In the dialplan you can use ChanIsAvail command
Show channels?
On Mar 31, 2006, at 2:09 AM, Ronald Wiplinger wrote:
In the past I used SetGroup and CheckGroup to figure out if my
allowed providers lines are all used or not.
Since most of my provider have given me a single line anyway, I
Anyone know how I can get SIP T working w/ Asterisk?
TIA,
Jon
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Message: 24
Date: Mon, 03 Apr 2006 19:21:57 -0500
From: Michael Graves [EMAIL PROTECTED]
Subject: [Asterisk-Users] New SkypeSIP gateway
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain;
On 4/5/06, Matt Florell [EMAIL PROTECTED] wrote:
In my experience capacity is a huge problem. You can't have sphinx
running on 48 channels at once. It is limited to only a few instances
at a time. Although I only did trials with sphinx2.
What version are you using? and what dictionary?
On Tue, 2006-04-04 at 10:44 -0400, Christian Buchter wrote:
Strange, but all the phones when called immediately return a user is on
the phone and the phone never rings.
Anyone else ever experience this before?
TIA
Have the users managed to set DND on the phones? That would give the
I have recompiled my zaptel drivers but I still get
the same error
--- Derek Whitten [EMAIL PROTECTED] a écrit :
ali asma wrote:
I modified the configuration but I still have the
same
error.
Please tell me in whach directory should I
execute:
modprobe zaptel
modprobe wcfxo
becose
Snom 190s and 220s, it seems to happen intermittently but not sure why
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Tuesday, April 04, 2006 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
What phones you using?
On 4/4/06, Christian Buchter [EMAIL PROTECTED] wrote:
Strange, but all the phones when called immediately return a user is on
the phone and the phone never rings.
Anyone else ever experience this before?
TIA
Hi,
Please take a look at the following extensions.conf:-
exten = _11,1,NoCDR()exten = _11,2,Dial(SIP/${EXTEN},10)exten = _11,3,VoiceMail()
I'm already using realtime for some extensions/users/voicemail.
Is there any way to do the following at point 3?:-
Lookup the realtime users
Hi,
I currently use Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian sarge
with a kernel 2.4.27 on a P4 3Gig with 1Gig
of memory
When i use i4l on any call, the called party ( on the telco operator
side ) ear me
Kelvin Williams wrote:
We are using Asterisk in a purely VOIP environment, on leased
dedicated server at a dedicated server provider. It is becoming
more and more apparent that this dedicated server is actually a
vritualized server.
We have now found a need to utilize the MeetMe application
Shad Mortazavi wrote:
Message: 24
Date: Mon, 03 Apr 2006 19:21:57 -0500
From: Michael Graves [EMAIL PROTECTED]
Subject: [Asterisk-Users] New SkypeSIP gateway
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
For phones, I've got a GS 101, a Sipura 841, and two analog phones hooked to an
GS386 ATA (one phone per port).
My troubles seem to be regardless of which phone is used, so I dont think it's
on the phone-end of asterisk, but rather where I interface w/ Vonage and
Verizon via POTS FXO... My
To me, your * config files look correct. At a guess I'd say the problem is in
your motherboard. It is a sis chipset and from the look of things a couple
years old. Try running the system on an intel chipset motherboard and see how
you go.
Alternately, if you are running X windows, then
maybe firewall tends to close iax connection,
you can try to decrease qualify check interval (maybe qualify=5000?)
PJ
Mimmus wrote:
Pavel Jezek wrote:
I have same problem, do you have asterisk box behind nat?
No, they are not behind NAT, peraphs there is a Checkpoint firewall.
Strange, but all the phones when called immediately return a user is on
the phone and the phone never rings.
Anyone else ever experience this before?
TIA
_
This email has been scanned by MessageLabs on behalf of E-INS
On Tuesday 04 April 2006 10:39, Lee Archer wrote:
I've been looking through the logs of a system trying to figure out why
it sometimes starts extra asterisk processes. In the logs I keep seeing
Define starts extra asterisk processes.
Apr 4 15:22:18 WARNING[5054] chan_zap.c: Can't fix up
ChanIsAvail allows you to see if a channel can *accept* calls, not if it is
currently in use. Here is a script that will fix you up:
checkchannel.agi - returns number of channels in use on a SIP peer
Sets a variable in the dialplan, MYCHANNELS, indicating number of channels
in use
#!/bin/bash
Title: Possible PRI fault?
