Hi all :)
Is it possible to send a specific error code (e.g. 500) to a connected
phone?
If AGI can do it that would be great :)
Thanks and kind regards,
Hauke
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On 10/6/05, Angus Comber [EMAIL PROTECTED] wrote:
Your link doesn't seem to work.
For me works ok. Nice job Cameron :)
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Regards
Marek
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On 10/7/2005, Arjan van Eersel [EMAIL PROTECTED] wrote:
Hi all,
I have installed an asterisk server at my office, the server is behind a
firewall. On the firewall I've set NAT a rule for incoming traffic on
port 5060 to be forwarded to the server.
Connecting from home with my sip client
On 10/7/2005, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On 10/7/2005, Arjan van Eersel [EMAIL PROTECTED] wrote:
Hi all,
I have installed an asterisk server at my office, the server is behind a
firewall. On the firewall I've set NAT a rule for incoming traffic on
port 5060 to be forwarded
Hi ryan,
Have u used safe_asterisk which runs asterisk as a dameon. try it
putting in yur startup script.
regards
/Gurmi
On 10/6/05, ryan nalupa [EMAIL PROTECTED] wrote:
hello! i've tried to use asterisk 1.2 beta version and
all installed fine except that when i config asterisk
to run at
Anyone ever experience noise instead of a dial tone. We are using a
Digium TE410P and 2 units of Rhino FXO FXS Channel Bank.
Everything works well when they are freshly turned on or rebooted, but
after a while there's only noise can be heard instead of dial tone.
Does this have anything to do
At 12:05 PM -0700 10/5/05, John Todd wrote:
At 2:43 PM -0700 10/4/05, trixter http://www.0xdecafbad.com wrote:
Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing
on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP
technologies. Does anyone know which
Can you try zyxel. I has graphical interface to do the configuration.
goksie
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy Hamilton
Sent: Friday, October 07, 2005 4:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi
I have a Zyxel 2000W wifi phone, setup is easy and quick to perform. However
I have found the range less that satisfactory. I have a Cisco 1200 AP and
our wireless laptop devices can acccess the network fine, however the Zyxel
is pretty rubbish. For example I can be 5 metres away with only a
On Fri, 2005-10-07 at 01:34 -0700, John Todd wrote:
To answer my own question: no, it doesn't seem like there is anything
Asterisk-specific in the suit. It seems that Sprint is claiming that
they own the rights to pretty much any VoIP technology. Carry on,
everyone; this will be thrown
Hello,
can anybody tell me where to get the latetest SIP Firmware 1.6.2 for the Polycom
phones?
Thanks and regards,
KB
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wireless generally struggles with brick walls.
- Original Message -
From: Matt Love [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, October 07, 2005 9:55 AM
Subject: RE: [Asterisk-Users] WiFi Phones
Hi
I
hi all!
I'm running an Asterisk-box with bristuff-RC8n and 2 HFC-S cards.
I m located in Vienna/Austria. I have the problem that on outgoing
calls i hear my voice as a short echo (about half a second). This
occurs not on every call.
I tried some changes in my zapata.conf, with rxgain and
Is there no way to have asterisk determine its IP either via upnp or else
resolve a dyndns hostname rather then having an entry in the config file?
smime.p7s
Description: S/MIME Cryptographic Signature
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Hi!
I use AAH and have 2 sip peers. First one is working
perfect both ways. Now I have set up another on and it works perfect for
calling out but I get busy when I try to call in. If I use an IP-phone
connected directly to the provider it is no problem. Anything special to think
about
When I used ZyXel 2000W with wrtg54s outdoor router, I got 300 meters reach
though no thick wall is involved.
Goksie
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angus Comber
Sent: Friday, October 07, 2005 10:21 AM
To: [EMAIL PROTECTED]; Asterisk
Hi,
I could be wrong but...have you taken a look at features.conf??
Usually settings like the one you need are stored there.
Giorgio
hak atil wrote:
How can I override *67 to *8? Is there an easy way to do this?
Thank you
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Agreed, but my laptops and PDA's work fine at 25m+.
It could be the particual phone I have, I think I'll box it back up this
weekend and get a replacement set out as it sonly a week old!
Thanks.
