Hello everyone.
I had heard about this open-source PBX once a while back.
I wasn't too interested in it at the time but I kept the info filed away
for possible future use. A couple of days ago, I was walking around Barnes
and Nobles and I found this book, called Asterisk: The Future of
Recently started using * and really am having fun. One problem I
encountered...
I am using an SPA-3000 3.1.10d
When I have transfer enabled - 'T' in the dial string I cannot reliably
send DTMF keys to a bank, voicemail, or other service requiring tones. If
I disable (remove transfer option) from
Doug Crompton wrote:
I am using an SPA-3000 3.1.10d
When I have transfer enabled - 'T' in the dial string I cannot reliably
send DTMF keys to a bank, voicemail, or other service requiring tones. If
I disable (remove transfer option) from the dial string all is fine. I
would like to be able to
On Jun 3, 2006, at 11:13 PM, Vahan Yerkanian wrote:
Doug Crompton wrote:
I am using an SPA-3000 3.1.10d
When I have transfer enabled - 'T' in the dial string I cannot
reliably
send DTMF keys to a bank, voicemail, or other service requiring
tones. If
I disable (remove transfer option) from
Any C gurus out there that can tell me if this code compiled ok to be
used in x86_64 (Pentium Dual Core). It's for the app_conference
application.
Im using Centos 4.3 x86_64
kernel: 2.6.9-34.ELsmp
libgcc-3.4.5-2
gcc-3.4.5-2
after the compilation part is the makefile
begin
I tried various DTMF options and it made no difference. I purused goggle
on this and I did see other mentions pointing to Sipura and Asterisk as
the fault. I would have to say * is to blame as why would it work fine
without the transfer option. The DTMF is obviously getting through.
Perhaps it is
Matt Florell wrote:
If you are interested here is our altered app_conference code, tested
to work on Asterisk 1.2.8:
http://sourceforge.net/project/shownotes.php?release_id=421962group_id=95133
Hope this question doesn't seem totally idiotic: I can't devine from the
READMEs just what
Hi,
I have a strange problem on a single customer's PRI. He can't call
certain destinations, receiving an incompatible destination ISDN
cause code back from the network.
I'm sure that the PRI is misconfigured by the telco; but they (as
always) insist there is nothing wrong. Another Asterisk
Kevin, if I use ulaw for my sip users and my sip providers will i
minimize the transcoding hit to uncompressed mode to my server?
or will the load be the same even if I use g729 everywhere? Im trying
to optimize my setup as to do transcoding/uncompressing to a minimum.
On 6/3/06, Kevin P.
*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# ##
Attended Transfer *2
One Touch Monitor *1
Disconnect Call
I have for each phone such a paragraph in my dialplan.
I would like to save this by using a Macro. How can I do that?
exten = 8863959,1,Dial(SIP/8863959,60,r)
exten = 8863959,2,NoOp(${DIALSTATUS})
exten = 8863959,3,Voicemail,[EMAIL PROTECTED]
exten = 8863959,104,Voicemail,[EMAIL PROTECTED]
exten
I have followed these two for configuration of sangoma
A101
http://www.ss7box.com/s01_setup.html
http://www.ss7box.com/support_wancfg_1.html
on my side wanrouter star/restart is working fine
when i am tring to ztcfg -vvv i am getting
and when i am tring to load asterisk getting error No
such
/spool/asterisk/monitor/$1.wav
-- Executing System(SIP/attilla-8407, /etc/asterisk/mail.sh
CALL-008000200570-20060604-115659) in new stack
But this just doesn't work.
Asterisk says it's executing the script, but the mail doesn't get
send. If I check my log-files I don't see that the mail
I have followed these two for configuration of sangoma
A101
http://www.ss7box.com/s01_setup.html
http://www.ss7box.com/support_wancfg_1.html
on my side wanrouter star/restart is working fine
when i am tring to ztcfg -vvv i am getting
and when i am tring to load asterisk getting error No
such
On Jun 4, 2006, at 11:50 AM, Ronald Wiplinger wrote:
I have for each phone such a paragraph in my dialplan.
