Hi,
Does anyone have this card(specifically the wct4xxp driver) working under
linux and running a SMP kernel?
I'm running it in a dual p4 xeon box and when I compile the kernel for SMP
and then recompile libpri/zaptel the module doesn't behave correctly(doesn't
pick up the pri's).
In addition
I did a quick check on the blindxfer config parameter and i cant find any
referense to that in the sourcecode for 1.2!
In previous version all the call transfer things where handled but the
flash button '#' but could also be done by a short hangup (200ms) on the
line so im not shure what you
A SIP phone with the possibility of showing message waiting can get that
information from Asterisk. My EyeBeam is showing a small image of a letter
in the display to show that there are messages waiting. SO you can use this
without mail being sent-out.
Best regards
jan
--On Wednesday,
Sean Kennedy wrote:
Is this a known thing? Can anybody give me an idea of how to change the
Blind Transfer key sequence to something else?
I assume you're using v1.2.
If you change anything in features.conf and then restart asterisk, you
can connect to the CLI and do show features to see
Would anybody please tell me,
If I keep enabled=yes, cdr_manager would be enable, I know
but an 'enabled' cdr_manager would help me?
How I can be benifited from this in terms of cdr management?
What exactly it does if I keep enabled=yes?
As I said: If you set enabled to yes you receive CDR
Hi Kris,
I have TE406P (same as your but quad span) working on 2.6.13 with
pre-empt.
I had it working fine with 2.6.14 but I could not switch card's IRQ from
CPU0 to CPU1 on the 2.6.14
On 2.6.13 CPU1 is handling IRQ's only for TE406P (with occasional timer
IRQ's sneaking in).
I suggest that you
Hi Boris,
I think it might have something to do with the HT(hyperthreading) support.
Since I have one working fine under a dual-amd setup.
Kind Regards,
Kris Amy
Network Engineer
Instant Communications
Australia's Favourite ISP
Tel: 07 3018 8402
Fax: 07 3278 5666
Email: [EMAIL PROTECTED]
Hi,
I have following setup : PBX - Voxip from Parlay -PRI- Asterisk
-SIP- SIP IP GSM Gateway (2n)
on outgoing call from pbx through Voxip and to IP GSM gateway : latter only
responds with SIP session progress but no SIP Ringing message when
connection starts to ring, so Voxip is
Hello,
on http://www.voip-info.org/wiki-Asterisk+zaphfc it is mentioned, that using
BRIStuff breaks PRI support.
We are using Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a with a Digium PRI card an a
beroNet quadBRI
in one server and it's running perfecty for months. It depends only on the
order the
I want to transfer a telephone call in a house number in determinate established hours, this syntax is correct or it must use a command different from DIAL? exten = _x.,1,GotoIfTime(08:30-12:30|mon-fri|*|*?4) exten = _x.,2,GotoIfTime(14:30-18:30|mon-fri|*|*?4) exten = _x.,3,Goto(cellulare,s,1)
On Wed, Nov 30, 2005 at 05:53:54AM -0800, Geo wrote:
OK, fine, thanks. Now I have 1.0.9 still BRIstuffed-0.2.0-RC8h pre-built
don't know how with hisax harmless (eventhough blacklisted), but all runs.
Halas, I have my initial problem. When I dial zap, I just get a continous
dial tone
Hi
I am trying to compile Asterisk 1.2 from source on Debian Sarge but am
getting errors. I have looked at the errors, Googled extensively and now at
a last resort am posting on this list. Believe me I have tried, but have
come up with nothing. I've also installed the following packages from
_x.
Se una chiamate è in incoming esegue s nel context
quindi sarebbe s,1,GotoIfTime.
poi perchè usi un goto ?
intoltre puoi evitare di fare Goto(cellulare,s,1) puoi fare semplicamente
Dial(ZAP/g2/${TELETRASFERIMENTO},60)
oppure se proprio vuoi il goto puoi ottimizzare goto(cellulare)
I dont need to configure zaptel device, you dont use it :)
2005/11/30, [EMAIL PROTECTED] [EMAIL PROTECTED]:
Hello friends,
I am using asterisk 1.2 with ooh323. I am using sip and h323 phones. My
question is I am using a Welltech FXO box and ip phones by Welltech. Do I
still need to
On Tue, Nov 29, 2005 at 09:37:24PM -0600, Kevin P. Fleming wrote:
Mr. James W. Laferriere wrote:
Hello All , no zapata diredtory , tho zaptel README says many
of the testing programs require its libraries .