Matt
-Original Message-
From: Angus Comber [mailto:[EMAIL PROTECTED]
Sent: 07 October 2005
Andy Hamilton wrote:
Anyone have good words to say about any of the WiFi handsets currently
available?
The UTStarCom F1000 (an 802.11b device) works pretty well. It's about
half the $$$ of a Cisco 7920 (which are also pretty nice), but it
seems like most of the config is done from the keypad.
hi all!
I'm running an Asterisk-box with bristuff-RC8n and 2 HFC-S cards.
I m located in Vienna/Austria. I have the problem that on outgoing
calls i hear my voice as a short echo (about half a second). This
occurs not on every call.
I tried some changes in my zapata.conf, with rxgain and
I've faced a problem, which I can't see any obvious solution for. When
we try to listen a conversation between A and B we can hear only
inaudible noises at the end connected to the chan-spy. This is only a
problem if either A or B, or the spy is softphone (IAX2 based). If any
of the calling
Hi all,
does anybody have $subj apps.
Thanks,
Bob.
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On Fri, Oct 07, 2005 at 12:51:25AM -0500, Carlos Prieto wrote:
Hi everyone.
I've installed Asterisk PBX using apt packages, but i don't have actually
any Digium card, so i want to use ztdummy.
I've tried to modify the Makefiles in the debian source package, i don't get
any error, but
On Thu, Oct 06, 2005 at 09:17:21AM -0600, Cameron Steadman wrote:
I have written a step-by-step setup for installing Asterisk on Debian
using the VIA EPIA M platform. It is oriented to the Linux novice
(myself being one). Feel free to use it :)
What CPU is it, exactly? Could you please
Why bother with packages anyhow? I just installed debian base and did a
cvs get for head, and all good to go.
Besides I found that using packages with asterisk on debian can do odd
things to your custom sound files if you do a remove.
Regards,
Greg
-Original Message-
From: [EMAIL
[EMAIL PROTECTED] wrote:
On 10/5/2005, Shawn Porter [EMAIL PROTECTED] wrote:
samples are at
http://tumtum.no-ip.com/faxes/1128432831.3.tif
http://tumtum.no-ip.com/faxes/853107320051004-150908.tif
Both of these were faxed from a Brother intellifax 750 through a ring-it
single-line
On Fri, Oct 07, 2005 at 07:37:26AM -0400, [EMAIL PROTECTED] wrote:
Why bother with packages anyhow? I just installed debian base and did a
cvs get for head, and all good to go.
And if you have several systems?
Besides I found that using packages with asterisk on debian can do odd
things
Hello.
I am new in the area, and I have a customer asking if asterisk can
connect directly to an S0 interface. Usually I only connect it to an
operator using T0 or T2 interfaces. This customer has a private ISDN
network, and has ISDN telephones. Then he is asking me if asterisk can
connect
On Thu, Oct 06, 2005 at 08:52:39PM -0700, Wiley Siler wrote:
Can I have a TDM400 and a T100P in the same machine? I am using AAH and
trying to combine two boxes.
First off, make sure you load the two modules in the same order each
time. Otherwise the channels will move.
If so, can anyone
Hello all,
I have a problem with overlap dialing and don't know how to get rid of
it.
My setup is: 1 HFC card with bristuff - ZAP/g1 (2B + 1D channels),
SIP phones (I just removed TDM400P with 4 FXS)
I created test extension 222 which goes directly to g1. In Zapata.conf
Two asterisk boxes 150 miles apart, both cvs-head as of this morning
(and since Sept 27th), connected via iax2 with low-utilized ds3 internet,
C7960 calls exten on remote system (also C7960), and call goes to VM.
No other calls in either system (eg, no load).
Both boxes have iax config'ed as:
On Fri, October 7, 2005 4:58, Steve Totaro said:
I bet it's the crack ;)
How could you have a pinout listed first? If the wire on one side of the
cable is going from pin one to pin four on the other side and two is going
to five on the other side then its correct and complete. I never
Hello,
I have 1 x100p and
asterisk/zaptel 1.2.0-beta1 installed on Debian 3.1
The Caller ID doesn't working
I am getiing an error from * that:
Oct 7 16:11:12 NOTICE[30118]: callerid.c:312
callerid_feed: Caller*ID failed checksum
Oct 7 16:11:15 NOTICE[30118]: chan_zap.c:5835
ss_thread: Got
We bought a couple of the UTStarCom phones. They work fine in the
office environment where noise is low, but on our production floor it is
impossible for me to hear what is being said and the person on the other
end of the call also says that they cannot hear a thing from the F1000
when the
Jim,
I have one of the Hitachi WIP-5000 wifi phones. Been using it about 6 months. I've even travelled with it and tried to make use of hotspots here and there. My primary application for the phone was to replace a Panasonic analog corldess phone and Sipura SPA-2000 combination for my home
On Friday 07 October 2005 13:52, Bohuslav Coufal wrote:
Hi all,
does anybody have $subj apps.