I would like to save this by using a Macro. How can I do that?
exten = 8863959,1,Dial(SIP/8863959,60,r)
exten = 8863959,2,NoOp(${DIALSTATUS})
exten = 8863959,3,Voicemail,[EMAIL
On Sun, Jun 04, 2006 at 05:50:41PM +0800, Ronald Wiplinger wrote:
I have for each phone such a paragraph in my dialplan.
I would like to save this by using a Macro. How can I do that?
exten = 8863959,1,Dial(SIP/8863959,60,r)
exten = 8863959,2,NoOp(${DIALSTATUS})
exten =
Hi,
* Matthias Fechner [EMAIL PROTECTED] [03-06-06 22:33]:
[portunity-in]
type=user
context=incoming-portunity
permit=82.139.223.1/255.255.255.255
disallow=all
allow=ulaw
sry that doesn't help.
ok correction, after forcing it to ulaw it is working fine.
Thx a lot!
Best regards,
Hi,
i try to make asterisk work behind a cisco pix 506. After deactivating the
sip fixup i´m able to register but I didn´t hear another party. It´s dialing
and connecting but silence. Does anybody has some tips or a sample config
for that issue ?
thanks
Andreas
Hi,
I've installed asterisk from the Ubuntu Dapper packages,
and it has been running for several weeks absolutely fine - calls work both in
and out (only using SIP on both sides).
I've come to setup Voicemail now, but when I dial the
extension with VoiceMailMain, I see:
-- Executing
i try to make asterisk work behind a cisco pix 506. After deactivating the
sip fixup i´m able to register but I didn´t hear another party. It´s
dialing and connecting but silence. Does anybody has some tips or a sample
config for that issue ?
allow any to asterisk?
--
Hi,
I need a few things modified on the current version of astcc. If there is
someone competent, please contact me off-list.
Regards,
Sahil Gupta
VoiceValley
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Ronald Wiplinger wrote:
What do I miss ???
Your current blind transfer setting is ##, so try ## 632 instead.
cYa,
Avi
--
National Manager - Special Projects
Sydney / Melbourne / Canberra / Hobart / London /
2/340 Gore StreetT: +61 (0) 3 9486 0411
Fitzroy, VIC F: +61
No, not working. Seems to be a prob w/ NAT
From: Woodoo People .pGa! [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re:
Hi Kamran,
your configs look fine, but your problem seems to be that Asterisk cant 'see' the card. What is the
output of cat /proc/interrupts.
I think you have not configured wanpipe correctly, thats why ztcfg wont work, which is why Asterisk
wont see the card. Here is my
Attilla De Groot wrote:
Hi all,
exten = h,1,System(/etc/asterisk/mail.sh ${CALLFILENAME})
But this just doesn't work.
Check the execute permissions on the script.
If that doesn't work, then echo the output of the script to a log and
review the log to find the error.
Doug
-- Ben Franklin
On Thursday 01 June 2006 17:06, Cyber Source wrote:
Hello All,
Complete newbie to asterisk (OH NO). Is it possible to use my skype
out account for an outgoing trunk? If so, can the syntax be found
somewhere? Thanks, Peter
Skype is not possible - however Michael Roberts has released a
I kind of assume from all of the mentions of speex in the code that it
is required.
MATT---
On 6/4/06, Brian Capouch [EMAIL PROTECTED] wrote:
Matt Florell wrote:
If you are interested here is our altered app_conference code, tested
to work on Asterisk 1.2.8:
Now that skype have published an API, I don't see why it should be
impossible to integrate using Asterisk as a proxy to a local skype
client on the server (similar principal to asterisk trunking). The
creation of a skype channel and using the skype API to create peers
might be able to do
On Saturday 03 June 2006 23:42, voiplist wrote:
So what are the smart folks doing when it comes to retricting/allowing
which area/country codes can and can't be called?
