On Wed, Nov 15, 2006 at 07:41:00PM +0100, Paco Brufal wrote:
Hello,
I have an Asterisk system with kernel 2.6.18.1 and one quadbri. I
have installed the latest bristuff patches (0.3.0-PRE-1s).
The latest is actually 0.3.0-PRE-1v .
The system works
fine, but when I do a reboot, the
Hi
i have to manage a particular implementation for a queue.
If the queue hasn't any members logged in (Agent or members i play a
message and hangup
If the queue has some members logged in i route the call to the user
with fewest calls.
The second can be implemented using the queue strategy,
Hi
actually, for out call, i use :
exten = _0.,1,Dial(SIP/out-l1/${EXTEN:1},50,rt)
exten = _0.,2,Dial(SIP/out-l2/${EXTEN:1},50,rt)
exten = _0.,3,Dial(SIP/out-l3/${EXTEN:1},50,rt)
exten = _0.,4,Hangup
can you say me with this config, if the first user call and use out-l1
the
On 2006-11-15 18:30:38 -0800, Lucas Barbuto [EMAIL PROTECTED] said:
Hi all,
Originally tried to post this without being subscribed, apologies if
the list gets this twice.
One of my users has a problem with many of his calls via my Asterisk™
server. He describes the problem as having the
That's pretty easy and included in the basic * implementation - you tell
the queue not to accept users and play a message after the queue command
terminates.
l.
On Thu, 16 Nov 2006 09:10:46 +0100, nik600 [EMAIL PROTECTED] wrote:
Hi
i have to manage a particular implementation for a
Grandstream phones are cheap because they use cheap stuff to manufacture
them, plus their software/firmware and remote provisioning and configuration
system is very basic. Their firmware was bad until newest version, and still
some features like day light saving doesn't work. Will you be changing
Hi,
Conrad Wood schrieb:
On Wed, 2006-11-15 at 17:50 +0100, Tobias Wolf wrote:
Thx, for you answer ;)
I have some trouble with setting my CallerID if i make an international
Call. No Problems with National Calls, i can set whatever I want. We pay
for this service but our telephone provider
Hi All
I have a TE110P card connected to Alcatel 4400 PBX thru PRI. Calls between SIP
extns are all okay but I cannot make or recieve calls between SIP and PBX. I
get :
WARNING[14759] app_dial.c: Unable to forward voice
in /var/log/asterisk/messages and following output on the CLI:
hi,
exten =6000,1,dial(SIP/6000,15,tr)
exten =6002,1,dial(SIP/6002,15,tr)
exten =6004,1,dial(SIP/6004,15,tr)
exten =6006,1,dial(SIP/6006,15,tr)
exten =6008,1,chanspy(SIP/6006 | wbq)
when i dial 6008 ,it is connected ,but i can't able to hear the voice of
the any one.
when coversation between
On Wed, Nov 15, 2006 at 12:04:39PM -0500, Diego Andres Asenjo G. wrote:
Hi!
I have configured the language support in asterisk to reproduce spanish
prompts. I have lines for it in sip.conf, iax.conf, zapata.conf and
voicemail.conf as shown:
[general]
It should be in the section
Hi,
I'm using spandsp-0.0.3
[http://www.soft-switch.org/downloads/snapshots/spandsp/
spandsp-20061116.tar.gz]
on a bristuffed asterisk (1.2.13)
[http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0-
PRE-1v.tar.gz]
libtiff is at version 3.6.0
Running on: Linux router2 2.6.17-2-686
Asterisk closes the channel properly.
When the channel is closed, the defunct AGI dies too.
-- Original Message ---
From: Andrew Joakimsen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tue, 14 Nov 2006
On 11/16/06 06:06 David Gagnon said the following:
Which version are you using? There was a problem in 1.2.12.1 with the page
application. Update to 1.2.13.
what was the problem ?