Thanks,
Bob.
you can download them from spandsp website
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harry gaillac [EMAIL PROTECTED] wrote:
What do you think of this project www.openpbx.org ?
Something like ser and openser !
Interesting. In their meeting minutes
(http://wiki.openpbx.org/tiki-index.php?page=Meeting+Minutes+10-5-2005)
I see that a BKW was elected to the board. Is this Brian
Further info. The domain is registered to Marc Olivier Chouinard. He
has posted in the dev list.
Doug
--
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com
___
Can they do this? Is this legal?
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug
Meredith
Sent: Friday,
I am using Teliax to terminate my calls, and I have 3 licenses' for
g729 from Digium. show translations verifies that the registration
took place.
When I place a call, having allow=g729 as the only allow option in
iax.conf, I get the following error:
WARNING[361]: chan_iax2.c:6017 socket_read:
gincantalupo [EMAIL PROTECTED] wrote:
why a fork???
I don't know any of the people involved, or what their motivation
might be, but I will make a guess:
Digium's model tends to stifle innovation. Look at eclipse.org for a
much better model. Eclipse is truly open source. IBM's commercial
Roman:
I created two bash
scripts called Mail2Fax and Fax2Mail for use with the asterisk
sever.
They leverage the
app_txfax and app_rxfax scripts, along with ast_fax. They make using these
apps a lot easier, including being able to mail to [EMAIL PROTECTED] for outgoing faxes and then
On Fri, Oct 07, 2005 at 02:02:06PM +1000, Rod Bacon wrote:
Upon closer inspection, I don't think my system ever tries to establish a
zaptel native bridge. Is there somewhere where this function is
enabled/disabled?
Yeah, if you have echocancelwhenbridged or any other options that would
make
On Fri, Oct 07, 2005 at 10:51:45AM -0400, Brian C. Fertig wrote:
Can they do this? Is this legal?
Google fork open source.
--
Mike
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Looks like Terracall
has not only reduced the rates but also reduced their ability to connect the
calls to India.
Today we are not
able to make even one call, but the CDRs are still coming as connected and we
are being charged.
Please note the
request I sent below for the credit.
Rich Adamson wrote:
Two asterisk boxes 150 miles apart, both cvs-head as of this morning
(and since Sept 27th), connected via iax2 with low-utilized ds3 internet,
C7960 calls exten on remote system (also C7960), and call goes to VM.
No other calls in either system (eg, no load).
Both boxes
Have you ever read the GPL?
-bill
On 7-Oct-05, at 10:51 AM, Brian C. Fertig wrote:
Can they do this? Is this legal?
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
-Original Message-
From:
There are people out there who wish to contribute, and not have their work
lost on an individual project website since they do not choose to accept
digium's terms to contribute to asterisk. This gives them an opportunity
to do so, and have their work aggregated with everyone else in the same
Does anyone know how to fix this error message?
Is it a fault with the card?
The symptoms are excessive disk access and then Asterisk stops responding,
a powerdown and restart is generally required to resolve the issue.
I'm running 2.6.13.1 with a P4 processor, Slackware Linux.
--
John
Hi,
In fact, a S0 is like a T0 interface except the fact that it is
internal. Normally, the S0 should be powered by a PBX or something.
So, normally, you should be ablt to connect to it in TE mode.
At our office, I have a PBX (Nortel) with a S0 bus on which I have
connected Asterisk...it
On Fri, October 7, 2005 16:26, John Crowhurst said:
Does anyone know how to fix this error message?
Is it a fault with the card?
The symptoms are excessive disk access and then Asterisk stops responding,
a powerdown and restart is generally required to resolve the issue.
