I have it installed and working 100% on Mandrake 10.1
Maybe missing development libs are the cause.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: 18 February 2005 09:53 AM
To: 'Asterisk Users Mailing List - Non-Commercial
I also tried installing rpm and after running asterisk -vvvc I got this
error:
[codec_ilbc.so] = (iLBC/PCM16 (signed linear) Codec Translator)
Ouch ... error while writing audio data: : Broken pipe
Warning, flexibel rate not heavily tested!
:(
-Original Message-
From: [EMAIL
Thank you so much, it worked!
Yes turn off silence suppression.
xlite - Menu - Advanced - audio settings - Silence Settings - transmite
Silence: (change to yes)
- Original Message -
From: Julius Kidubuka
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 16,
Seems you have to change something in the xuser or simething file.. Just
read it on the wiki, add a ,
Now it compiled without errors... On to test the demo system :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nic le Roux
Sent: Viernes, 18 de
I had similar problems, transferring a call from a queue with # transfer did
not work too.
Solution for me was to update to CVS-HEAD-02/13/05. This fixed a lot other
problems too.
Hope, this helps...
Guido Hecken
Von: Senyo Gualt-Williams [mailto:[EMAIL
On Fri, Feb 18, 2005 at 03:38:28AM +0100, Michael Bielicki wrote:
upgrade to the following wanpipe and also upgrade the firmware o the
crd (it's included in the wanpipe softwaare)
ftp://ftp.sangoma.com/linux/custom/2.3.2/wanpipe-beta5g-2.3.2.tgz
I did it before asking on the list. I have
[EMAIL PROTECTED] wrote:
Kevin P. Fleming wrote:
I have a patch in my local system that allows the canreinvite setting
(which I renamed) to actually be based on IP address masking, so that
Asterisk can make a more intelligent decision, but even that has
problems, because we don't actually
Hi all,
I've been searching the wiki and google for a couple of days
now but cannot find any reference to a timing source on
OpenBSD. I have * CVS-v1-0-02/15/05-21:54:52 (I always do a
cvs -q up -Pd before compiling) running like a charm on
OpenBSD 3.6
Now I want to setup some IAX trunks to work
While on sangoma
We are getting a samngom pri?Is there any driver I need to install,how
does it work,like a Zaptel card.
Any doc
Please Let me know
altus
On Fri, 2005-02-18 at 11:06, Kumak wrote:
On Fri, Feb 18, 2005 at 03:38:28AM +0100, Michael Bielicki wrote:
upgrade to the following
On Thu, Feb 17, 2005 at 04:15:55PM -0800, Senyo Gualt-Williams wrote:
start the MOH. Has anyone else encountered this?
yes exactly the same problem here. I already posted this a while ago but
without getting any response.
Would be really nice if we could fix this.
Stefan
I've got the same problem. It works fine on some of my older asterisk
boxes that haven't been upgraded, CVS-HEAD-12/09/04, but not on the
latest box, CVS-HEAD-01/19/05. I've tried both t, T, and tT no luck, I
checked my features.conf and it has
[featuremap]
blindxfer = #
i do not have
Hi all,
I'm just wondering about these VoIP services -- do you have to sign up one
account -per- client that will be using the service? I've got multiple
extensions behind my Asterisk box, and I want to be able to allow all my staff
to place calls via the provider.
So if I sign up for one
How much can be the load (How much register and
calls Asterisk can Handle simultaneously by asterisk) and what will be the
performance of Asterisk (Call Quality) if all the users are on SIP only and uses
same Codec, I have all three codecs loaded G.711, G.723, G.729) without media
support
I'm not shure, but I think something changed in CVS HEAD concerning the #
transfers...
In older CVS from December 2004 the # transfer had to be terminated by #
to start transferring.
In actual CVS, you have to press # twice, type in the number, wait 2 seconds
and the call get's transferred.
Is
Sorry Newbie asking everyones option.
I am setting up a couple of small asterisk phone systems for my work, I
started using some snom 190 and bt102 sip phones (the bt102 works really
well with iLBC), but the complaint from my workmates is there is no way
to see if other people are on there phone
Hi,
I need to use the trailing 5 digits of a callerid. callerid may be anything
from a length of 4 to 10 digits in this case.
Using this:
---
SubString,cid=${CALLERIDNUM}|-5|5
Works great, BUT shows this message:
The use of Substring application is deprecated. Please use
quote who=beonice
If I understood the little documentation I found on
's', it's supposed to be a catchall for ALL incoming
calls. That's why I assumed it would catch a DID as
well. If that's not the case, it really should be
updated in some meta-doc somewhere. :)
s is the start extension if
Hi all,
is there a way to sense the automated announce messages that are sent by
cell phone operators?