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0)
On Thu, 2006-11-16 at 08:29 +1300, Hadley Rich wrote:
On Thursday 16 November 2006 06:44, Conrad Wood wrote:
On Thursday 16 November 2006 06:42, Matthew J. Roth wrote:
As per ManxPower at #asterisk, it is not possible to record a call
dialed from an analog phone connected to the Phone In
Hi,
Zeeshan Zakaria schrieb:
Grandstream phones are cheap because they use cheap stuff to manufacture
them, plus their software/firmware and remote provisioning and
configuration
system is very basic. Their firmware was bad until newest version, and
still
some features like day light saving
I can confirm this behaviour with same spandsp, same app_rxfax and
app_txfax, and plain (NON_BRIstuffed) asterisk 1.2.13 and asterisk 1.2.12.1.
Also latest libmfcr2 from libunicall package for chan_unicall can't compile
with spandsp0.0.3pre24 and newer versions from snapshot dir of
/spandsp/
spandsp-20061116.tar.gz]
on a bristuffed asterisk (1.2.13)
[http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0-
PRE-1v.tar.gz]
libtiff is at version 3.6.0
Running on: Linux router2 2.6.17-2-686 #1 SMP Wed Sep 13 16:34:10 UTC
2006 i686 GNU/Linux
Debian testing distro
This is one long letter about T.38 and Asterisk. I hope it will help me, and
lots of other Asterisk users to understand some T.38 problems with Asterisk.
This is my situation:
I have Panasonic DX600 FAX machine. It's connected to Asterisk 1.2.13 thru ATA
adapter (I have used both, Cisco 186
Hello list,
I am struck with a problem: I have a setup where I accept calls on an FXO
port from the PSTN and redirect them immediately to an FXS port where a
data acquisition appartaus is waiting for data sent in DTMF format. The
problem I am experiencing is that Asterisk will in any case
Check is:
very good
http://www.it4u2.com/asterisk2.htm#SIPmacaddress
http://www.loligo.com/asterisk/cisco/79xx/current/
- Original Message -
From: Edwin Lam [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
On 16 nov 2006, at 12:12, [EMAIL PROTECTED] wrote:
please check if the old spandsp-version is kompletly removed
It is.
do you use the rxfax/txfax version out of the soft-switch/snapshots-
folder
??? if not - try them
From my original msg:
The app_rxfax.c in use is from:
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
...
(c) T.38 is the way to go...
Hi George!
You said that T.38 is the way to go. I have problems with T.38 and I don't know
how to solve them. Maybe you can help me. I often get this message on CLI:
Nov 15
Larry,
There is later firmware 3.1.10 dated March 2006.
I gave up on the SPA3K. I could not solve the echo problems.
Rich Adamson indicated that the SPA3K did not have logic to fall back to
the PSTN on SIP failure, only loss of link to the network. I would have
thought that Sipura could have
On Thu, Nov 16, 2006 at 01:36:50AM +0530, Vicky wrote:
its normal .if there are many calls going . You should worry if your load or
memory usage is very high .
On 16/11/06, Andre Courchesne - Consultant [EMAIL PROTECTED] wrote:
We have 1 server that after a few hours operating has multiple
On nov/16/2006, Tzafrir Cohen wrote:
I'm not sure it was resolved yet. As a workaround, issue 'ztcfg -s'
before removing that module.
Thanks. I will try.
--
Paco Brufal[EMAIL PROTECTED]
ServiTux Servicios Informáticos S.L.
Tel. 966 160 600 / Fax. 966 160 601
On Thu, Nov 16, 2006 at 07:57:12AM +0300, Rogers Ochieng wrote:
Hi,
Am trying to make zaptel with ztdummy uncommented in FC5 but am
getting make error. Has anyone gotten this to work?
What is that error, exactly? Could you provide a log from the build?
make 21 | tee log_file
What
Noah Miller wrote:
I never ran 1.6.6 for any length of time. 1.6.7 and 2.0.1 don't seem
to suffer this issue. 2.0.1 has some buddy watch problems, so you may
not want to use it, but 1.6.7 should be OK.