I'm running
You could use a modem/router with a DynDNS server in it, that would take
care of finding the Asterisk address from the outside.
On the Asterisk server you can can (I think by default it is anyway) tell it
to bind to all IP addresses.
-Original Message-
From: [EMAIL PROTECTED]
Brian C. Fertig [EMAIL PROTECTED] wrote:
Further info. The domain is registered to Marc Olivier Chouinard. He
has posted in the dev list.
Can they do this? Is this legal?
Yes - anyone can register a domain name.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/
I'm game for using them /and testing them.
Ben..
Roman:
I created two bash scripts called Mail2Fax and Fax2Mail for use with the
asterisk sever.
They leverage the app_txfax and app_rxfax scripts, along with ast_fax.
They
make using these apps a lot easier, including being able to mail
Jon Pounder [EMAIL PROTECTED] wrote:
There are people out there who wish to contribute, and not have their work
lost on an individual project website since they do not choose to accept
digium's terms to contribute to asterisk. This gives them an opportunity
to do so, and have their work
sigh.. meaning take the fork
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Walsh
Sent: Friday, October 07,
I would be interested as well...
Why not post them somewhere?
Regards,
Marc
[EMAIL PROTECTED] wrote:
I'm game for using them /and testing them.
Ben..
Roman:
I created two bash scripts called Mail2Fax and Fax2Mail for use with the
asterisk sever.
They leverage the app_txfax and
Nice smartass remark... of course anyone can register a domain name.
Is forking asterisk legal? Of course it is! Asterisk is under the GPL,
which means that anyone can fork it at any time for any reason.
Look at this in a positive light... many open source projects have
forked, and the
On Friday 07 October 2005 11:28, Jon Pounder wrote:
contributors more choice. As long as the two streams stay compatible
(which they likely will) it should be better for everyone.
Don't count on it, the rumblings in the IRC channel sound like it will be
totally INcompatible except to pass
sigh.. meaning take the fork
if you want a ford, buy a ford, if you want a gm buy a gm, they are both
cars. if no one wanted a gmc, they would not be around much longer. No one
is going to question your reasons for wanting one or the other, you are
free to choose. There is room for both and if
Hi,
I'm provisioning an office with limited
cabling. I'm looking for a desk based wifi phone. Most of the ones
I've seen are handsets. Any ideas?
Thanks, WILL
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Hey all,
I just tried running a 'make rpm' on a fresh install of Fedora Core 4
and ran into an error near the end of the build process. This is the
output of the build when the error occurs:
done
rm -f /tmp/asterisk/var/lib/asterisk/mohmp3/sample-hold.mp3
mkdir -p
Hi
While loading asterisk I am getting this error can any one guide me resolve
this
[cdr_addon_mysql.so]Ouch ... error while writing audio data: : Broken pipe
Shaikh
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On Friday 07 October 2005 11:28, Jon Pounder wrote:
contributors more choice. As long as the two streams stay compatible
(which they likely will) it should be better for everyone.
Don't count on it, the rumblings in the IRC channel sound like it will be
totally INcompatible except to pass
Two asterisk boxes 150 miles apart, both cvs-head as of this morning
(and since Sept 27th), connected via iax2 with low-utilized ds3 internet,
C7960 calls exten on remote system (also C7960), and call goes to VM.
No other calls in either system (eg, no load).
Both boxes have iax config'ed
Clipcomm has a WIFI extensible SIP desk phone that we have successfully
integrated with Asterisk.
Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
454 Sonwil Drive
Buffalo, NY 14225
+++
tf voice - 800-398-VOIP X22
l voice -
Hi All,
With spandsp.0.0.2 pre20 installed I can't seem to send faxes with
tx_fax over a Zap channel (POTS). rx_fax works just fine so no issues
with libtiff and (presumably) libxml2.
Basically I get 'slow carrier up' and 'slow carrier down' together with
accompanying beeping noises until
Personally, I believe it's a good thing. It gives more choice.
Look at other products: IPCop (Linux based firewall) is a fork derived from
Smoothwall. They made such a nice job that Smoothwall were playing catch-up
with IPCop for quite some time. I don't know the current situation.
GPL allows
Dear Group,
I have been able to configure my Asterisk BOX to receive calls from
Mediatrix 1204.