I use different contexts. i.e. the phones context defined in sip.conf only
includes contexts that define patterns for (say)
On Sunday 04 June 2006 11:46, Ronald Wiplinger wrote:
*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# ##
Attended Transfer *2
One Touch Monitor
At least two such Skype gatewats already have been written. I have tried PSWG but not been happy with the results. The principle works by decoding the Skype call using a Skype client (requires Windows PC with Skype logged in) and using some faux audio drivers in the host system to pass the
Hi,
I am having problems with asterisk (different
versions) leaking memory all
the time.
I tried Asterisk 1.2.1 and 1.2.7.1 but I am getting
the same behavior.
The machine has 3 telephony cards: 2 Voicetronix
Openline4 cards and 1
Digium TE110P card.
It is now running Asterisk 1.2.7.1 on Fedora
Hi,
Did anyone try to install Asterisk on the Mini-Box
M300 with a Versa
mini-ITX board 1GHz VIA x86 CPU?
The box looks promissing, but I am not sure if Digium
cards are compatible
with the mother board (Versa mini-ITX)
Also I am not sure if the 1GHz VIA processor can
handle a Digium 24 port
For those with an Fritz!board, have a look at:
http://www.fltronic.de/~olly/avm/
--
pgp-id: 926EBB12
pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12
Registered linux user: 75761 (http://counter.li.org)
___
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Can anybody tell me how I can key in # codes after the call is established?
All what happens now is that the call will be placed on hold and a new
call will initiate!!!
bye
Ronald Wiplinger
___
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Well this is exactly what I described in my message earlier today about
DTMF problees with SPA-3000 and the transfer option.
I think that the 'feature' transfer (and others) method, at least with
analog adapters like the SPA-3000, is prone to failure and difficult to
use. I don't see any easy
Enable the SIP fixup, and let the PIX handle the SIP/NAT issues. Do not
use the nat features in asterisk
forward all requests for UDP port 5060 to the asterisk box. also
forward all requests for UDP 1-2 to the asterisk box. On
asterisk rtp.conf file limit your RTP range 1-2.
On Jun 4, 2006, at 2:16 PM, Doug Lytle wrote:
Check the execute permissions on the script.
If that doesn't work, then echo the output of the script to a log
and review the log to find the error.
Doug
Hi Doug,
Well the permissions are set ok, but I just found out that the
problem is
- Erick Perez [EMAIL PROTECTED] wrote:
Kevin, if I use ulaw for my sip users and my sip providers will i
minimize the transcoding hit to uncompressed mode to my server?
or will the load be the same even if I use g729 everywhere? Im trying
to optimize my setup as to do
Sorry, it´s not working. When enabling the fixup I got a Wrong password.
Asterisk has private IP 192.168.100.160 and an official IP. I set a static
fom the official IP to the internal private one and allowed UDP 5060 and
1-2 to everybody
Attached the log:
Jun 4 18:20:38 DEBUG[1951]
Antoine Megalla wrote:
Hi,
Did anyone try to install Asterisk on the Mini-Box
M300 with a Versa
mini-ITX board 1GHz VIA x86 CPU?
The box looks promissing, but I am not sure if Digium
cards are compatible
with the mother board (Versa mini-ITX)
Also I am not sure if the 1GHz VIA processor
Attilla De Groot wrote:
Hi Doug,
Well the permissions are set ok, but I just found out that the problem
is that I give an argument in the system application. Because I say
System(script.sh ${var}), but if I do that variable static in my
script and then run it without argument, it's
Hello
How can we do statistics with asterisk
thanks
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To UNSUBSCRIBE or update options visit:
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Check these ones.
http://www.micpc.com/qloganalyzer
Queuemetrics
http://www.ag-projects.com/CDRTool.html
Take a look on voip-info.org for more options
Regards
issam escribió:
Hello
How can we do statistics with asterisk
thanks
/sh /etc/asterisk/mail.sh
CALL-008000200570-20060604-181756)
Call me stupid, but thats exactly the same.
Attilla
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http
On 6/4/06, Attilla De Groot [EMAIL PROTECTED] wrote:
Hi all,
I'm trying to make a context that will monitor a call and when it's
completed it would e-mail the wav to a specified mail adres.