I would like to switch to my own voicemail system if I dial a coworker's
cell ph. number and I am connected to the provider voicemail announce (or
if the cellphone is unavailable without
On Thu, Feb 17, 2005 at 09:44:49AM +0100, Olle E. Johansson wrote:
It is time to check the CVS head (v1.1dev) version of Asterisk now, we
are heading towards code freeze and production of a new stable release.
We do need help testing all new features, finding bugs, reporting them,
fixing
On Wed, Feb 16, 2005 at 04:10:17PM -0500, Sergey Kuznetsov wrote:
They are the same. That's what I've checked first.
Have you restarted Asterisk? Not all changes picked up with a reload,
sometimes you have unload/reload the module or do a full restart for
all changes to take effect...
Hope this
On Thu, Feb 17, 2005 at 06:32:52PM -0800, beonice wrote:
To answer my own question, at least partially, here is
a quote from the Asterisk Configuration chapter in
Paul Mahler's book VoIP Telephony With Asterisk:
Table 1. Reserved Extension Names
On Thu, Feb 17, 2005 at 01:05:30PM +0200, Michael Manousos wrote:
Did you try asterisk-oh323?
http://www.inaccessnetworks.com/projects/asterisk-oh323
Is there any particular reason to prefer oh323 over the builtin h323? I
can't find any feature comparison and I can't have both since they
On Thu, Feb 17, 2005 at 09:07:46PM +, Alistair Cunningham wrote:
Gonzalo,
Yes, pricing would be included, as would minimum call volumes. Providers
could choose not to disclose these, but then they'd be shown at the
bottom of the page.
A feedback system is a good idea; I'll think
Friends,
I'm in trouble, I tried to install de Asterisk, based on the site manual,
into a RedHat 9.0, I followed every step, and it doesn't work.
When I does the libpri make install, the message is:
quote:
[EMAIL PROTECTED] zaptel]# cd ..
[EMAIL PROTECTED] src]# cd libpri/
[EMAIL PROTECTED]
-lzap means it's looking more libzap, presumably the zaptel library.
Have you got it somewhere where the makefile will find it?
Hope this helps,
On Thu, Feb 17, 2005 at 05:54:42PM -0500, Arlen Raasch wrote:
I do 'make pridump' from the libpri source directory and receive the
following:
#
Hello, I have bought a targeta quadbri.
I want to realize a PBX server to send and to receive fax on lines BRI.
I have installed asterisk and the drivers of quadbri (bristuff_0.2.0-RC7b).
I downloaded and installed the module spandsp-0.0.2pre10
When i send a fax from the fax machine to asterisk,
i installed Asterisk on linux FC3 box and i was able
to make third party calls from voipjet. I then
installed an X100P from digitnetworks and i was able
to execute modprobe zaptel and modprobe wcfxo and i
had to add some lines to the file: 50-udev.rules
before i was able to perform ztcfg without
On Thu, Feb 17, 2005 at 10:47:28PM -0800, kolo sos wrote:
is there any version mismatch or path needed to have a
succesful build? i got an error when i done MAKE to
the asterisk-oh323.
Obviously people have successfully built it, people here use it all the
time. Perhaps you can post the actual
I have a question: Why is't possible to see Caller ID
on the analog phones?
If I'm wrong pls tell me how to do to see Caller ID on
analog phones.
Thank you.
mihaid
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
The SNOM 190 Phones are working quite stable with the hint feature.
We have two customers, with at least 12 SNOM 190/SNOM 200 Phones connected
with SIP to Asterisk and our hotline is relativ quiet ;-)
If the SNOM Phones are within your budget, I think they could be a good
choice.
Guido Hecken
is there anyone who tested the
3Com Business Phone 3102
with Asterisk?
Florian
Hecken, Guido schrieb:
The SNOM 190 Phones are working quite stable with the hint feature.
We have two customers, with at least 12 SNOM 190/SNOM 200 Phones connected
with SIP to Asterisk and our hotline is relativ quiet
Stig Andersson wrote:
So, I try
-
SetVar(cid=${CALLERIDNUM:-5:5})
The result is a empty string if CALLERIDNUM is less than 5 digits long,
which is NOT the case of SubString. SubString command returns what remains of
the variable,
that is - if CALLERIDNUM is 4 digits in length, it
Did you use the caller parameter?