I'm having problems sending calls out via my Mediatrix unit.
The SIP Invite is sent to the Mediatrix but the Mediatrix unit sends
back a Status : 480 Temporarily Unavailable.
This is my
Is there no way to have asterisk determine its IP either via upnp or else
resolve a dyndns hostname rather then having an entry in the config file?
For SIP you need to change the externip= variable and then sip
reload. If you detect the ip change using a cron script (dyndns.org
has info on
Please send them to me at [EMAIL PROTECTED]
regards,
Rajesh
- Original Message -
From:
Technical Support
To: asterisk-users@lists.digium.com
; 'Roman'
Sent: Friday, October 07, 2005 9:54
AM
Subject: [Asterisk-Users] RE: faxing
to/from asterisk - new scripts
I hope this does take off as I am starting to feel a bit uncomfortable with
the Digium model and where it is headed. Mark Spencer and Digium deserve
full credit for creating this beautiful thing called Asterisk. They did it
knowing full well these sorts of possibilities existed in the future.
Doesn't asterisk cache the IP? So even if your IP changes and
Dynamic DNS has updated your IP to point to the new IP, asterisk will
still see the old IP until you reload asterisk?On 10/7/05, Wilson Pickett [EMAIL PROTECTED] wrote:
Is there no way to have asterisk determine its IP either via upnp
In article [EMAIL PROTECTED],
Doug Meredith [EMAIL PROTECTED] wrote:
Further info. The domain is registered to Marc Olivier Chouinard. He
has posted in the dev list.
Yes, it looks like the main people behind it are bkw, anthm and moc.
They will be a great loss to the Asterisk community if
On 10/7/05, Will Glass-Husain [EMAIL PROTECTED] wrote:
Hi,
I'm provisioning an office with limited cabling. I'm looking for a desk
based wifi phone. Most of the ones I've seen are handsets. Any ideas?
Thanks, WILL
Will,
I don't know of a specific wifi deskphone... but I have run my
I wouldn't think anyone would consider Sprint a dying company. They just
acquired Nextel so they've got money to spend.
Maybe as an ILEC (which they are here in Ohio) they are viewing Vonage
and Voiceglo as a force that needs to be stopped to prevent further
eroding of their POTS network. I know
I've also only heard of the Clipcomm
Along the same lines...
Why doesn't anyone make a wireless ATA? Am I the only one with a need
for such a thing? By the time I plug in a wireless bridge, an ata and a
cordless phone, I need a five outlet powerstrip and shoebox to hide all
the
I've got a Polycom 501 that I run off a Wifi Game Adapter in my home. It works fine.
On 10/7/05, John Reynolds [EMAIL PROTECTED] wrote:
On 10/7/05, Will Glass-Husain [EMAIL PROTECTED] wrote: Hi,
I'm provisioning an office with limited cabling.I'm looking for a desk based wifi phone.Most of the
My setup is: 1 HFC card with bristuff - ZAP/g1 (2B + 1D
channels), SIP phones (I just removed TDM400P with 4 FXS)
I created test extension 222 which goes directly to g1. In
Zapata.conf overlapdial is set to yes.
First I created this extension:
exten = 222,1,Dial(zap/g1,100,tc)
and channel
Hello,
I'm installing asterisk 1.0.9 in a colinux 0.6.2
I have download asterisk with cvs and when i have do the command
make install in /usr/src/asterisk, i have this error:
checking for gcc... gcc
checking whether the C compiler (gcc ) works... no
configure: error: installation or
IMO, there's absolutely nothing wrong with a fork. In fact, were I
someone with some seroius coding skills and/or the resources to make
it happen, I'd have forked the damned thing 2 years ago, and likely
would have been able to migrate it over to a true OSS license (BSD) by
now.
Tss, tss.
Linksys makes the WRT54GP2-NA that has ATA functionality.
Garrett Smith
[EMAIL PROTECTED]
716-250-3408 Direct
716-903-9495 Cell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Garth Summey
Sent: Friday, October 07, 2005 1:45 PM
To: Asterisk Users
Good Afternoon,
The next Asterisk Users Group meeting has been scheduled for tomorrow,
October 8th at 11:30am.
Meetings are held monthly on the second Saturday of each month, excluding
July and December.