I've been looking through the logs of a system trying to figure out why it sometimes starts extra asterisk processes. In the logs I keep seeing
Apr 4 15:22:18 WARNING[5054] chan_zap.c: Can't fix up channel from 1 to 2 because 2 is already in use
Apr 4 15:22:18
hi all,
i have an asterisk install with a digium 4 port fxo card and cisco 7960
sip phones -- running on a compaq Pentium III (Coppermine) at 800Mhz
256KB cache and 1GB of ram.
when a call comes in on zap/1-1 for example, the delay between when zap
sees the line going to ring state, and when the
Steve Jones wrote:
I thought the whole thing with the hardware echo cancellation is that
it was basically in liu of the equivilent echo cancellation done in
software... The reason to go to the hardware was for high-density
systems?? For two FXOs, I thought I'd be safe in getting the
non-echo
I had the same problem with i4l.
It seems to be a driver problem. I think i4l is depricated for a reason in
the newer Asterisk versions.
Funny thing is: When I switch the remote users into a MeetMe room. And have
the local users dial in to the same meetme room.
Then the problem disappears (at
I am having this exact same problem. I have tried 7.5, 7.4, and 8.2. I
have tried setting ALERT_INFO and _ALERT_INFO and have tried several
ringtones without any luck.
According to the WIKI, it should work:
[snippet]
Controlling ring tones from Asterisk
By setting the Asterisk variable
Check your DTMF Settings.
--- hensem boy [EMAIL PROTECTED] wrote:
Hi all
I have a problem when I want to call out using VPB
trunk line, it cannot send the DTMF. Is there anyone
has the same problem? Please share with me the
solution.
Thanks.
The load on the system will crash your server with that many instances
of real-time sphinx running. Take a look at 'top' while you run it on
tow channels at once an see what the load is.
MATT---
On 4/5/06, Matt [EMAIL PROTECTED] wrote:
On 4/5/06, Matt Florell [EMAIL PROTECTED] wrote:
In my
That is why I back up my web server to an ftp server
in a diffrent data center :)
--- Rob Thomas [EMAIL PROTECTED] wrote:
Well, I wake up this morning, and aussievoip isn't
up. I ring godaddy,
who _were_ hosting it, and they say that the
machine's been compromised,
and you can't have your
oej's MeterMaid patch for monitoring parked calls through hints:
http://bugs.digium.com/view.php?id=5779
Anyone tried it on 1.0.9?
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Hello,
While upgrading * from 1.0.9 to 1.2.5, I have
installed chan-capi-head and I cant start asterisk under asterisk group
asterisk -gc -U asterisk and asterisk
-gc -U asterisk -G dialout work well but asterisk -gc -U
asterisk -G asterisk fail.
I am thinking about a
i have an asterisk install with a digium 4 port fxo card and cisco 7960
sip phones -- running on a compaq Pentium III (Coppermine) at 800Mhz
256KB cache and 1GB of ram.
when a call comes in on zap/1-1 for example, the delay between when zap
sees the line going to ring state, and when the desktop
On Wed, 5 Apr 2006, Alain Degreffe wrote:
Hi,
I currently use Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian
sarge with a kernel 2.4.27 on a P4 3Gig with 1Gig
of memory
When i use i4l on any call, the called party ( on the telco operator
Wondering if anyone has experienced an intermittent one way
audio (called party can not hear) problem in these conditions;
Several IP501 phones local, same subnet.
Remote asterisk
No NAT anywhere
Polycom IP501 ulaw only, canreinvite=yes
Asterisk
Call termination path is to a
On 04/05/06 21:37 Dov Bigio said the following:
- The agent transferred the call to an user (not a queue), by dialing
the atxtransfer (1) key defined in features.conf
on a related note, we notice that if we've set atxfer = *1 in features.conf
and blindxfer=#1, then attended transfers dont
List,
how can I put fax server functionality on Asterisk? * as a reliable fax
server for 500-1000 fax/day (mostly incoming)? Fax server should be like
HylaFax, i.e. stable, low maintenance and functionality like receiving
fax as email with PDF attachment, sending faxes per WHFC.
Faxing with
Well give it a day and I will reply to my own
questions. I guess my friends are right that I do talk to myself :)
Anyways, Sprint called back and according to their
technician, "Oh, I'm sorry, it looks like we do have milliwatt test lines
that support 1004 Hz or 1.004 kHz test tones"
So
I am having this exact same problem. I have tried 7.5, 7.4, and 8.2. I
have tried setting ALERT_INFO and _ALERT_INFO and have tried several
ringtones without any luck.