I've been running 1.6.6 for quite a while, and I have been quite annoyed
by this bug.
On 16 nov 2006, at 12:12, [EMAIL PROTECTED] wrote:
please check if the old spandsp-version is kompletly removed
It is.
do you use the rxfax/txfax version out of the soft-switch/snapshots-
folder
??? if not - try them
From my original msg:
The app_rxfax.c in use is from:
Hello
I have a problem with mapi Originate action, i try to make calls with
it, but it seems the dialing is late.
Sometime the late is remarkeble, 1-2 hour.
I see that manager.c caled ast_pbx_outgoing_exten function, and it call
__ast_request_and_dial, why could the dialing late?
Tomislav Parčina wrote:
Asterisk is connected with my SIP provider.
I'm not planning to use T.38. I'm only trying to make FAX work. To do that I
need to find out who is causing the problem that is giving me this warning
WARNING[2237] chan_sip.c: Unknown SDP media type in offer: image 31358
Please go to bugs.digium.com and file this bug they will difinitely get it
working .
On 16/11/06, Thirumal Saminathan [EMAIL PROTECTED] wrote:
hi,
exten =6000,1,dial(SIP/6000,15,tr)
exten =6002,1,dial(SIP/6002,15,tr)
exten =6004,1,dial(SIP/6004,15,tr)
exten =6006,1,dial(SIP/6006,15,tr)
TU == Tim Uckun [EMAIL PROTECTED] writes:
TU I am seeing the following in my log file (standard trixbox
TU install). One seems to be complaining about an error in the
TU dialplan but it won't tell me what file or what line. The other
TU (maybe related) is complaining about a channel lock.
TU
g729 is not a free codec . YOu have to buy it from digium at rateof $10 per
channel license . If you are just using asterisk and havent bought g729
license then asterisk will just do bridging of g729 and wont edit/transcode
stream .
On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote:
I have
I just installed a Digium TDM04B card. The lines that are plugged into
it do not send caller id. As a result (i believe) of no caller ID, the
name of the fax pdf is .pdf. The subject is Fax from. Windows
clients do not like opening files with just an extension.
Is there a simple way to
I upgraded from Tormenta2 to Sangoma A101. I followed the instructions, and
installation was successful. zttool, ztcfg, all show card is installed
properly. I copied the parameters from my old working zaptel.conf,
zapata.conf and zapata-auto.conf. Verified on Sangoma website that these
files are
You said voxbox is better, but even the link you gave for them didn't work.
I googled, and apparantly links are broken on their website.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
I wanted to say thanks to those who responded to my query. You all gave me
some good ideas to explore that I had not considered before, which is what I
was hoping for. :^)
On Tuesday 14 November 2006 10:50, Henry.L.Coleman wrote:
By the time you purchase PCI cards for you extensions (FSO
On 16 Nov 2006 13:55:52 +0100, Benny Amorsen [EMAIL PROTECTED] wrote:
TU == Tim Uckun [EMAIL PROTECTED] writes:
TU I am seeing the following in my log file (standard trixbox
TU install). One seems to be complaining about an error in the
TU dialplan but it won't tell me what file or what line.
Remember to do a service xinetd restart after changing disabled to
no in /etc/xinetd.d/tftp.
On 11/16/06, JOAO CARLOS MOURA [EMAIL PROTECTED] wrote:
Check is:
very good
http://www.it4u2.com/asterisk2.htm#SIPmacaddress
http://www.loligo.com/asterisk/cisco/79xx/current/
- Original Message
Zeeshan Zakaria wrote:
I upgraded from Tormenta2 to Sangoma A101. I followed the instructions,
and installation was successful. zttool, ztcfg, all show card is
installed properly. I copied the parameters from my old working
zaptel.conf, zapata.conf and zapata-auto.conf. Verified on Sangoma
Hi,
I installed a new PBX to replace my old one with same TDM400 and beronet
BRI card but different mISDN driver. I upgraded from 0.2.1-rc13 to
0.3.1-rc23 (both taken from beronet site).