!! NEW ADDRESS !!
Sound Choice Communcations has moved to Bloomington Minnesota,
How can they be a great loss if their ideas and work never made it into the
codebase?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Friday, October 07, 2005 12:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: www.openpbx.org
In
try compiling with 586 and change the makefile to disable mmx codes (if
any). I remember to have this working on a few different processors, but
forgot how I did it.
Not necessary to disable mmx codes. The C3 processors have the full mmx
set. They are missing some of the SSE
Or why not create all sound files under /usr/share/asterisk/sounds and
then subdirs from there for your own touched files i.e.
/usr/share/asterisk/sounds/custom ?
--
Michael Coburn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent:
Jean-Michel Hiver wrote:
IMO, there's absolutely nothing wrong with a fork. In fact, were I
someone with some seroius coding skills and/or the resources to make
it happen, I'd have forked the damned thing 2 years ago, and likely
would have been able to migrate it over to a true OSS license
Rod Bacon wrote:
Do the echo cancellation settings in zapata.conf have any effect when
hardware echo cancellation is installed on a 406p/411p?
The only setting that has any effect is enabled/disabled.
How can I tell if the echo is being cancelled by hardware or software?
The software echo
I don't know what to look for in my sip debug logs, can anybody suggest
what sorts of messages my phones might unexpectedly give asterisk
causing it to drop the zap leg?
Mojo with Horan Company, LLC wrote:
Hello - I have 8 polycom 501s all setup great using ulaw. We have put
them through a
With verbose and debug both on 255, here's all I get at the CLI. The X
is during the call, at the instant the Zap leg seems to drop, almost
concurrently with the 'Hungup Zap/1-1'.
-- Executing Macro(SIP/112-a88a, internaldialout|7476011) in new
stack
-- Executing
On Fri, Oct 07, 2005 at 09:15:05AM -0700, Chris Jones wrote:
Hey all,
I just tried running a 'make rpm' on a fresh install of Fedora Core 4
and ran into an error near the end of the build process. This is the
output of the build when the error occurs:
done
rm -f
Hi all,
I just installed an TDM02B. My system is a dell pc with
linux 2.6.12-1.1456_FC4
asterisk-1.2.0-beta1
zaptel-1.2.0-beta1
libpri-1.2.0-beta1
in /etc/zaptel.conf I have (all others are default):
fxsks=3-4 --- I saw light in the ports
channels=1-2
Can you please also send to phpkidathotmail.com.
Rplace the at of course.
Thanks.
TRoy
Rajesh kumar wrote:
Please send them to me at [EMAIL PROTECTED]
regards,
Rajesh
- Original Message -
From: Technical Support
To: asterisk-users@lists.digium.com ; 'Roman'
Sent: Friday,
This post is exactly my problem:
http://lists.digium.com/pipermail/asterisk-users/2005-July/117988.html
Has anybody encountered this and been able to solve it and use g729
successfully? Are there other g729 implementations available as a codec
for asterisk?
Mojo
Mojo with Horan Company,
Whenever we call IBM, the call counter on the phone never starts and in
the CLI the zap channel never gets the answered signal from the PRI.
See below.
-- Executing Dial(SIP/5933-645d, Zap/g1/18004267378) in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called
I am trying to setup Asterisk as the voicemail server for Cisco Call Manager. I
have just about everything working except for the message waiting indicator.
I have the following setup in context [ccm] in my extensions.conf file:
;MWI
exten = _2807XXX,1,SetCallerID(${EXTEN:3})
exten =
Hi there,
We are experiencing an issue on RHEL 4 (2.6.9-22.EL) with our TDM110P -
whenever we enter 'ztcfg -s' to stop the span, the entire system
crashes, requiring a reset. I have seen this
(http://lists.digium.com/pipermail/asterisk-users/2005-June/
112097.html) and thought it might be
Before I go to the trouble myself, has anyone else created a BBEdit
Language Module for the syntax in asterisk config files?
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
smime.p7s
Description:
I would appreciate seeing the scripts as well. Nice job!
Desktophero at gmail.com
Thank you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Troy Swaine
Sent: Friday, October 07, 2005 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Does anyone have the lates firmware for the
AudioCodes MP-104 FXS?
If so, please send me a link or e-mail
directly.
Thanks
Michael
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