So I made a standard context that records a call, like this:
exten =
Hi
Just started Learning OPENSER, installing at my office
Now iam able to install and configured with my VoIP provider
and iam able to make calls out, using Public IP, not tried yes NAT
when i want to voice mail, openser, send email,
But iam looking store and retrieve voice message, so i
I've recently noticed some oddities in my CDR records. In some cases
the original CallerID that I've set in the .conf file for the extension
showed-up in the CDR as the originating extension (on Zap/ devices on
the channel bank), and in other places it was the one that I set
using
Hello Josué,
the qsig feature came with Asterisk in version 1.2.x
Greetings,
Michael
Josué Conti schrieb:
Hello Michael, thank´s for help.
But what´s version asterisk you use? The qsig protocol supported for
what version?
Best Regards
Josué
2006/6/3, Michael Konietzny [EMAIL
Did anyone try to install Asterisk on the Mini-Box
M300 with a Versa
mini-ITX board 1GHz VIA x86 CPU?
The box looks promissing, but I am not sure if Digium
cards are compatible
with the mother board (Versa mini-ITX)
Also I am not sure if the 1GHz VIA processor can
handle a Digium 24
I have followed these two for configuration of sangoma
A101
http://www.ss7box.com/s01_setup.html
http://www.ss7box.com/support_wancfg_1.html
Why not go straight to the source and use the instructions at
http://sangoma.editme.com?
Nabeel
___
Attilla De Groot wrote:
On Jun 4, 2006, at 5:29 PM, Doug Lytle wrote:
I pass variables to my scripts all the time.
When I first started using scripts with variables, I was having
issues with the scripts running, but producing no results. Echoing
the variables to a log, I was able to track
Michael, thank´s for your attention.
If I will be able to help in some thing you can ask, ok? Has an excellent week.
GreetingsJosué
2006/6/4, Michael Konietzny [EMAIL PROTECTED]:
Hello Josué,the qsig feature came with Asterisk in version 1.2.xGreetings,Michael
Josué Conti schrieb: Hello Michael,
I was just wondering if there are any problems using the latest FreePBX with
SATA Raid 1 using hardware assisted software Raid like most modern chipsets
support?
I know that Digium and FreePBX were not recommending it awhile back but I
think that was based on 2.4 Kernel and Digium hardware
I have asterisk running more or less ok but I would like to turn off
call waiting and be selective about the incoming sip connections. This
is running asterisk 1.2.8 with a fxs and fxo card and a configured voip
(sip) line. Currently I'm using freePBX 2.1.1 to configure asterisk.
Problem 1)
I am running asterisk behind nat, and 2 sip phones on 2 different adsl
neted connections, asterisk is staying always in rtp media path, while
canreinvite=yes is configured in both extensions. I need asterisk
to stay away from the rtp media path, what is wrong with that setup?
Regards,
Osama Kamal
I believe that depends what you do with your Asterisk. If your hard
drives I/O is not intensive
then go for it. I got a SOHO Asterisk installed on Athlon XP 2000+ and
ATA 100 Software
RAID-1. The server is mostly idle but there are times it does some work
and it's also serving
as extremely low
On Jun 4, 2006, at 8:11 PM, Doug Lytle wrote:
Okay,
I decided that I would give it a try, I ended up having the same
issues as you. I got it figured out though.
When using the monitor application, it splits the wave files into
incoming and outgoing legs. Once the call has been
Debugs are showing that the request is getting there but the password is
wrong. Are u using cleartext or md5 hash passwords? I would double
check your secret= lines in sip.conf and make sure that the passwords in
your clients match.
Andreas Wilkens wrote:
Sorry, it´s not working. When
Hi, everybody:
I have looked at the Polycom entries on www.voip-info.org, and they're
outdated and convoluted and full of errors.
All I want to do is get my Polycom 501 to register with a working
Asterisk server. I want to do the configuration locally on the phone
through the console. (The
I use cleartext password. I double checked that is the right on several
times. When disbaling the fixup sip, the password is accepted.