Please enlighten me . Tia , JimL
The zapata directory was not
If u are using 1.2 there is global var SIP_CODEC or IAX_CODE
exten = 88,1,NoOP(${SIP_CODEC})
exten = 88,2,NoOP(${IAX_CODEC})
Try
29 Nov 2005 15:41:38 -0500, jonc [EMAIL PROTECTED]:
I have a rather curious integration problem. I need to direct a call
connection based on the codec used for the
try [EMAIL PROTECTED]
http://asteriskathome.sourceforge.net/
2005/11/29, bram kortleven [EMAIL PROTECTED]:
Are there any example configs? Or does anybody have a default config
for this setup:
1 analog digium clone card for an analogue line (my home line)
Several sip phones (a few of them on
2005/11/29, Francesco Peeters [EMAIL PROTECTED]:
try ztcfg -vvv
sleep 3
ztcfg -vvv
Also helpful is
cat /proc/zaptel/*
This is what I see:
[EMAIL PROTECTED] ~]# lsmod |grep zaptel
zaptel206724 7
ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2
crc_ccitt
On Wed, Nov 30, 2005 at 12:38:28PM +0200, Hagen Rode wrote:
Hi
I am trying to compile Asterisk 1.2 from source on Debian Sarge but am
getting errors. I have looked at the errors, Googled extensively and now at
a last resort am posting on this list. Believe me I have tried, but have
come
On Wed, November 30, 2005 12:03, Alejandro Vargas said:
2005/11/29, Francesco Peeters [EMAIL PROTECTED]:
try ztcfg -vvv
sleep 3
ztcfg -vvv
Also helpful is
cat /proc/zaptel/*
This is what I see:
[EMAIL PROTECTED] ~]# lsmod |grep zaptel
zaptel206724 7
I use the Zaptel card and when I call a client busy, Asterisk don't play standard tone of busy, but Asterisk play forbidden tone. What can I doing for play busy tone to Asterisk? Thanks Fabio
Yahoo! Messenger: chiamate gratuite in tutto il mondo ___
2005/11/29, Francesco Peeters [EMAIL PROTECTED]:
Then add to a startup file like rc.local:
modprobe zaptel
modprobe zaphfc
ztcfg -vv
I just made exactly as you sed: removed all bristuff, uncompressed it
again, execuded download.sh, downloaded florz patch
(zaphfc_0.3.0-PRE-1_florz-10.diff) and
U dont manage Dial status after dial command
E' perchè non gestisci il risultato del dial
Ex.
tipo
exten = _X.,1,Dial(ZAP/g0/${EXTEN})
exten = _X.,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Playback(il-numero-chiamato-non-risponde)
exten = s-NOANSWER,2,Hangup
exten =
Probabily zaphfc not loaded
retype ztcfg -vvv
2005/11/30, Alejandro Vargas [EMAIL PROTECTED]:
2005/11/29, Francesco Peeters [EMAIL PROTECTED]:
Then add to a startup file like rc.local:
modprobe zaptel
modprobe zaphfc
ztcfg -vv
I just made exactly as you sed: removed all bristuff,
Robert Rozman wrote:
Hi,
I have following setup : PBX - Voxip from Parlay -PRI-
Asterisk -SIP- SIP IP GSM Gateway (2n)
on outgoing call from pbx through Voxip and to IP GSM gateway : latter
only responds with SIP session progress but no SIP Ringing message when
connection starts
2005/11/30, Francesco Peeters [EMAIL PROTECTED]:
You are running the HFC-PCI in NT mode. This means you have an ISDN
telephone connected to it, rather than using it to connect to the PSTN?
