I have seen mention of limiting a single server to no more than 100
trunks. With Sangoma having an 9 T1 card that kind of blows that limit.
How many T1's can we have on a single server assuming dual Xeons?
Our application calls for 122 IP Phones and 120 analog ports for faxes.
Its an executive
I keep running into the dead end that it can't find config.h in the source
tree.
It looks like newer kernels don't use it anymore.
Someone ran into this awhile back when compiling 1.2 and it looks as
though the issue was never resolved.
Any ideas?
Last time I tried this, I was on fedora core 5
Got it mate. thanx for that.
Am using mysql for voicemail storage, unlike in the script you've
written which works on mails on disk on a certain path.
All I've to do is query for INBOX(new) and Old(old) voicemessages count.
cheerz
-
Ben
Scott Keagy wrote:
A while back I posted a fully
I'm trying to pick up a ringing SIP phone (203) across the office with
exten = *9,1,Pickup(783743)
where 783743 is the local part of the number that our ISDN works on.
I tried all of these first:
exten = *9,1,Pickup(203)
exten = *9,1,Pickup(SIP/203)
exten = *9,1,Pickup([EMAIL
Gidean Chan schrieb:
Hi, How to stop Asterisk to pick up incoming PSTN signal but keep the
functionality to make the call out?
Thanks
Gidean
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Hi,
I am running asterisk-1.2.10 compiled from BRI-Stuff-0.3 on a Fedora
Core 5 box. We were having a problem with our firewall rules that
allowed signalling through but no media when the user picked up the
receiver. That has been solved now by our sys-admin guys but there were
calls (SIP to
What skills are needed to write a code yourself for X10, RS-485 or RS-232. I
am planning to learn some programming so I can do the stuff myself which
others haven't done yet. I once knew C/C++, and other electronic stuff, but
because of not using it for years, revise and update them.
Hello,
Are there any issues with Centos 4.4
and asterisk.
Thanks in advance
Varun
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asterisk-users mailing list
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1) You can connect the vonage lines to an FXO interface. I have a
customer who has the linksys router/ATA connected to FXO ports of his
nortel meridian PBX switch. You might try that with digium cards, FXO
port of SPA-3000 or some multiport FXO gateway.
2) Vonage softphone accounts work for
I have CentOS 4.4 on several boxes with Asterisk 1.2 and they run great.
Have not tested conferencing yet though.
Karl
Hello,
Are there any issues with Centos 4.4
and asterisk.
Thanks in advance
Varun
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Hello,
I got a pair of DigiNetwork X100P
FXO cards.
The packet has a installation with suggest
to install ' voicepet ' package.
Is it really required for setting up asterisk PBX ?
I am slightly confused.
Anybody knows why they ask to install that
package ?
Please guide ?
Thanks
Varun
varun wrote:
Hello,
Are there any issues with Centos 4.4
and asterisk.
Thanks in advance
Varun
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I'm currently running CentOS 4.4 64-bit with Asterisk with no problems!
Ove Aursand wrote:
varun wrote:
Hello,
Are there any issues with Centos 4.4
and asterisk.
Thanks in advance
Varun
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Hi,
In short - Asterisk is not able to recognize that the 'other' person to whom
call was made has hung up - hence the channel stays busy.
In long:
I've been able to configure asterisk 1.1.12 with freepbx 2.1.3 and Digium
TDM400 card (4 FXO ports). I've terminated 2 PSTN lines on these
David Parcerisa wrote:
Answer a call that is not ringing on your extension.
I want to pick up an external call that is ringing on another
extension that is not mine. Now in my old standard pbx I press 5 and I
get the call.
How to do this with asterisk?
See /etc/asterisk/features.conf
Louis-David Mitterrand wrote:
Short story: IAX is still crap in 1.2.13 (haven't tested 1.4), it's
unreliable and perfectly good hosts will become UNREACHABLE for no
apparent reason, while SIP connections keep going through.
Is this with or without the qualify= option in IAX2.
Roger Lewau wrote:
Resending this since I got no response
Hello list
We have a situation where calls need to be transfered to another extension.
We are using # to accomplish this but we found this is only working for
calls answered at the original called extension. If the call has been
Hi, How to stop Asterisk to pick up incoming PSTN signal but keep the
functionality to make the call out?