Steve
Blas wrote:
Hello, I have bought a targeta quadbri.
I want to realize a PBX server to send and to receive fax on lines BRI.
I have installed asterisk and the drivers of quadbri (bristuff_0.2.0-RC7b).
I downloaded and installed the module spandsp-0.0.2pre10
Hello
It is all very confusing due to little information available :)
I have a w6692 PCI card, so
1) What ports or modes i can use it? Currently i am plugged into a T0
port, can it be used? And what's the difference from S0? Please point
me to some reading full of clues.
2) Due to lack of my
Hi all,
I am trying to configure * to work with SER (Sip Express Router),the configuration that I am trying is as follows.
I have 2 windows machines running X-Lite soft phonesThe * registers with the SER,I want a call from one X-Lite to asterisk(after registration) which is to
be forwarded to
Hi,
I want to install an Asterisk Box in my Network and work with some IP
phones and ISDN phones.
Is this configuration is possible :
-E1AsteriskE1 or T1---channel bankISDN phones
Wich type of channel bank can I use to do this config?
Wich type of ISDN phones can I use?
Thanks
Hello,
I've got the following situation:
- Asterisk1 - SER -- other world
|
|
--Asterisk2 -
In addition i'm doing a sort of vhost on the asterisk machines, so there
Hello!
When the oprator transfers calls to internal extensions to unavailable
or busy extensions, how can I prevent these calls from going to
voicemail, and route them back to the oprator? But other calls, ie
internal between extensions, and calls coming in via DID should get
voicemail if
Hi,
I just read thru the changelog.txt of the current CVS version and what
catched my eye was the following line: 'Adding Q.SIG switchtype option to
chan_zap' .
But there is no sample config in zapata.conf for Q.SIG and no
'feature-list'. Does this exist anywhere or has anyone already has
Wait a second, whats the problem you having?
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I install the Asterisk into a RedHat9, exactly like manual says, and I'm
having the attached error message when try to install libpri.
Please, help on it.
[EMAIL PROTECTED] zaptel]# cd ..
[EMAIL PROTECTED] src]# cd libpri/
[EMAIL PROTECTED] libpri]# make clean; make install
Makefile:93: .depend:
Never mind, i saw it futher down. I guessing that you've plugged a
phone line from your telephone jack to the x100p (on the right side)
then if you've loaded zaptel wcfxo the in you dialplan add something
like this:
exten = 100,1,Dial(Zap/1/any telephone number)
with out the quotes. try that.
Vonage, to my knowledge, does not let you connect your own SIP device
to their service. They provide their own IAD.
As for Broadvoice, I know people that have successfully deployed
asterisk with many people sharing the same account.
- Pedro
On Fri, 18 Feb 2005 17:54:38 +0800, el Flynn [EMAIL
make clean; make install
Shouldn't ist be make clean; make; make install ?
Hope it helps...
Guido Hecken
-Ursprüngliche Nachricht-
Von: Paulo - Ibest [mailto:[EMAIL PROTECTED]
Gesendet: Freitag, 18. Februar 2005 14:29
An: Asterisk Users Mailing List - Non-Commercial Discussion
hello list,
i have problem when i am useing wrapuptime with
agents.conf
my agents.conf looks like this
[agents]
autologoff=15
musiconhold = default
wrapuptime=5
group=1
agent = 1001,4321,Mark Spencer
recordagentcalls=yes
my aim is every call needs have wrapuptime of 5000 ms
but when
Keith Burns wrote:
Hi,
I am looking for SER/Asterisk consultants in Denver, please contact me at
[EMAIL PROTECTED]
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On Fri, 18 Feb 2005, Jeremy SALMON wrote:
I want to install an Asterisk Box in my Network and work with some IP
phones and ISDN phones.
Is this configuration is possible :
-E1AsteriskE1 or T1---channel bankISDN phones
Wich type of channel bank can I use to do this
* Should I mail something to digium? ;)
fax them the agreement from http://www.digium.com/disclaimer.txt
roy
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Chris Blake wrote:
Greetings *`s,
I have a Digium TDM01B card which I want to connect to a standard phone
socket on the wall, for the purposes of testing [EMAIL PROTECTED]
On the 4 pin connector going to the wall socket, I have the wires from a
CAT5 cable inserted as follows :
Brown/White -
On Thu, 2005-02-17 at 23:20 -0800, Spencer Nassar wrote:
Does anyone know if the tutorial materials from Atricon 2004 are
available for download anywhere? I'm particularly interested in
Joachim Vanheuverzwijn's Performance and Scalability tutorial slides
(Asterisk - building your system
On Friday 18 February 2005 13:44, Michael Welter wrote:
Keith Burns wrote:
Hi,
I am looking for SER/Asterisk consultants in Denver, please contact me at
[EMAIL PROTECTED]
[... quoted signature deleted ...]