Using the current svn trunk, here is what works:
exten = 3010,1,Set(_ALERT_INFO=bellcore-r3) ; selects Ringer
exten =
Can you think of any reason that this would not pick up on times after call
is placed, and then disconnected. I noticed that the time does not change on
the call times after a call has been made.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo
Hi All
I have a TE110P card connected to a PRI line. In my zaptel.conf I have:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone = us
defaultzone=us
and my zapata.conf is:
[channels]
context=inbound-pri
switchtype = national
pridialplan=unknown
;pridialplan=international
signalling = pri_cpe
Well, I wake up this morning, and aussievoip isn't
up. I ring godaddy,
who _were_ hosting it, and they say that the
machine's been compromised,
and you can't have your data. Nyah Nyah.
Have you tried the Internet archieve (wayback machine). I was once lucky
to find my web pages
My kernel is a 2.4.27 and I think that mISDN is available only for a 2.6.x
But I can't use a 2.4.26 for some security reasons...
Alain
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Armin
Schindler
Envoyé : mercredi 5 avril 2006 18:05
À : Asterisk
Hi Andy -
Anyone care to quote on 25 Linksys PAP2-NA units unlocked can email me
direct.
Straight forward sale best price new equip etc etc... I am a buyer located
in the U.S.
Need someone with stock that can ship right away. Will want 25 more in less
than a week.
You may get a better
Hi,
SHOWCHANINFO outputs no data in the following line:-
exten = 1571,2,VoiceMailMain(${SIPCHANINFO(peername)[EMAIL PROTECTED])
So that command becomes:-
exten = 1571,2,VoiceMailMain(@incoming)
Can anyone help?
Thanks
Dan Journo
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Frank Ochmann wrote:
List,
how can I put fax server functionality on Asterisk? * as a reliable fax
server for 500-1000 fax/day (mostly incoming)? Fax server should be like
HylaFax, i.e. stable, low maintenance and functionality like receiving
fax as email with PDF attachment, sending faxes
Hi Marco
My asterisk for all my users, everything was fine for 3 days, but now
i can't access it.
But it is running...
Could any one help me on this?
Can you provide some specific information? At least the following:
Asterisk version
Operating System
Hardware
Technologies used (zap, sip,
Unfortunetly, you learnt the hard way. Never rely on any third party. Make sure you have backups of all your data on machines which you have direct access to.
There are a large number of offsite companies offering backup services. Checkthem out and make sure their contracts still allow you to sue
Any clue for other countries (western Europe, for example) ?Cheers
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change asterisk.conf:
mkdir /var/run/asterisk
chown it to your asterisk user.
change astrundir = /var/run to astrundir = /var/run/asterisk
My guess would be that you are running asterisk as a non-root user and that
this user can not write to /var/run .
if so, the ctl and PID files are not
OOps
The correct answer is
My kernel is a 2.4.27 and I think that mISDN is available only for a 2.6.x
But I can't use a 2.6.x for some security reasons...
Alain
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
[EMAIL PROTECTED]
Envoyé : mercredi 5
Hello,Which is your favorite SIP softphone with command line interface (ie with text imputs and outputs along with graphical GUI) ?Regards
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I run it, make a call, and after the call disconnects, when I run the
script again, I do get changed numbers:
[EMAIL PROTECTED] ~]$ total_account_codes /var/log/asterisk/cdr-csv/Master.csv
total is 151974 seconds or 2532.9 minutes or 42.22 hours
[EMAIL PROTECTED] ~]$ pbxmonitor
Mojo 7478633
2006/4/4, Remco Barende [EMAIL PROTECTED]:
I suspect that in your case the fax channels are not natively bridged. I'mnot sure whether native bridging will work if you are using 2 cards.How would you prove that native bridging works (I mean independantly of current server processor or PCI bus load)
Hi there,
Anyone knows the Tornado M5 IP Phones? I need to connect
them to Asterisk, but I could not found any info.
Best regards,
Ing. Juan Carlos Huerta
Director
de Desarrollo
Nucleum,
la voz de tu red
[EMAIL PROTECTED]
www.nucleum.com.mx
Hello,
I have connected asterisk box with legacy PBX Alcatel OmniPCX 4400 (and also
another * box connected to A4200).