From: Jason Bachman [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users
Attilla De Groot wrote:
On Jun 4, 2006, at 8:11 PM, Doug Lytle wrote:
Okay,
Your a real life saver. Thank you.
Something you may want to keep an eye on though. I'm not sure if after
the script completes, if it closes that shell. Maybe an exit in the
script for safety?
Doug
-- Ben
Hi All,
I need a function that I believe isn't available in Asterisk, but I
don't know if I'm correct about this.
I have a queue and I want agents that are in that queue to have the
ability to answer a call in the queue with calling an extention. For
example, if I'm an agent and my
Hi Stephen,
I use the 601's but I don't think they are THAT much different that
this information won't be helpful or get you in the right direction.
What is your network setup like? Are you using NAT or does the phone
have a public IP address? Also are you seeing any errors on the CLI of
Hi Attilla,
I'm not sure if there is something like that available or not, but I
know there are some alternatives. You can set the time out limit to say
15 seconds, which for me is about 3-4 rings on the phone before it goes
looking for the next agent. The other option you can manually remove
I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N
Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x
quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is
any registerable incoming volume from these lines. I've been running them
at
Hi, Kevin:
Kevin Smith wrote:
Hi Stephen,
I use the 601's but I don't think they are THAT much different that
this information won't be helpful or get you in the right direction.
What is your network setup like? Are you using NAT or does the phone
have a public IP address? Also are you
On Jun 4, 2006, at 10:33 PM, Kevin Smith wrote:
Hi Attilla,
I'm not sure if there is something like that available or not, but
I know there are some alternatives. You can set the time out limit
to say 15 seconds, which for me is about 3-4 rings on the phone
before it goes looking for
Attilla De Groot wrote:
Hi All,
I have a queue and I want agents that are in that queue to have the
ability to answer a call in the queue with calling an extention. For
example, if I'm an agent and my colleague forgot to logout I could
take the call when his phone is still ringing without
On 6/4/06, Andrew D Kirch [EMAIL PROTECTED] wrote:
I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N
Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x
quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is
any registerable incoming
What version of PIX IOS are you running. Some older versions had issues
with SIP.
Andreas Wilkens wrote:
I use cleartext password. I double checked that is the right on
several times. When disbaling the fixup sip, the password is accepted.
From: Jason Bachman [EMAIL PROTECTED]
Do anybody could compile app_conference on x86_64??? I tryied with two
versions of app_conference and got the same problem on compiling:
relocation R_X86_64_32 against `a local symbol' can not be used when
making a shared recompile with -fPIC
app_conference.o: could not read symbols: Bad
Stephen Bosch wrote:
That's the trouble. So many places to configure!
snip
Here's one source of confusion --
The parameters in the Server: ... category are
Address: [this is supposed to be the DNS or IP address of the SIP server]
Port: 5060
DNS Lookup: UDP only [I set this to UDP only
Hi:
I'm using a TDM-400 to terminate PSTN lines at my Asterisk server with
kewlstart signalling.
When an outside caller calls the server, the TDM-400 goes off-hook and
provides a ringing tone to the caller. If the caller hangs up before the
receiving party answers the phone, Asterisk fails to
On Sun, June 4, 2006 5:10 pm, Gonzalo Servat wrote:
On 6/4/06, Andrew D Kirch [EMAIL PROTECTED] wrote:
I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS
K8N
Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x
quad) FXO lines. Using ztmonitor (From
Attilla De Groot wrote:
On Jun 4, 2006, at 10:33 PM, Kevin Smith wrote:
Hi Attilla,
I'm not sure if there is something like that available or not, but I
know there are some alternatives. You can set the time out limit to
say 15 seconds, which for me is about 3-4 rings on the phone before
Hi Stephen,
Sorry if the e-mail is a bit choppy but I figured it would be best to
cut/paste answers in. Now again, I am using the 601's so things may be a
little different, but for the most part should be similar.
No NAT. This is just one Polycom 501 that is dialing out through an
Asterisk
Andrew D Kirch wrote:
I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N
Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x
quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is
any registerable incoming volume from these lines.