The problem is after upgrade my Asterisk stopped to detect dtmf on some
calls but the misd.conf file is the
I have an Ubuntu system and went into Synaptic and checked asterisk for
installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc
and got the following output with several errors and notices. Do I need to
do more or are these ok? I expected to have some conf files in
I have call recording enabled for some extensions and they make lot of calls
. I see there are many files in /var/spool/asterisk/monitor but i need to
know which file belongs to which call .. In old version of asterisk this was
available in lastapp field of mysql table . Now lastapp shows
Hi all!
Is it anyone that have set up a solution where the schedule backup are sent
by mail ore FTP automatic?
//Mattias
--
Mattias Andersson
Storskiftesvägen 6
145 60 Norsborg
m. +46-70-799 44 41
h. +46-8-641 38 97
Email: [EMAIL PROTECTED]
You need to make sure that you install the asterisk-config package as well.
--Brian
On 11/16/06, blackwater dev [EMAIL PROTECTED] wrote:
I have an Ubuntu system and went into Synaptic and checked asterisk for
installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc
and got
You need to make sure that you install the asterisk-config package as well.
--Brian
On 11/16/06, blackwater dev [EMAIL PROTECTED] wrote:
I have an Ubuntu system and went into Synaptic and checked asterisk for
installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc
and got
Lacy Moore - Aspendora wrote:
Can anyone point me in the direction of a
WAV or ULAW recording of those names?
http://www.digium.com/en/products/voice/allisonsmith/
Thanks, but I meant a recording that already exists - I thought
since both Steve and Ian already exist in GSM there
Out telco assigned us a number range, say from 0231 - 555 to 0231 -
555 . These are number wich are routed to our asterisk server.
This will make 0231 - 555 my base number, sorry if i have chosen
a name with more an one definition.
On the other hand, i can set whatever number
Hello List -
I've got a question regarding POS terminal transactions (credit card
machines, ATM, etc...).
Currently we setup customers in the following manner:
Customer Location -- Data T1 -- DataCenter - PSTN Termination
We are currently using Mediatrix gear for fax transmissions from the
Ronald Wiplinger wrote:
Tom Lynn wrote:
Ron,
The guy is trying to help you. Go to the link and read it. There is
a feature that you can use to play a recording into the voice
channel. Mine is set so when you press #9, the caller hears the
lots of monkeys recording.
The best part of it
Hi,
I got a svn trunk of zaptel, while doing make clean or make (as root) I get:
make: execvp: build_tools/make_svn_branch_name: Permission denied
I am not sure what the error can be.
thanks for your help.
--
Erick Perez
Folks,
I have a NEC 2400 pbx(non-voip) behind a Cisco 3725, connected via
standard wic-t1 card. The NEC needs to call two different asterisk
servers with 4 digits. I have two way calling working with the one * box,
but the other is perplexing me.
Here's the layout
* -- Cisco
Not exctally Tomislav, i need to get if an outbound calls (over Zap channels)
is ringing (so if is not busy or not aviable for some reason) before that the
operator transfer the calls to normal SIP user.
I would just able to get the ringing state fo the call, if so, i can transfer
the call.
Hi,
We've been using zaphfc single ISDN cards as cheap Zaptel timing
sources for our Asterisk boxes for a long time, and in the asterisk
1.0.x series, had zero problems doing so.
I now have some boxes with Zaptel 1.2.x (with a mixture of 1.0.x
asterisk and 1.2.x asterisk), and this setup
Replying to myself with one further piece of information - In the most
recent log, the otherwise stable florz version of the driver seems to
have died with this message:
Nov 16 15:48:34 pabx kernel: zaphfc[0]: empty HDLC frame received.
Might this mean something to someone?
Thanks again,
Steve
I never said voxbox is better than trixbox.
I said You like trixbox Should try voxbox.