Hello Keith,
My name is Michael Welter, and I have been installing Asterisk
Ok I know I'm not the only one having echo problem with asterisk but the
weird thing is that when I receive a call from a PSTN line on my TDM04B
card I don't have any echo problem at the beginning of the call then after
a few minutes I start having echo on my side only
(the person calling from
Almost all of those links don't work including all of the audio files.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Sent: Friday, February 18, 2005 8:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Show channels is only going to show you what channels are
actually in use,
not what is configured. Try 'zap show channels'. If that
indicates zap/1
exists, then your issues are likely in the extensions.conf area.
Post the results of:
zap show channels
relavent part of zapata.conf
All you really need is a two
wire/two pin cable with a 6 position modular plug on each end (
incorrectly referenced frequently as an RJxx ) Simply plug into the
desired 8 position connectors, and the PSTN wall connection. If you
are not US, then it's up to you to find the dialtone from the
Greetings,
I looking for Digium TDM400P... it substitute a complete PABX with 6
lines and 6 extensions for traditional telephnes?
Any advice ir link are welcome
Thanks in advace
Pablo Fernandes
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In article [EMAIL PROTECTED],
Patrick [EMAIL PROTECTED] wrote:
On Thu, 2005-02-17 at 23:20 -0800, Spencer Nassar wrote:
Does anyone know if the tutorial materials from Atricon 2004 are
available for download anywhere? I'm particularly interested in
Joachim Vanheuverzwijn's Performance
Well I have 3 Digium TDM04B (4 port FXO) installed in my server. I use
10 channels out of 12. There's 5 PCI slots on my motherboard, currently
they fill the first 3 PCI slots. I can try to move them arround leaving
one free PCI slot between each of them. The motherboard I use is a Tyan
On Fri, 2005-02-18 at 15:50, Michael Welter wrote:
Get a standard RJ14 cable--the kind with a two-pair (four wire)
connector at each end. You can plug the RJ14 into the RJ45 socket on
the TDM card.
Howdy Michael,
Thanks for replying...I took your advice and all is working.
Whaaa !!
I have two home accounts with Vonage and I allow all the family to use
Vonage with there extensions.
David
On Fri, 2005-02-18 at 08:39 -0500, Pedro wrote:
Vonage, to my knowledge, does not let you connect your own SIP device
to their service. They provide their own IAD.
As for Broadvoice,
On Feb 17, 2005, at 2:32 PM, Justin Richards wrote:
I don't do a lot
of faxing, but I would like to know I'm going to receive them when I
do get one..
I think therein lies the key to your problem. If you're not doing a lot
of faxing then its hard to know if the problem is at your end or if its
Folks,
I've tried to find a reference, but I've had no luck, and would
appreciate your thoughts:
I'd like to be able to monitor a telco line for Message Waiting
Notification, however I cannot figure out if this capability is
available.
Detecting either FSK or Stuttered Dial tone would serve,
IAX trunks require that you have a hardware timing source (from a
zaptel interface). I believe you can use the ztdummy driver if you
don't have a zaptel interface.
Mohit.
On Fri, 18 Feb 2005 10:14:51 +0100, Michiel van Baak
[EMAIL PROTECTED] wrote:
Hi all,
I've been searching the wiki and
Did you install the drivers for the x100p (zaptel) first and then
install asterisk. and what version of asterisk you using
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On Fri, 2005-02-18 at 14:33 +, Tony Mountifield wrote:
[snip]
Links to presentations are up at http://www.laimbock.com/asterisk/
Joachim's stuff is at http://www.securax.be/astricon/
The second link doesn't appear to work. :-(
Yes just noticed that too. Hadn't visited those link in a
Everytime that I make a call to a Budgetone 101 phone. I always see the following:
-- Executing Dial("SIP/1001-bac5", "SIP/1000|20|tT") in new stack -- Called 1000 -- Got SIP response 302 "Moved Temporarily" back from 172.22.5.4 -- SIP/1000-465e is busy
I can use X-Lite all the time to make a
If its not on rpmfind.net good luck...
just goto kernel.org and get the tar-ball.