[from-pstn]
exten = s,1,Hangup
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On Tue, Dec 05, 2006 at 08:02:35AM -0600, Eric ManxPower Wieling wrote:
Louis-David Mitterrand wrote:
Short story: IAX is still crap in 1.2.13 (haven't tested 1.4), it's
unreliable and perfectly good hosts will become UNREACHABLE for no
apparent reason, while SIP connections keep going
On Tue, 5 Dec 2006, Tom Fanning wrote:
I'm trying to pick up a ringing SIP phone (203) across the office with
exten = *9,1,Pickup(783743)
where 783743 is the local part of the number that our ISDN works on.
I tried all of these first:
exten = *9,1,Pickup(203)
exten =
On Tue, Dec 05, 2006 at 07:57:27AM -0500, Zeeshan Zakaria wrote:
What skills are needed to write a code yourself for X10, RS-485 or RS-232. I
am planning to learn some programming so I can do the stuff myself which
others haven't done yet. I once knew C/C++, and other electronic stuff, but
I downloaded these 4 files:
app_rxfax.c app_txfax.c asterisk.patch spandsp-20061130.tar.gz
for use with asterisk 1.4 (these are the 1.4 files)
I installed spandsp, copied app_rxfax and app_txfax into
/asterisk-1.4beta3/apps
my question is what do I do with asterisk.patch?
I tried to put
anyone using/experimenting with this new feature in asterisk 1.4?
is anybody able to post some info how to use and what features are
supported?
I have general knowledge how SLA should work, ie. monitor status of
another line like BLF with additional features like answer ringing call,
barge
I suggest you get the code I mentioned in my last message - it is c/c++
code and as is usually the case with Linux, all the source code is there.
Looking at examples is a great way to learn.
Doug
On Tue, 5 Dec 2006, Zeeshan Zakaria wrote:
What skills are needed to write a code yourself for
Hi,
I had MAJOR problems with this. I ended up just putting the .so files into
the asterisk modules directory.
That worked for me. I can send you the files I used if it's any help to
you.
Regards,
Jim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thanks Karl.
On Tue, 2006-12-05 at 08:20 -0500, [EMAIL PROTECTED] wrote:
I have CentOS 4.4 on several boxes with Asterisk 1.2 and they run great.
Have not tested conferencing yet though.
Karl
Hello,
Are there any issues with Centos 4.4
and asterisk.
Thanks in advance
Varun
We are in the process of redesigning our single Asterisk server that
handles several queues for our clients. We offer our clients hosted
queueing/call center basic services. All the agents are in remote
locations behind NATs using either softphones or PAP2-like devices.
What we would like to
Dear all,
sometimes very strange things happen in * server at the office where i
work. E.g. sometimes a call is committed but it suddenly hangs up.
Like this output shows:
Executing Dial(SIP/502-9823, Zap/7/.|60|Tt) in new stack
-- Called 7/.
-- Zap/7-1 answered SIP/502-9823
Hi...
I just started working with Asterisk and found something that looks
like an error, but i want to be sure, so that's why I'm asking you.
When i make a call from A to B (both SIP clients), I don't see the
name of the called party in the phone that initiated the call, just
the dialed number.
And if you get someone over at Vonage that knows that to do you can
connect without the FXO
It is like FWD you have to get the KEY from Vonage for this to work.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VOIP PBX) 1-866-638-1254
(Voip PBX) Free World
Ok,
I have asterisk installed and downloaded a phone from xten.com. I just want
to get it connected so I can play and went into sip.conf and commented out
the xlite1 section. My xten phone keeps saying Awaiting proxy login info
but in there I have entered:
Username: guest
Password:supersecret
I have noticed this as well. I have seen a few configs like your DUNDi
setup, that use the fullcontact URI to directly contact a phone. I was
always puzzled how everyone was making this work with NAT.
I have not looked into it much yet, but I wonder if the new netfilter
SIP conntrack/NAT
Hello all,
I was wondering if anyone has had much experience with Realtime
Asterisk. I like the ability to setup my extensions and voicemail boxes
in MySQL, but I have a huge worry. What if MySQL crashes. I played with
rtcachefriends, but can't seem to find a way to have asterisk store the
Anyone know where I can get hold of this application (for 1.4 / trunk) ?