The link is: http://www.easyvoxbox.org/
Trixbox has good and bad points (loads from RPM's)
Voxbox has good and bad points (Loads from source)
I like source better than RPM's -- Thats me, but I
Why are you using VOIP for credit cards? You have the Internet look into
a bank with a credit card gateway.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
Are you outside of the US?
Do you need to call US Toll Free Numbers?
We can help you save money on
I know, that jitterbuffer should be set at receiving side and on
outgoing call leg,
ie. if sipphone calls to asterisk and outgoing to zap chanel, I should
set jitterbuffer on zap channel (to dejjitter audio stream from sipphone)
but what about pure voip situation (i.e. iax-iax, sip-iax,
On Thu, 16 Nov 2006, Tzafrir Cohen wrote:
On Wed, Nov 15, 2006 at 12:04:39PM -0500, Diego Andres Asenjo G. wrote:
Hi!
I have configured the language support in asterisk to reproduce spanish
prompts. I have lines for it in sip.conf, iax.conf, zapata.conf and
voicemail.conf as shown:
El jue, nov 16 de 2006 a las 18:28 +0530, Vicky comentaba:
g729 is not a free codec . YOu have to buy it from digium at rateof $10 per
channel license . If you are just using asterisk and havent bought g729
license then asterisk will just do bridging of g729 and wont edit/transcode
stream .
Conrad Wood schrieb:
You might have more luck asking on a telecom specific list, rather than
here.
Well, thx for your time anyway. I just wanted to double-check that there
aren't any asterisk config faults that i were not aware of.
Tobias Wolf
Time Bandit wrote:
Is it possible to record outbound analog calls using an X100P?
I was asked if I knew how to record all calls for a shop with 4 analog
phones transparently to the end users. I thought Asterisk was a good
fit for this and I envisioned using either Digium TDM400Ps or Sangoma
Thank you for the confirmation and the warning about disk space.
Now I need to decide between the Sangoma A20202 and the Digium TDM2411.
I'm leaning heavily toward the Sangoma card for the following reasons:
- It doesn't require a 12V power connector for the operation of FXS modules.
Maybe,
I have a client who is looking for hosted asterisk in Australia, as far
as I can tell ATP is the only company offering this.
Does anyone else on this list know of someone?
Regards,
Dean Collins
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
Celliax development and download site: www.celliax.org
Celliax is a channel driver for the Asterisk Free PBX that manages GSM
and CDMA cellular phones through an adapter, composed by a datacable
(for commands) and an audiocable (for the voice) interfacing the
computer soundcard.
chan_celliax is
Matthew J. Roth wrote:
Now I need to decide between the Sangoma A20202 and the Digium
TDM2411. I'm leaning heavily toward the Sangoma card for the
following reasons:
- It doesn't require a 12V power connector for the operation of FXS
modules.
It certainly does!
- It is compatible with
Hello -
Because this is the equipment some end users have purchased.
-Thanks,
Chris
On 11/16/06, Al Bochter [EMAIL PROTECTED] wrote:
Why are you using VOIP for credit cards? You have the Internet look into
a bank with a credit card gateway.
Best regards,
Al Bochter
Bochter Services
El jue, nov 16 de 2006 a las 11:35 -0600, Victor Toofic comentaba:
El jue, nov 16 de 2006 a las 18:28 +0530, Vicky comentaba:
g729 is not a free codec . YOu have to buy it from digium at rateof $10 per
channel license . If you are just using asterisk and havent bought g729
license then
So I'm all excited, ready to install Trixbox at home. Purchased my X100p card
installed in a computer. I run Trixbox setup and boom I get this error message
FXO PCI Master abort It repaets across the screen and I have to reboot. When
I reboot the system hangs at adding hardware. Loading wcfxo
Pre 1.2.8, chan_sip would try to get the lock indefinitely, and would
basically hang the thread until the lock was fetched. Since 1.2.8, it
gives up after 100 tries and logs that message.
Of course that does not explain why the lock is failing...