-Matthew
- Original Message -
From: Muhammad Muzzamil Luqman [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, February 18, 2005 1:48 AM
Subject: [Asterisk-Users] any good redhat 9.0 rpm
On Fri, 18 Feb 2005, E rikje wrote:
Hello,
I've got the following situation:
- Asterisk1 - SER -- other world
|
|
--Asterisk2 -
In addition i'm doing a
On Fri, 2005-02-18 at 14:11 +, Mike Wright wrote:
desk*CLI zap show channels
No such command 'zap' (type 'help' for help)
If that is the case you have no zap loaded.
Did you make install in zaptel, then libpri and finally asterisk?
--
Dave Cotton [EMAIL PROTECTED]
With unlimited calling plans you need to read the terms of service.
Sharing the account within a household or business usually fits in with
that. Reselling services in any way is usually prohibited.
Some providers with unlimited plans will allow you to set the outbound
caller ID to any number
There is a page about this on the wiki.
I've heard from real-world sources that you get about 60-70 G729-PSTN calls
on a dual 3.6Ghz Xeon Dell.
Since SER doesn't handle the media at all, its theoretical limit is around
5000.
-Matthew
- Original Message -
From: Ritesh Jalan [EMAIL
So far i've grasped that to use a card in NT mode it should have
layermask=3 as module option. Is it the only thing that sets TE or NT
mode for card? Perhaps there are settings in misdn.conf ? I can only
get the card to work in TE mode and even then when asterisk is ran as
asterisk -vvvgc it exits
What part of please contact me at [EMAIL PROTECTED] did you not
understand?
-Matthew
- Original Message -
From: Michael Welter [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, February 18, 2005 7:44 AM
Subject:
Hello all.
I am trying to get my second x100p card set up and am having
some troubles.
My zaptel.conf reads:
fxsks=1-2
fxoks=3-4
defaultzone=us
loadzone=us
before adding this card my zaptel.conf
read:
fxsks=1
fxoks=2-3
defaultzone=us
loadzone=us
But now that Ive
Title: Asterisk on Solaris 10
Does anyone have experience compiling Asterisk STABLE 1.0.5 on Solaris 10 for x86? I have looked at http://www.voip-info.org/wiki-Asterisk+Solaris+Support but I'm looking for other people's experience in actually using Asterisk under that platform. We only need
Yes. This is my process:
1.- Create a /tmp/sample.call
--
Channel: Zap/G1/X --- Here fax machine number
Application: txfax
Data: /root/fax.tif
--
2.- Shell in a linux terminal:
---
mv
G'Day All;
So I purchased a Cisco 7960 and am now trying to get it configured for
*.
No can do without the variuos files/images through a FTPF server. I
configured the TFTP server on my RHES 3 box, now to get the required
CISCO files.
So I contacted CISCO to purchase the required maintenance
Folks,
In light of all the troubles people report when running more than one
TDM400 card in a system, I wouldn't mind hearing what your solution of
choice would be when having to connect 5 or more analog telco circuits
to an Asterisk.
I'll try and compile the answers together and get them into
Is anyone else seeing any problems with CDR when using MySQL,
specifically dropped legs of the call?
ie:
+-+-++-+
| calldate| disposition | lastapp| channel |
+-+-++-+
|
Hello!
When you say sharing the account do you mean multiple simultaneous
outgoing calls or just whoever picks up the phone and get's a dialtone
can make the call?
-Randy
Paul wrote:
With unlimited calling plans you need to read the terms of service.
Sharing the account within a household or
Has anyone out there successfully set up
their * box to terminate their VONAGE calls?
I (and I am sure lots of others) would
love to hear how you did it.
Id like to be able to get rid of
the extra hardware I have hanging around here and use the ASTERISK machine to
handle the SIP
Hi,
We try to do something likesomone did in
redirectAPI, but not fully success...
This is what we did, Both channel has been setup and
talking...
Action: RedirectChannel:
SIP/210.201.75.100-081b9170ExtraChannel:
SIP/route886x-79cbExten:18Context:sipPriority:1
I have two issue:
1.
On Fri, 18 Feb 2005 14:33:43 + (UTC), Tony Mountifield
Joachim's stuff is at http://www.securax.be/astricon/
The second link doesn't appear to work. :-(
You are looking for http://www.astertest.com actually. Joachim has
started a new site regarding Asterisk performance testing and
On 10:00, Fri 18 Feb 05, Mohit Muthanna wrote:
IAX trunks require that you have a hardware timing source (from a
zaptel interface). I believe you can use the ztdummy driver if you
don't have a zaptel interface.