These PBXes have function to assign name to extensions and display it on
phone.
Asterisk box is connected via PRI with euroISDN signalling (also I have tried
QSIG).
Is it
On Wed, 2006-04-05 at 11:02 -0500, Rich Adamson wrote:
snip
It would appear the progress is associated with waiting for callerid
info. If you are in the US, callerid occurs between the first and second
ring. That's about 7 seconds or so.
If your pstn line does not have callerid, then add
I've been told that the problem was:
I've a daily cron job:
/usr/sbin/asterisk -r -x stop when convenient
then i had
/usr/sbin/asterisk start
I've been recomended to replace:
/usr/sbin/safe_asterisk
I've done that, let's see how it goes tomorrow when i arrive at the office.
I didn't have
Title: Messaggio
My sip phones
are connected to asterisk PBX 1.2.4. The PBX is connected to the provider
through IAX2 connection. Sometimes randomly the voice is stopped and both caller
and called don't hear the other's voice. During this silence period Asterisk is
not logging any errors.
Hi all,
I've a some users on my network, reporting this:
Sjphone is registered , and some times just looses registry in
Asterisk, I don't know if it is expiration ( instead of loosing
registry).
Then to get registered again they need to restart their own PC.
Why could this beeing happening?
It should work with that permissions. Does it work with other group/user
settings?
Just for a try, set /dev/capi20 to rw-rw-rw
Armin
On Wed, 5 Apr 2006, amaury BOSSE wrote:
Hello,
While upgrading * from 1.0.9 to 1.2.5, I have installed chan-capi-head
and I can't start asterisk under
I suppose that works. I get two short rings. Is there a way to change the
actual sound of it, though?
On Wed, 5 Apr 2006, Rich Adamson wrote:
I am having this exact same problem. I have tried 7.5, 7.4, and 8.2. I have
tried setting ALERT_INFO and _ALERT_INFO and have tried several
Miroslav HOSTINSKY napisał(a):
Hello,
I have connected asterisk box with legacy PBX Alcatel OmniPCX 4400 (and also
another * box connected to A4200).
These PBXes have function to assign name to extensions and display it on
phone.
Yes. They do :-D.
A4400, current amount: 3. A4220E
5 apr 2006 kl. 08.52 skrev René Enskat [Teamware GmbH]:
hi
i updated asterisk today via svn no i can'T start asterisk i get
core dumps.
i have to comment some modules then i can start:
noload = format_au.so
noload = format_mp3.so
noload = format_pcm_alaw.so.so
noload = format_pcm_alaw.so
The subject says it all I think. I'm looking at maybe needing to run it
under BSD 5
Thanks in advance
Bruce
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Frank Ochmann escreveu:
how can I put fax server functionality on Asterisk? * as a reliable fax
server for 500-1000 fax/day (mostly incoming)? Fax server should be like
HylaFax, i.e. stable, low maintenance and functionality like receiving
fax as email with PDF attachment, sending faxes per
Hi all
I finaly set up a second * with two ZapHFC Cards. One in TE the other in NT
mode.
So I have a 1.2.5 Asterisk to run Meetme etc... and a 1.2.4 Asterisk to run
all that Zaptel stuff. First I used mISDN on 1.2.5 which worked, but
sometimes had strange behaviour.
So my hope was that zaptel
Hi List,
is there a reason why Asterisk Realtime queues don't support
periodic_announce_frequency and periodic_announce options?
I have tried adding the 2 fields to my MySQL table,
but they seem to be ignored?
Any hints are appreciated.
Regards
Kristian
On 11:12, Wed 05 Apr 06, Bruce Ferrell wrote:
The subject says it all I think. I'm looking at maybe needing to run it
under BSD 5
It runs fine on OpenBSD 3.8
No zaptel though, but for FreeBSD there's a zaptel port.
http://ezine.daemonnews.org/200409/asterisk.html
Friends,
At this point, we're close to 300 issues open in the bug tracker
at http://bugs.digium.com
Some of us spend many hours each week,
if not each day, to work with the bug tracker. It's a tool for us, a
very important tool to handle new features
and find bugs in Asterisk, tracking them
It wacked up to maybe 20% for all of 300ms while it was processing the
data from the caller... hrmmm
On 4/5/06, Matt Florell [EMAIL PROTECTED] wrote:
The load on the system will crash your server with that many instances
of real-time sphinx running. Take a look at 'top' while you run it on
tow
- Original Message -
From: Wes Baehr [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Monday, April 03, 2006 3:16 PM
Subject: [Asterisk-Users] Queues - Dumb question
It was my understanding that when an agent answers
Hello list,
is there any open-source software that recodes g729 sound files to wav
sound files ?