IOS 6.35
From: Jason Bachman [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] asterisk behind cisco pix
This is my makefile, it compiled ok. I will test it tomorrow but if
you have somewhere to test today, let me know.
# $Id: Makefile,v 1.9 2005/10/27 17:53:35 stevek Exp $
#
# Makefile, based on the Asterisk Makefile, Coypright (C) 1999, Mark Spencer
#
# Copyright (C) 2002,2003 Junghanns.NET
Ronald Wiplinger wrote:
Can anybody tell me how I can key in # codes after the call is
established?
All what happens now is that the call will be placed on hold and a new
call will initiate!!!
Just enter the required digits, just as if you are accessing voicemail.
Don't press the send
I too had the same problems. If you find out the best way for this let me
know!
Thanks
Curt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch
Sent: Sunday, June 04, 2006 4:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Andreas Wilkens wrote:
IOS 6.35
From: Jason Bachman [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users]
They key point is to disable de x86 CFLAGS and add this one
CFLAGS += -march=k8 -fPIC
k8 is the machine type for x86_64
On 6/4/06, Erick Perez [EMAIL PROTECTED] wrote:
This is my makefile, it compiled ok. I will test it tomorrow but if
you have somewhere to test today, let me know.
# $Id:
I also forgot to mention that our provider gave use three different IP addresses. One IP address for "Signalling" and two addresses for "Media." If this helps any.Thank you, StevenSteven Haldeman [EMAIL PROTECTED] wrote:Thank you for your response.All that I get when I dial in is all
Stephen Bosch wrote:
All I want to do is get my Polycom 501 to register with a working
Asterisk server. I want to do the configuration locally on the phone
through the console.
The console is very tedious. Why not use the web interface instead? Let
the phone get an IP address via DHCP and
Just make something like this:
exten = 8863959,1,Macro(dial,8863959)
[macro-dial]
exten = s,1,Dial(SIP/${ARG1},60,r)
exten = s,2,NoOp(${DIALSTATUS})
exten = s,3,Voicemail,[EMAIL PROTECTED]
exten = s,104,Voicemail,b$([EMAIL PROTECTED]
exten = s,105,hangup
Just to make a bit more
For Problem #1:
exten = _X.,1,SetGroup(${EXTEN})exten = _X.,2,GotoIf($[${GROUPCOUNT}= 1]?104:3)exten = _X.,3,Dial,SIP/usernameexten = _X.,104,voicemail(u${EXTEN})exten = _X.,105,hangup
This will limit the amount of incoming calls to 1 and send everything else to the VM.
For Problem #2:
I'm not
For Problem #1:
exten = _X.,1,SetGroup(${EXTEN})
exten = _X.,2,GotoIf($[${GROUPCOUNT}
= 1]?104:3)
exten = _X.,3,Dial,SIP/username
exten = _X.,104,voicemail(u${EXTEN})
exten = _X.,105,hangup
This will limit the amount of incoming calls
to 1 and send everything else
to the VM.
Hey. I was
Yes you are correct... by default asterisk will send the call to priority N+101... what is yourpoint?
You asked about turning off call waiting. In the example that I provided, if the amount of active calls is 1 then it will forward to VM without dialing the exten. That is what you asked for...
I'm having a related issue with could not fill input buffer and
submitted this bug report:
http://bugs.digium.com/view.php?id=7264
Rather then fixing volume issues, though, I'm trying to fix major echo
issues. [The card used to work fine. fxotune even fixed the echo the
first time I tried
Yes you are correct... by default asterisk will send the call to priority
N+101... what is your point?
You asked about turning off call waiting.
In the example that I provided,
if the amount of active calls is 1 then
it will forward to VM without
dialing the exten. That is what you asked
I have a problem receving calls via the ISDN line, using the followin
components
Asterisk 1.0.9 with [EMAIL PROTECTED]
chan_capi-cm-0.6
AVM Fritz card
datalink protocol = point to multimode
I can make calls out with no problems so the issue is only incoming calls.
When I make the call from an
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