Mohit.
I see in the readme this needs the Linux kernel sources.
As I am running
You should be able to specify your caller ID in your zapata.conf for the
port corresponding to your analog phone.
I have a question: Why is't possible to see Caller ID
on the analog phones?
If I'm wrong pls tell me how to do to see Caller ID on
analog phones.
Thank you.
mihaid
Hello All
I am looking for a solution that can do this:
1-) Receive incoming fax;
2-) Read content and identify a zone in the fax where there is a hand written
name;
3-) Based on name, query a database;
4-) Act based on the result in the database;
I understand asterisk can receive fax and
Hello,
I'm trying to use Asterisk as a SIP PBX with H.323 trunk connectivity.
Everything works except that calls that comes from the H.323 side do not get
audio both ways.
Since the other way round works fine (calls to H.323 side), I suspect the
problem to be in the way VAD or Silence suppresion
Message: 21
Date: Fri, 18 Feb 2005 09:56:42 -0500
From: Giovanni Powell [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Trying to install X100p
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type:
Hi all
Do someone know about a softphone that can register in 2 or more SIP
servers?
It would be useful for me to have a softphone registered in my company´s SER
and in my nacional SIP server.
I think X-lite can't do it.
Thanks
Joao
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Asterisk-Users
Thanks for the tip! :)
~Senyo
I had similar problems, transferring a call from a queue with # transfer did
not work too.
Solution for me was to update to CVS-HEAD-02/13/05. This fixed a lot other
problems too.
Hope, this helps...
Guido Hecken
Von: Senyo
On Thu, Feb 17, 2005 at 04:15:55PM -0800, Senyo Gualt-Williams wrote:
start the MOH. Has anyone else encountered this?
yes exactly the same problem here. I already posted this a while ago but
without getting any response.
Would be really nice if we could fix this.
Stefan
I believe we
Hello,
I am trying to setup an Asterisk GUI with the help of astman(please visit
http://astman.sourceforge.net/am-user-guide.html).
I have installed astman and currently assessing my GUI using;
http://ipaddress-of-asteriskbox/cgi-perl/am-main.pl
I am trying to get the menu options in my GUI to
1.0.5.16 - the latest version. Michael 'Moose' Dinn [EMAIL PROTECTED] 2/18/2005 8:14:41 AM
What firmware are you running on your 101?On Fri, Feb 18, 2005 at 08:04:51AM -0700, Josh Wilson wrote: Everytime that I make a call to a Budgetone 101 phone. I always see the following: -- Executing
Does anyone know the default EM Wink timings for Nortel DID ports?
The default settings on Asterisk are:
;prewink: Pre-wink time (default 50ms)
;preflash:Pre-flash time (default 50ms)
;wink:Wink time (default 150ms)
;flash: Flash time (default 750ms)
;
Try this site: http://fedoralegacy.org/ they have most of the things there
for RedHat 7.1 on to Fedora Core 1 items.
- Original Message -
From: Muhammad Muzzamil Luqman [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, February 18, 2005 1:48 AM
Subject:
On 15:40, Fri 18 Feb 05, Mike Wright wrote:
Message: 21
Date: Fri, 18 Feb 2005 09:56:42 -0500
From: Giovanni Powell [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Trying to install X100p
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
On Fri, Feb 18, 2005 at 10:29:09AM -0300, Paulo - Ibest wrote:
I install the Asterisk into a RedHat9, exactly like manual says, and I'm
having the attached error message when try to install libpri.
I don't see any errors that should affect it. If you're referring to the
Makefile:93: .depend:
You need to use the caller parameter. Something like:
Channel:Zap/G1/
Application:txfax
Data:/root/fax.tif|caller
might work better.
Regards,
Steve
Blas wrote:
Yes. This is my process:
1.- Create a /tmp/sample.call
--
Channel: Zap/G1/X --- Here fax
Olle E. Johansson wrote:
Actually, we could solve Matthew's problem by checking the IP addresses
against the localnet setting and checking if both phones are on the same
side. If both are within the localnet, we can reinvite. If both are on
public side, we can reinvite. But if one is localnet
On Fri, Feb 18, 2005 at 02:18:37PM +0100, Kurt Bauer wrote:
I just read thru the changelog.txt of the current CVS version and what
catched my eye was the following line: 'Adding Q.SIG switchtype option to
chan_zap' .
But there is no sample config in zapata.conf for Q.SIG and no
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