The only way (at least) to do such transformation is with interactive
form on: http://www.asteriskguru.com/audio_conversion.php
Tofik Suleymanov
___
5 apr 2006 kl. 16.40 skrev Jon Weisman:
Anyone know how I can get SIP T working w/ Asterisk?
Start with explaining your definition of SIP T then we can look
into it :-)
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
Well what I need is to get the info digits on a sip call (toll free
orignation) and send that call out a PRI to my PSTN switch via FeatureGroupD
so that I know where the call is originating from. Can I do this with
Asterisk? And how???
-Jon
- Original Message -
From: Olle E
Hi all,
I have here several IAX2 Softphones(IDEFISK, DIAX and an own develop
based on iaxclient.lib). I have follow dialrules in my std-test extension:
[std-test]
exten = *601,1,Answer
exten = *601,n,Dial(IAX2/pbxnetwork/xx,30,m)
exten = *601,n,Hangup
exten = *602,1,Answer
exten =
hello, maybe quite off topic, but is there any way, how to do some like:
exten =
3010,2,Dial(SIP/3010/ALERT_INFO=normal_ringtoneSIP/3011/ALERT_INFO=beep_ringtone)
so, ring on two lines concurently, but using two distinguish tones (eg.
I would like to be informed, about incomming call for
CFN - Jan Serve wrote:
Hi all,
I have here several IAX2 Softphones(IDEFISK, DIAX and an own develop
based on iaxclient.lib). I have follow dialrules in my std-test extension:
[std-test]
exten = *601,1,Answer
exten = *601,n,Dial(IAX2/pbxnetwork/xx,30,m)
exten = *601,n,Hangup
exten =
Hi Tofik -
is there any open-source software that recodes g729 sound files to wav
sound files ?
The only way (at least) to do such transformation is with interactive
form on: http://www.asteriskguru.com/audio_conversion.php
The wiki also lists GX::Transcoder which looks like it can do g729
Is there an easy way to find out what ringtones a Cisco 79XX has
installed? I've tried going through the Telnet interface, but can't
find any lists of ringtones. Trying the code below produces a different
kind of ring, but not two short rings as indicated. I've also seen the
ringtone listed as
I have an Asterisk sever running with a
TE406P card, and 4 pri T1s.
I am trying to findout
how to send access codes to the switch. After a long distance call is dialed,
we get a second dial tone and I need to enter a 4 digit access code, then the
switch will place the call. Does anyone
Joshua Colp wrote:
Can you do an iax2 debug to see if packets are travelling when you
hear nothing?
Sure, but I not really can decrypt this:
- Executing Dial(IAX2/test-6, IAX2/pbxnetwork/xx|30|tTr) in new
stack
-- Called pbxnetwork/xx
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno:
Jeremy McNamara wrote:
Digium paid for ooh323, for whatever reasons that is beyond me, but it
has proven to be no better than any H.323 channel driver, so I hope
they
got their money back.
Better is subjective in this case. There's no doubt that chan_ooh323
has some warts. On the other
Gary,
What I do is the following:
In SIP.conf
Add the line accountcode= and set it equal to each users unique four digit
pin
example:
[user1]
secret=
accountcode=1234
type=friend
host=dynamic
context=default
canreinvite=no
nat=yes
qualify=2000
disallow=all
allow=g729
And in
Well, if I look at my tftp directory where the phone downloads its
config files, etc, on v7.1 I see a RINGLIST.DAT that contains the names
of the ring files to be downloaded. On my system that includes
ringer1.pcm and ringer2.pcm. I recall someone posting something about
how to generate the
On Wednesday 05 April 2006 16:42, Jon Weisman wrote:
And in Extensions.conf
exten=_X.,1,Prefix(${ACCOUNTCODE})
exten=_X.,2,Dial,Zap/g1/${EXTEN}
That won't work for this case, as he needs to enter the access code *after*
dialing. Right offhand, I can't think of doing anything other than
The resulting file is not going to sound any better and its going to
take up more space. What is the reason you need a WAV file? Perhaps
there is a more efficient way to do what you are trying to do.