Thanks, now I changed this to mode=0
What is in your /etc/asterisk/zapata.conf? I do not recall seeing
Hi, I´ve recently installed my first Asterisk and it´s working. I can only
make outbound calls trough internet. I was willing to record the phone calls
in files maybe with wav or gsm extension. Can someboy help me a little with
this?
Thanks
Felix
On Wed, November 30, 2005 12:28, Alejandro Vargas said:
2005/11/29, Francesco Peeters [EMAIL PROTECTED]:
Then add to a startup file like rc.local:
modprobe zaptel
modprobe zaphfc
ztcfg -vv
I just made exactly as you sed: removed all bristuff, uncompressed it
again, execuded download.sh,
Waiting a bit for 1.2, not yet ready to rewrite the dial-plan.
There were enough fixes, etc, in v1.2 that I'd consider it a priority
to get there fairly soon.
What do you mean Yes the calls out are/were to Zap/g1/xxx?
Your outbound extensions.conf entry should look something like:
There is a Kirk distributor for NZ.
http://www.wavelink.com.au/
Good luck,
Joe
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
On Wed, November 30, 2005 12:55, Alejandro Vargas said:
2005/11/30, Francesco Peeters [EMAIL PROTECTED]:
You are running the HFC-PCI in NT mode. This means you have an ISDN
telephone connected to it, rather than using it to connect to the PSTN?
Thanks, now I changed this to mode=0
What is
On Wed, November 30, 2005 13:10, Felix Amaral said:
Hi, I´ve recently installed my first Asterisk and it´s working. I can
only
make outbound calls trough internet. I was willing to record the phone
calls
in files maybe with wav or gsm extension. Can someboy help me a little
with
this?
Felix Amaral schrieb:
Hi, I´ve recently installed my first Asterisk and it´s working. I can only
make outbound calls trough internet. I was willing to record the phone calls
in files maybe with wav or gsm extension. Can someboy help me a little with
this?
2005/11/30, Francesco Peeters [EMAIL PROTECTED]:
Wrong order: First zaptel, then zaphfc. Do an lsmod to verify both are
loaded before ztcfg
Ok, now I will remove all and try. But when I applied Florz patch
every time I load zaphfc the system hangs.
First, when compiling zaphfc (after applying
On Wed, November 30, 2005 13:54, Alejandro Vargas said:
2005/11/30, Francesco Peeters [EMAIL PROTECTED]:
Wrong order: First zaptel, then zaphfc. Do an lsmod to verify both are
loaded before ztcfg
Ok, now I will remove all and try. But when I applied Florz patch
every time I load zaphfc the
2005/11/30, Francesco Peeters [EMAIL PROTECTED]:
If it remains after correct order (zaptel then zaphfc), please try insmod
zaphfc debug=3.
In that case also show us the complete output from lspci, and check dmesg
and /var/log/messages for any zaptel and zaphfc messages.
[EMAIL PROTECTED]
2005/11/30, Francesco Peeters [EMAIL PROTECTED]:
These are weird warnings...
Have you done make clean before make?
Have you first compiled the patched zaptel?
AHH!! I must compile and install the zaptel module included
with bristuff replacing the one included whith asteriskathome, is
On Tue, Nov 29, 2005 at 03:25:19PM -0500, Michaël Gaudette wrote:
Hi,
I`m a beginning Asterisk and Sendmail user.
Note that the sendmail need not be sendmail. It can be basically ant
mail transfer agent (MTA). Postfix, exim and maybe qmail will do s well.
I am trying to setup my
voicemail
On Wed, Nov 30, 2005 at 01:14:35PM +0100, Francesco Peeters wrote:
On Wed, November 30, 2005 12:28, Alejandro Vargas said:
2005/11/29, Francesco Peeters [EMAIL PROTECTED]:
Then add to a startup file like rc.local:
modprobe zaptel
modprobe zaphfc
ztcfg -vv
I just made exactly as you
On Wed, November 30, 2005 14:19, Alejandro Vargas said:
2005/11/30, Francesco Peeters [EMAIL PROTECTED]:
If it remains after correct order (zaptel then zaphfc), please try
insmod
zaphfc debug=3.