Julian
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Hello,
We have several regional asterisk's connected to a central one making
the the PRI calls through a TE410P card.
When using SetCallingPres(prohibited) on a call at the regional level,
that setting it not forwarded to the central asterisk and the call is
made as if no callerid had been
On 5 Dec 2006, at 16:39, blackwater dev wrote:
Ok,
I have asterisk installed and downloaded a phone from xten.com. I
just want to get it connected so I can play and went into sip.conf
and commented out the xlite1 section. My xten phone keeps saying
Awaiting proxy login info but in
I am not sure but i think that fix is for compiling zaptel not asterisk .
Asterisk runs on centos with 0 problems :)
On 05/12/06, varun [EMAIL PROTECTED] wrote:
Thanks Karl.
On Tue, 2006-12-05 at 08:20 -0500, [EMAIL PROTECTED] wrote:
I have CentOS 4.4 on several boxes with Asterisk 1.2 and
Hi guys,
just to let u known the pinout for the adapter :
Adapter for connect the E1 telco lines on my digium card
DB9 RJ45
3 1
8 2
2 4
6 5
Adapter used for connect the Hicom150 traditional PBX on my digium card
Is there anything special that needs to be done? I am trying realtime
voicemail and no matter how I set it up, be it a user in mysql or through
host access rights with no username/password all I get is err 1045 which is
access denied. I have all the mysql stuff installed and it writes to the cdr
Hi All,
I just noticed something interesting. When a sip device registers and
regcontext is setup in sip.conf, a NoOp priority 1 extension is
dynamically created in the dialplan within the specified regcontext.
Works great. If for some reason, modification is made to the
extension.conf and a
Let me guess: The context in which you have the 2 thru n priorities is
the same one as you're using for regcontext right?
Don't do that, bad things will happen (as you've noticed).
I'd have to review the code again, but I think what you're seeing is as
a result of this.
Regards,
- Brad
Hi Jerry,
Where did you find the 1.4 versions of this software? I don't see
anything on the official spandsp downloads site, just pre2 and pre3
releases, no 20061130.tar.gz :)
Thanks,
Matt G
On 05/12/06, Jerry Geis [EMAIL PROTECTED] wrote:
I downloaded these 4 files:
app_rxfax.c app_txfax.c
On 13:59, Tue 05 Dec 06, Watkins, Bradley wrote:
Let me guess: The context in which you have the 2 thru n priorities is
the same one as you're using for regcontext right?
Don't do that, bad things will happen (as you've noticed).
I'd have to review the code again, but I think what you're
Hi List,
I'm attempting to set up a queue and agents using agent call back. This is
all working fine with the queue and the agents login etc
However.
In my dial plan I a set variable when a call is entered into the queue to
identify the origin of the call, then when the agent is
Steve,
It seems like you don't have the full sources on your linux box.
Do you have a directory /usr/src/linux, which is a soft link to
/usr/src/kernel/2.6.18.-1.2849 ?
If not, or if the directory is empty means that you need to complete your
sources first.
I have this version of Zaptel
I'm trying to make a new box with Asterisk 1.4.0, work with one ATA
GrandStream 496 and G.726.
However I modified the rtp.c as suggested for the Sipura's ATA with
USE_DEPRECATED_G726=1 is not working.
Someone knows about this?
Thanks,
Carlos Alperin
On Tue, Dec 05, 2006 at 02:25:02PM -0500, Carlos Alperin wrote:
Steve,
It seems like you don't have the full sources on your linux box.
Do you have a directory /usr/src/linux, which is a soft link to
/usr/src/kernel/2.6.18.-1.2849 ?
Actually, the source is availble at
Dear List,
I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable
attended transfer feature. but i just can't do it work. I've already
set atxfer = * (and many other combinations) and all extensions on
extensions.conf have the t and T option. But when I'm going to test,
it doesn't
Thanks Tim,
I'm an asterisk newbie so am lost with the entire sip setup and conf files.