Judging by the lack of response here it seems
When i start asterisk : asterisk -vvvc
[app_curl.so]Nov 16 15:23:57 WARNING[4469]: loader.c:325 __load_resource:
/usr/lib/libidn.so.11: undefined symbol: stringprep_rfc3454_A_1
Nov 16 15:23:57 WARNING[4469]: loader.c:554 load_modules: Loading module
app_curl.so failed!
Do you have any insght ?
Tim Uckun wrote:
Judging by the lack of response here it seems like this is broken and
nobody knows how to fix it.
98% of the people here don't use Trixbox.
Doug
-- Ben Franklin quote: Those who would give up Essential Liberty to
purchase a little Temporary Safety, deserve neither Liberty
Deadlocks are not a config or Trixbox issue.
Doug Lytle wrote:
Tim Uckun wrote:
Judging by the lack of response here it seems like this is broken and
nobody knows how to fix it.
98% of the people here don't use Trixbox.
___
--Bandwidth and
My company has recently purchased AgileBill/AgileVoice and have had
numerous problems getting it up and running. We submitted support
tickets within our 30 day trial period, of which only a few were
resolved. As of right now, their software is crashing when it attempts
to provision new
I have tried the configuration exactly the same of yours and it's still
not working. What could be wrong with my installation? I first of all
tried the packages in Debian stable, this didn't work so I compiled from
source but still the problem occurs.
Thanks
-Original Message-
From:
I just downloaded and installed asterisk-1.2.13
Reading http://www.voip-info.org/wiki/view/Asterisk+AEL2 says I should
be using AEL2, but what's downloaded is pbx_ael not pbx_ael2 as expected.
Where and how do I get current release of AEL2 - Is there some 'How To'
somewhere?
TIA
Bart
Charlie Grosvenor wrote:
I have tried the configuration exactly the same of yours and it's still
not working. What could be wrong with my installation? I first of all
Show us what being displayed on the Asterisk console, also show the
output from sip show peers.
Doug
-- Ben Franklin
Trixbox has the same asterisk core... As asterisk... As EasyVoxBox...
To back up my point I can download the backup file from one and install
them on the others
The ONLY thing is FreePBX I must play with the conf for that.
So what is your point ??
Please do tell me what your point of
AEL2 is will be in 1.4
AEL2 is not in 1.2.x
The Wiki was wrong.
Bart Fisher wrote:
I just downloaded and installed asterisk-1.2.13
Reading http://www.voip-info.org/wiki/view/Asterisk+AEL2 says I should
be using AEL2, but what's downloaded is pbx_ael not pbx_ael2 as expected.
Where and how
Al Bochter wrote:
Judging by the lack of response here it seems like this is broken and
nobody knows how to fix it.
98% of the people here don't use Trixbox.
The response was to the statement above. Judging by the lack of response.
I'm sure I'm not the only one, that when seeing the
Dean,
I know Qtec definately do, however their offering is pretty much focused
only on businesses and they offer their service only via their private IP
network - not via the Internet.
http://www.qtec.com.au
-- Eric
On 11/17/06, Dean Collins [EMAIL PROTECTED] wrote:
I have a client who
Chris:
We were evaluating AgileVoice currently, could you please elaborte on your
problems? Did they not do the instattion for you?
Regards,
Andrew
On 11/16/06, Chris Mazuc [EMAIL PROTECTED] wrote:
My company has recently purchased AgileBill/AgileVoice and have had
numerous problems
And if the provider offered it, his asterisk would not support it.
On 11/16/06, Doug Lytle [EMAIL PROTECTED] wrote:
Tomislav Parčina wrote:
Asterisk is connected with my SIP provider.
I'm not planning to use T.38. I'm only trying to make FAX work. To do
that I need to find out who is
Hi,
Anyone here has any experience with the Nokia E70 and
asterisk ?
I read on the nokia website this phone is capable of talking
SIP and do Presence based on SIP/SIMPLE.