In that case also show us the complete output from lspci, and check
dmesg
and
On Wed, November 30, 2005 14:31, Alejandro Vargas said:
2005/11/30, Francesco Peeters [EMAIL PROTECTED]:
These are weird warnings...
Have you done make clean before make?
Have you first compiled the patched zaptel?
AHH!! I must compile and install the zaptel module included
with
On Wed, November 30, 2005 14:44, Tzafrir Cohen said:
On Wed, Nov 30, 2005 at 01:14:35PM +0100, Francesco Peeters wrote:
On Wed, November 30, 2005 12:28, Alejandro Vargas said:
2005/11/29, Francesco Peeters [EMAIL PROTECTED]:
Then add to a startup file like rc.local:
modprobe zaptel
2005/11/30, Francesco Peeters [EMAIL PROTECTED]:
AHH!! I must compile and install the zaptel module included
with bristuff replacing the one included whith asteriskathome, is it??
Yep. Even worse: you must replace ALL of Asterisk...
(except config files)
It can be most easily done
I am trying to allow my
conference participants to see who they are talking to.
My dialplan
calls: Meetme(${ARG1} | vMd)
I get audio and no video.
I thought the v option might
do the trick? Am I way off? Any tips?
Thanks..
trond
Trond G. Andersen wrote:
I am trying to allow my conference participants to see who they are
talking to.
My dialplan calls: Meetme(${ARG1} | vMd)
I get audio and no video.
I thought the v option might do the trick? Am I way off? Any tips?
Doesn't work.
Some people have developed
OK. Thank you everybody It is working now. The short solution is this:
download bristuff, execute download apply patch, execute compile and
check configs of asterisk in order to run it.
To add the module to the start, it is easy to add this to /etc/modprobe.conf
options zaphfc modes=0
install
Tzafrir Cohen wrote:
Is it obsoleted? It looks like a nice toy. See e.g. the recent
http://linuxgazette.net/120/smith.html
No, it's still on our CVS servers and will be there indefinitely.
If there is demand (I assumed there wouldn't be) I can easily import it
into SVN as well...
ok got the patchfile
to work but now i have compiling errors:
gcc
-pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686
-DZAPTEL_OPTIMIZATIONS
-fomit-frame-pointer -fPIC -c -o app_rxfax.o app_rxfax.cIn
On Wed, November 30, 2005 15:15, Alejandro Vargas said:
OK. Thank you everybody It is working now. The short solution is this:
download bristuff, execute download apply patch, execute compile and
check configs of asterisk in order to run it.
To add the module to the start, it is easy to add
We're running Asterisk at Home but upgraded to version 1.2 of Asterisk.
After the upgrade the 'Hold' button on our Snom 360 phones now
immediately hangs up a call instead of putting the call on hold. Has
anyone else had this problem and figured out how to fix it?
I ran 'sip debug' in the
Hi,
I have the same problem, same error but loading modules changes nothing.
I'm using debian sarge and Asterisk 1.2: after compiling asterisk I
launched install-misdn from beronet site.
When I started Asterisk the same error arose:
Nov 30 15:43:06 ERROR[4914]: chan_misdn.c:3455 load_module:
When I call a internal Sip telephone, the calling transfer to Teleohone external, but Asterisk show this error: Executing Dial("SIP/201-1e2a", "ZAP/g1/3472543320|60") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/3472543320 Nov 30 15:52:09 WARNING[1866]:
Hello Kevin ,
On Tue, 29 Nov 2005, Kevin P. Fleming wrote:
Mr. James W. Laferriere wrote:
Hello All , no zapata diredtory , tho zaptel README says many
of the testing programs require its libraries .
Please enlighten me . Tia , JimL
The zapata directory was not
Mr. James W. Laferriere wrote:
Any reason why ? Tia , JimL
There were many 'stale' projects that I didn't bother to import. Given
that nothing has been changed in that project for over a year, and that
nothing in Zaptel (in normal use) relies on it. it seemed a good
candidate to be
2005/11/30, Francesco Peeters [EMAIL PROTECTED]:
When you do, make VERY sure the PCI slots are NOT sharing an IRQ! That'll
break it every time!
