcarded extensions in the
Umar Sear wrote:
Hi all,
I am trying to write an application to monitor queues using the
Asterisk Management API.
So far I have had some level of sucess, basically reverse engineering
the protocol and the event messages using ethereal etc.
I know there are a couple of pages on the Wiki that
Wierdly that Tellabs still has issues with some calls (particularly with
loud background noise)
but upgrading to a newer 2572 64ms unit pretty much took care of that too.
ah yes the old 2571/2572 trick, where you able to find sources for the 2
port chassis power supply ?
the 2571, 2572 cards are
On Tue, 2005-03-08 at 17:13 +0100, Michael Vogel wrote:
So I want to register the SIP client at the asterisk server that itself
is registered at the different SIP providers.
Does that work the way I want?
It's what people do here all the time. One issue might arise though,
i.e. where your
Hi Everyone.
Some time ago, I was told that it's possible to implement an incoming
Fax server with extensions using only one PSTN line, like this:
PSTN number: 1234567, that's the line connected,
FAX numbers like: 1234567-00, 1234567-01, 1234567-02 and so on are
routed to their respective
Hello,
is it possible with Asterisk to force SIP authentication? Right now, it
seesm that just any SIP client can at least connect to my PBX, which I
don't want. I want users to authenticate with username and password and
otherwise deny them access.
Thanks
Florian
The call is a local call so that should be fine.
This worked in the past and I tried it the other way also with the
same error message about "Failed to authenticate on INVITE".
Thanks
Jerry
Don't you need 1 in front of the number?
Attempting call on
Hi Steven,
Do you have wildcarded extensions in the context or in contexts that are
included?
no, I don't.
Florian
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I'm looking to add a single FXO port to my Asterisk box. It looks like
my options are a Digium Wildcard X100P off eBay for $6.99, or a
Wildcard TDM400P with an FXO Module from Digium for $125.
Can anyone explain the tradeoffs (other than the ability to put 4
FXO/FSO modules on the TDM400P).
Hi all,
i have a problem with some cisco 7960. Yesterday i did a
firmware-upgrade from 3.1 (1.2) with P0S30203.bin as described in the
most documents. Now i get the message Phone unprovisioned and in
TFTP-Log i find the following line:
07.03.2005 19:58 :Timeout error sending P0S3-07-3-00.bin
Andrew Kohlsmith wrote:
Note that it's only a good idea if your system has MMX capabilities (and a
P4), obviously. :-)
While this is being mentioned -- Never do this on a Celeron... sound
quality is bad and there are random crashes if I recall correctly from
my experience. Turning MMX off
???
The only way a SIP client can connect to Asterisk is if there is an
entry defined in sip.conf. That unto itself requires passing the
extension name and the secret which is essentually username/password as
you are requesting.
Google sip.conf at the Wiki.
W
-Original Message-
From:
Port count should be the only net difference.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Spencer
Nassar
Sent: Tuesday, March 08, 2005 10:56 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Wildcard X100P or TDM400P?
I'm looking
Can you can post the relevant information from your sip.conf and
extensions.conf ?
Don't forget to hide password/phone/...
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry
Geis
Sent: Tuesday, March 08, 2005 12:49 PM
Mine is a p4 xeon so I'm good there. MMX didn't help me though. I have had to redefine aggressive at the moment. The updated mec2.h and mec2_const.h files from bug#2820 didn't seem to help either.
I found an article about in the asterisk sources chan_zap.c a READ_SIZE 160 causing echo but I
On Tue, 2005-03-08 at 18:52 +0100, Florian Effenberger wrote:
Hi Steven,
Do you have wildcarded extensions in the context or in contexts that are
included?
no, I don't.
Then next step is to include your config files so we aren't just
guessing at your problem.
--
Steven Critchfield
Hi all,
is possible to make a hotline with a ata with fxo port ?
I have a ata with fxo port with sip and asterisk.
I d'like with when dial to fxo port asterisk call another ata with fxs
and sip.
Is possible, or i have to configure a hotline in the ata with fxo .
Thanks
Sergio Faulhaber
[EMAIL
Hello,
The only way a SIP client can connect to Asterisk is if there is an
entry defined in sip.conf. That unto itself requires passing the
extension name and the secret which is essentually username/password as
you are requesting.
seems my client made a fault when I unchecked authenticate. After
Here is my configs. from a previous post...
Jerry
--
; Broadvoice
register = PHONE at sip.broadvoice.com
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On Tue, 2005-03-08 at 09:56 -0800, Spencer Nassar wrote:
I'm looking to add a single FXO port to my Asterisk box. It looks like
my options are a Digium Wildcard X100P off eBay for $6.99, or a
Wildcard TDM400P with an FXO Module from Digium for $125.
Can anyone explain the tradeoffs (other
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On Tue, 8 Mar 2005, Marco Parmeggiani wrote:
I have some problems starting asterisk with a hfc card using zaphfc:
[chan_zap.so]Mar 8 17:53:06 WARNING[2447]: loader.c:258 ast_load_resource:
/usr/lib/asterisk/modules/chan_zap.so: undefined symbol:
On March 8, 2005 12:56 pm, Trevor Peirce wrote:
Andrew Kohlsmith wrote:
Note that it's only a good idea if your system has MMX capabilities (and a
P4), obviously. :-)
That is unusual; I can run it on celerons without any issue whatsoever.
They're only castrated, not illiterate too. :-)
I am trying to test how the GotoIf and $LEN functions work but am not
succeeding is
this venture. When I dial and access voicemail with an ani of 3000
the gotoif statement does not push the call to s|6. Its goes through
each line( 1,2,3,4,5,6,7) . In additon when I dial with a 10 digit
ani the
Steven is totally right BTW. No support for clones.
However, I will tell you this. I built my first box with a clone so I
could see if I could do all I needed on this system. Then I promptly
purchased my two TDM400s for my 8 POTS lines. Wish it were PRI but
those are the breaks for this
Can you post your dialplan for that extension. Also, NoOp works great for debugging these issues.
On Tue, 2005-03-08 at 12:29, kurt x wrote:
I am trying to test how the GotoIf and $LEN functions work but am not
succeeding is
this venture. When I dial and access voicemail with an ani of
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