Darrell S. Long
BestWeb Corporation
Tofik Suleymanov wrote:
Hello list,
is there
Dinesh Nair wrote:
more tests reveal that with ohphone, calls from SIP-ohphone work fine
with audio passed both ways. however when ohphone calls a SIP device,
the call is hungup when the SIP device answers.
This was sort of my problem too. I have two Asterisk servers, with an
IAX2 trunk
Hello all,
I am looking for a way to restrict users from logging in two
separate phones with the same authorization name/password at the same time.
Meaning, I only want users to be able to place a call from one phone in one
location, but have the ability to move from computer to computer.
Hi,
What does the following error mean:
Apr 5 12:39:40 NOTICE[22755] app.c: Huh? no dial for indications?
Here is the 'full' log around the error:
Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to
agent '3002', on 'Local/[EMAIL PROTECTED],1'
Apr 5 12:38:24
Hi
What causes deadlock?
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x82acb10', 10 retries!
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x8298160', 10 retries!
Here is the portion of the log:
Apr 5 14:02:42 NOTICE[23363] chan_zap.c:
does one know how to program so i can have 2 lines on one sip account on that phone ?im runnign my own asteriskdo i need 2 local accounts ? one for each line ? that rebounds to same SIP forp VOIP provider ?
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Title: WebMeetme Problem Please help!!!
I am running Feodra, I have downloaded the WebMeetMe Program, untar it to /var/www/html/WebMeetMe. I can access teh web page as of now. I cannot for the life of me figure out where defines.conf is. The install tells me it is in
I was just getting to work on fax for my * system, so I thought I
would bring everything up to date since there would be some new
compilations involved.
yum update gave me kernel-2.6.16-1.2069_FC4
but after recompiling zaptel, I kept getting FATAL module zaptel not
found
Chased this for
Bryan Mahin wrote:
Hello all,
I am looking for a way to restrict users from logging in two separate
phones with the same authorization name/password at the same time.
Meaning, I only want users to be able to place a call from one phone in
one location, but have the ability to move from computer
Andrew Kohlsmith wrote:
On Wednesday 05 April 2006 16:42, Jon Weisman wrote:
And in Extensions.conf
exten=_X.,1,Prefix(${ACCOUNTCODE})
exten=_X.,2,Dial,Zap/g1/${EXTEN}
That won't work for this case, as he needs to enter the access code *after*
dialing. Right offhand, I can't think of doing
Title: WebMeetme Problem Please help!!!
Sorry folks, my DSL took a bullet during a move this week
and I'm still trying to
get it back.
Now I do see one problem, the correct file is defines.php
not .conf. If my README
file points to .conf, I will fix that (but from memory I
don't think it
On Wed, 2006-04-05 at 17:54 -0400, Jimmy Smith wrote:
does one know how to program so i can have 2 lines on one sip account
on that phone ?
im runnign my own asterisk
do i need 2 local accounts ? one for each line ? that rebounds to same
SIP forp VOIP provider ?
Yes.
On Wed, 5 Apr 2006, Greg Oliver wrote:
On Wed, 2006-04-05 at 17:54 -0400, Jimmy Smith wrote:
does one know how to program so i can have 2 lines on one sip account
on that phone ?
im runnign my own asterisk
do i need 2 local accounts ? one for each line ? that rebounds to same
SIP forp VOIP
Ronald Wiplinger wrote:
I tried now many places to put these lines in. The system still
announces This card number is in use.
Can you give me a place where to put it in?
It's not receiving a card number.
Find the following 3 lines:
#
# At this point we have a valid card number.
#
Insert
Eric ManxPower Wieling wrote:
Bryan Mahin wrote:
Hello all,
I am looking for a way to restrict users from logging in two separate
phones with the same authorization name/password at the same time.
Meaning, I only want users to be able to place a call from one phone in
one location, but have
Is it possible for Asterisk to set the ptime attribute on outbound calls in SDP invite?
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Hi-
I'm a newbie to Asterisk, and in the process of setting up a working
system. I'm kind of stuck with a problem regarding the Digital
Receptionist, and I was hoping someone on this list might be able to
shed some light on whats going on.
So basically, I have the SIP phones/extensions and
An obvious typo in here... Here is the corrected version:
Here is what the extensions_custom.conf looks like:
-
[tsvxsj-in]
exten = 4081234567,1,Answer
exten = 4081234567,2,Wait(1)
exten = 4081234567,3,Background(pls-hold-while-try)
exten = 4081234567,4,NoOp(Incoming call on TelaSIP
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