Did you try to use APIC? This is suposed to solve the problem of IRQs
--
Alejandro Vargas
___
--Bandwidth
Can anyone on the list recommend any IAX Service providers
in Australia for unlimited inbound in the 02 area code?
Ive been using Faktortel for A$9.50 per month and
although the outbound is fantastic (I mean the quality is fantastic the
fixed price 10c per call Australia
wide is pretty
I keep getting this error message from one of my Avaya 4620SW hard phone.
Got SIP response 400 Invalid Subscription-State back from
192.168.xx.xx which is the IP address assigned to that hard phone. Also
the phone will still have dial tone but cannot make or recieve any
calls.
Thanks
Dean Collins [EMAIL PROTECTED] uttered the following thing:
Can anyone on the list recommend any IAX Service providers in Australia
for unlimited inbound in the 02 area code?
You can try www.austechpartnerships.com.au though their outbound is a
bit more expensive.
BB
Rich Adamson wrote:
I've noticed that if I enable the jitterbuffer in iax.conf with Asterisk
1.2.0 that Asterisk stops responding to incoming DTMF frames for calls
between Teliax and my server. I've used iax2 debug and Ethereal to
confirm that Teliax is, in fact, sending the frames.
I only
In checking this out, how does one implement it.. the readme is very vague.
I really like the IAXmodem with hylafax for incoming, and has been working
great. I would like to explore the outbound faxing capabilities, but
havent had a chance to go down that road. Right now I can fax out using
the
Hi,
I'm setting up an Asterisk 1.2 PBX based on a Debian Sarge distro with a
quadBRI beroNet card.
I've followed beroNet instructions so I compiled Zaptel, Libpri and
Asterisk and then launched install-mISDN script downloaded from beronet
site (install-mISDN.tar.gz).
I try to start Asterisk
Why Asterisk show this message: Nov 30 17:05:17 WARNING[1351]: chan_sip.c:9600 handle_response_register: Got 200 OK on REGISTER that isn't a register Thanks
Yahoo! Messenger: chiamate gratuite in tutto il mondo ___
--Bandwidth and Colocation provided
I am working with a 3rd party provider who is providing me with IAX2
dialtone. I am using GSM codec end-to-end and my provider insists on ULAW
only. When my remote IAX clients attempt to use the provider for PSTN calls
by calling my primary * box, and my primary's dialplan is set to dial the
hi all i have a pbx siemens connect via E1 to my asterisk box.
the asterisk box can call without problems to pbx extensions. but when y
press the numbers form example 402 in the pbx phones asterisk give me
this
-- Saved useragent X-Lite release 1103m for peer 402
-- Going to extension s|1
You may have already done this, but my first approach would be to look hard
at the Vocal Data switch and see if you can disable G723 support on the
switch.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - -
Answered my own question, partially from the route call based on codec
thread:
If u are using 1.2 there is global var SIP_CODEC or IAX_CODE
exten = 88,1,NoOP(${SIP_CODEC})
exten = 88,2,NoOP(${IAX_CODEC})
So I can modify my dialplan to check the codec. If it's anything but GSM,
route to the IAX
You could try using one of the dial functions that listen to DTMF i.e. t or T
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
Hi all,
I'm new in Asterisk so I'll thank a lot any help. When I start
Asterisk: asterisk -cvv, the output is as follows:
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk , Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer
Hi folks,
I am having a small problem with a few Sipura units. The settings are
pretty much factory stock: the unit is set up to not register and the
IP address for the unit is static and defined in the SIP setup for
that unit. All other calls are sent and received properly, this is
the
Here is the example
The output of asterisk:
-- Registered IAX2 '4000' (AUTHENTICATED) at 127.0.0.1:33384
The output of iaxmodem:
[EMAIL PROTECTED] iaxmodem-0.0.5]# ./iaxmodem ttyIAX
Setting device = '/dev/ttyIAX'
Setting port = 4569
Setting refresh = 300
Setting server = '127.0.0.1'
Martin Joseph wrote:
It's format=wav49|gsm|wav
Try swapping the wav49 and the wav; my voicemail messages were garbled
until I did this:
format=wav|gsm|wav49
You should try not to just tack one line on top of a long message to
list... ;~)
ok, sorry :]
When I dial *99 from the phone connected to line 1,
I cannot complete a call.