Please share your experience, I'm thinking of getting one
but want to be sure I can use it with * before I do.
Thnx.
--
First look if you have the libidn.so.11 library. if you don't
then install it, otherwise you can simply copy-paste it into the /usr/lib
folder where Asterisk is looking for it or make a symbolic link to it.
Alyed
Return-Path: [EMAIL
Andrew Joakimsen wrote:
Chris:
We were evaluating AgileVoice currently, could you please elaborte on
your problems? Did they not do the instattion for you?
They did the installation.
I'm going to be very careful with my wording here, but if you are
currently evaluating their software, I
Michiel van Baak ha scritto:
Hi,
Anyone here has any experience with the Nokia E70 and
asterisk ?
I read on the nokia website this phone is capable of talking
SIP and do Presence based on SIP/SIMPLE.
Please share your experience, I'm thinking of getting one
but want to be sure I can use it
Asterisk 1.2.12.1
The * key and the 0 key do not seem to be detected in my dialplan. I am
using a and o to detect them.
It simply falls thru to my i where it says, I am sorry that is not a valid
extension.
joe a.
___
--Bandwidth and Colocation
That's pretty easy and included in the basic * implementation - you tell
the queue not to accept users and play a message after the queue command
terminates.
l.
Don't forget to analyze the QUEUESTATUS variable. :)
--
Sign my PETITION.
___
I have no idea how that entire message body ended up the Subject *blush*
--
Lucas
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On 16/11/2006, at 7:49 PM, Martin Joseph wrote:
If the softphone has an auto-gain adjust, it's probably the cause
of this?
Worth looking into, thanks Marty.
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Lucas
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What do you mean by asterisk console, I ran asterisk -r and entered sip show
peers and this is what i got:
server1*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
4289/4289 192.168.2.3 D N 8925 OK (2 ms)
John
DL == Doug Lytle [EMAIL PROTECTED] writes:
DL Tim Uckun wrote:
Judging by the lack of response here it seems like this is broken
and nobody knows how to fix it.
DL 98% of the people here don't use Trixbox.
I have the same issue as reported earlier. I have never tried Trixbox.
/Benny
Thats really strange .. if you have made canreinvite=no then it should not
even attampt native bridging and should transcode codecs ..something's fishy
here .. Also try to put canreinvite=no in testulaw exntension too .
On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote:
El jue, nov 16 de 2006
On Thu, 16 Nov 2006, Zeeshan Zakaria wrote:
You said voxbox is better, but even the link you gave for them didn't work.
I googled, and apparantly links are broken on their website.
Not to mention that the humongous ad for voxbox, in response to my
TECHNICAL QUESTION, was completely out of
On Thu, 16 Nov 2006, Al Bochter wrote:
I never said voxbox is better than trixbox.
I said You like trixbox Should try voxbox.
Yes.
You didn't answer my question, and then you posted all of THIS crap...
Are you outside of the US?
Do you need to call US Toll Free Numbers?
We
We have 3 sites located across the US. Each has its own Asterisk PBX
with a stand-alone installation. The sites are connected via VPN, fully
messed, with fractional DS3s to the same service provider.
We'd like to set it up so that if the PBX at site A fails, it fails over
to B or C, if the
98% of the people here don't use Trixbox.
I don't think this is something with trixbox. I asked the person
having the same problem as me if he was using trixbox to see if that
would narrow down the realm of the problem.
Anyway the error message is in the asterisk log. Googling around I
see
I can confirm the BAD! BAD! BAD! message on both our servers running
1.2.13. Our servers running 1.2.7.1 do not exhibit the problem.
Pre 1.2.8, chan_sip would try to get the lock indefinitely, and would
basically hang the thread until the lock was fetched. Since 1.2.8, it
gives up after 100
I'm sure I'm not the only one, that when seeing the message posted, I'm
having x issues with Trixbox I press the delete button most of the
time. Your lack of response might be because of this.
On the on the one hand I can understand this because trixbox has it's
own configuration schema and
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