Go to the Regional tab in the advanced admin menu, find the Vertical
Service Activation Codes section. Remove which ones you don't want the
Sipura to handle (i.e. *99).
Luki
I am having a small problem with a few Sipura units. The settings are
pretty much factory stock: the unit is set up to not register and the
IP address for the unit is static and defined in the SIP setup for
that unit. All other calls are sent and received properly, this is
the only
Hello
Im managing a WAN with a lot of Universities. Some of them already
installed a VoIP solution based on SER (to manage SIP clients) and
Asterisk (for services and PSTN GW). The DNS routing provided by SER is
working perfectly, but we want to start routing all calls thru IP
transparently.
I have a very similar server, pstn setup, phones, and user base, and I
switched over to G729 codec 'cause the polycoms support it. While the
call quality has dropped ever so slightly (I have received no complaints
from my users however), snaps, crackles, clicks and pops are gone. I
did not
On Wed, November 30, 2005 16:29, Alejandro Vargas said:
2005/11/30, Francesco Peeters [EMAIL PROTECTED]:
When you do, make VERY sure the PCI slots are NOT sharing an IRQ!
That'll
break it every time!
Did you try to use APIC? This is suposed to solve the problem of IRQs
Yep, tried APIC,
On Nov 30, 2005, at 12:39 PM, Luki wrote:
When I dial *99 from the phone connected to line 1,
I cannot complete a call.
Go to the Regional tab in the advanced admin menu, find the Vertical
Service Activation Codes section. Remove which ones you don't want the
Sipura to handle (i.e. *99).
Just a reminder tonight at midnight is the deadline for pstn connected
VoIP providers operating in the US to provide E911 or face fines upto
$11,000 per day. There is also a filing requirement with the FCC which
is due tonight as well.
Enforcement Bureau Outlines Requirements of November 28,
On Wed, 2005-11-30 at 17:45 +, Joao Pereira wrote:
Hello
Im managing a WAN with a lot of Universities. Some of them already
installed a VoIP solution based on SER (to manage SIP clients) and
Asterisk (for services and PSTN GW). The DNS routing provided by SER is
working perfectly, but
You should take a look to ENUM protocol:
http://www.voip-info.org/wiki/view/ENUM. It could provide a decentralized
and simple solution for your requirements.
Regards
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Joao Pereira
Enviado el: miércoles, 30
On Mon, 2005-11-28 at 15:01 -0800, trixter aka Bret McDanel wrote:
Just a reminder tonight at midnight is the deadline for pstn connected
VoIP providers operating in the US to provide E911 or face fines upto
$11,000 per day. There is also a filing requirement with the FCC which
is due tonight
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Looks like your zap channels are droping into the default context...
better to set up a from-pstn context and start there.
Pablo Allietti wrote:
hi all i have a pbx siemens connect via E1 to my asterisk box.
the asterisk box can call without
Hi everyone,
Does anyone have this working? I'm looking at these phones for my
receptionist phone, with the requirement that the two bars of buttons
and lights on the side show line presence for programmable extensions (
ie: line 1 show the presense of my 101 user, line 2 = 102 user, ect..
Try www.oztell.com
they have a somewhat complicated website interface but once you figure it out
its ok and I found them to be by far the cheapest provider in Australia.
They offer DIDs at $1.95 per month.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
Are you using the libiax2 that came with iaxmodem? If you are, then I'm
not sure what to say... the client-server behavior looks bizarre.
Lee.
Miguel Soto wrote:
Here is the example
The output of asterisk:
-- Registered IAX2 '4000' (AUTHENTICATED) at 127.0.0.1:33384
The output of
The Remote hangup messages disappear if I set qualify=no in the
iax.conf file. But is this correct?
Miguel
-Original Message-
From: Miguel Soto [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 30, 2005 10:31
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
Yes, I am using libiax2 that came with iaxmodem :)
-Original Message-
From: Lee Howard [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 30, 2005 11:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] iaxmodem
Are you using the libiax2 that
On 10:25, Wed 30 Nov 05, Sean Kennedy wrote:
Hi everyone,
Does anyone have this working? I'm looking at these phones for my
receptionist phone, with the requirement that the two bars of buttons
and lights on the side show line presence for programmable extensions (
ie: line 1 show the
Hey, you are my new best friend. I have never had a phone to use with
the hint priority, would you mind giving me a sample of your
configuration so I can figure it out?
Much apprecaited!
Sean
Michiel van Baak wrote:
On 10:25, Wed 30 Nov 05, Sean Kennedy wrote:
Hi everyone,
I want to play a file for an agent that answers a queue call, before
the agent is actually connected with the call. I want something along
the lines of,Answer as member of team X, or similar, before the
agent is connected with the caller. Is this possible? And how would
I do it?
--
Trey
Hi Sean,
Works fine for me as well. Took some working to get right. There's a
very recent thread on this, see:
http://lists.digium.com/pipermail/asterisk-users/2005-November/136343.html
Also, you'll need to go into the web interface for your Snom phones and
configure each button for each
Hi all,
Pulse dialing is not working on my asteriskathome configuration with
asterisk 1.2.0 and zaptel 1.2.0 in France.
I've pulsedial=yes in zapata.conf.
Tone dialing is working 100%.
In file zapata.h (zaptel 1.2.0), I've found the following :
#define ZT_DEFAULT_PULSEMAKETIME 50 /* 50
I use 1.2 Beta 1 in pulse dial mode, as an interface to an EM switch. A
bunch of collectors in the US and 2 in the UK have a private network of
historic switches interconnected via the Internet and Asterisk.
If the make break time is a problem you can change in the source but
will have to
We just upgraded our current asterisk cluster to the release version of
Asterisk 1.2.0. Strange enough, out of the 11000+ calls, only 720 (and
counting) have a disposition of FAILED in the cdr's. These 720+ have
only occurred after the upgrade, and I'm rather confused as to why it
would show up
On 11:33, Wed 30 Nov 05, Sean Kennedy wrote:
Hey, you are my new best friend. I have never had a phone to use with
the hint priority, would you mind giving me a sample of your
configuration so I can figure it out?
Much apprecaited!
Hey hey new pal ;)
First of all, have a look at this
I am new to Asterisk.
Asterisk 1.2
I started * like this: asterisk
-vgc
now I am in CLI mode: *CLI
How do I get out this CLI mode to linux shell
without kill asterisk process?
I tried EXIT, QUIT, exit and quit. None of them
work.
If I use ^c, this also kill asterisk
process.
GC
gc wrote:
I started * like this: asterisk -vgc
now I am in CLI mode: *CLI
How do I get out this CLI mode to linux shell without kill asterisk process?
if you want to run it like this, first do a screen (more info: man
screen) so you can run it in a background shell. But I recommend on
Title: Message
just
press Ctrl-C or type exit
You
will kill asterisk, of course...
Start
asterisk by typing asterisk
and
then go toCLI by typing asterisk -r
then,
when u will quit, asterisk will not be killed
U will
be then in CLI mode
have
fun
-Message
Kill
the Asterisk process
Launch
Asterisk as a background process by typing asterisk or use the
safe_asterisk shell script (better)
type asterisk
r to connect to the console
Press
Ctrl C to exit the console. Use ps a | grep Asterisk to determine if the
I'm having problems setting up the CDR functionality. Namely, it doesn't
always wok (but I do have some records). When typing cdr mysql status in
the Asterisk console, it does say connected for 3 minutes 22 seconds, with
0 records added since last restart. But I did call a few times into my
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