On 12/5/06, Tim Panton [EMAIL PROTECTED] wrote:
On 5 Dec 2006, at 16:39, blackwater dev wrote:
Ok,
I have asterisk installed and downloaded a phone from xten.com. I
just want to get it connected so
http://soft-switch.org/downloads/snapshots/spandsp/
;p
--
Humberto Figuera - Using Linux 2.6.17
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603
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On 12/5/06, Steve Gladden [EMAIL PROTECTED] wrote:
I keep running into the dead end that it can't find config.h in the source
tree.
I ran into this problem yesterday trying to compile ztdummy on FC6 i586.
The latest Digium tarball gave me the config.h error.
I was able to compile an SVN
Guh! :)
Thanks!
Matt G
On 05/12/06, Humberto Figuera [EMAIL PROTECTED] wrote:
http://soft-switch.org/downloads/snapshots/spandsp/
;p
--
Humberto Figuera - Using Linux 2.6.17
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA
On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote:
And if you get someone over at Vonage that knows that to do you can
connect without the FXO
It is like FWD you have to get the KEY from Vonage for this to work.
And more to the point there are so many VoIP providers out there,
most
I have not seen anybody on the web to have found this so I thought
I would check here. Anybody got this firmware? I've found
firmware for the 400, but it doesn't seem to load in the 410.
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Let me guess: The context in which you have the 2 thru n priorities is
the same one as you're using for regcontext right?
Don't do that, bad things will happen (as you've noticed).
I'd have to review the code again, but I think what you're seeing is as
a result of this.
Regards,
- Brad
Brad Templeton wrote:
On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote:
And if you get someone over at Vonage that knows that to do you can
connect without the FXO
It is like FWD you have to get the KEY from Vonage for this to work.
And more to the point there are so many
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michiel van Baak
Sent: Tuesday, December 05, 2006 2:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] regcontext,NoOp extension
vanishes when extension reload and doesn't
Brad Templeton,
Thats a very good point.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VOIP PBX) 1-866-638-1254
(Voip PBX) Free World DialUp: 780-217
WebSite: http://www.freeworlddialup.com/
We have Toll Free DID's instock
* * * NO MONTHLY FEE - LIMITED
Från: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] För Eric ManxPower
Wieling
Skickat: den 5 december 2006 15:06
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [asterisk-users] RESEND: Blind transfer # not working forforwarded
or picked calls
Roger Lewau wrote:
Resending
Thanks for all the feedback on the message, if i do the vonage integration
using FXo card, is there any possibility of working on G729 or GSM codec,
because linksys boxes by default use G711, which consumes hell lot of B/w.
Regards
Vijay Gandhi
-Original Message-
From: Al Bochter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
JR Richardson
Sent: Tuesday, December 05, 2006 3:49 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] RE: regcontext,NoOp extension
vanishes when extension reload
Let me
Hi,
I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem is still occur. My setup is as follows:
I've got Asterisk running on a high end Pentium-IV box running Linux
serving 5
On Tue, Dec 05, 2006 at 04:22:35PM -0500, Singer Wang wrote:
Hi,
I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem is still occur. My setup is as follows:
I've got
On Tue, 2006-12-05 at 15:32 -0600, Kyle Sexton wrote:
On Tue, Dec 05, 2006 at 04:22:35PM -0500, Singer Wang wrote:
Hi,
I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem
OK this was an easy one to fix. All I had to do is RTFM. Note on the wiki:
ATTENTION: Make sure you take a look at bug report 7144
Just do what Kevin said, include the regcontext in whatever static
context you have the priority 2 extension and don't make a static
regcontext in extension.conf.
I'm new to Linux, as I've been using Asterisk on FreeBSD via the ports
collection. My question is simple - for using the release branch of
Asterisk (1.2.13 for now), should I get in the habit of using svn to
retrieve the source or should I just download the tarball? Is there a
best practice or a
On Tue, Dec 05, 2006 at 04:50:22PM -0500, Phil Finkler wrote:
I'm new to Linux, as I've been using Asterisk on FreeBSD via the ports
collection. My question is simple - for using the release branch of
Asterisk (1.2.13 for now), should I get in the habit of using svn to
retrieve the source or
Okay, a bit more information..
Some more information:
the non connected problem only happens to about 5-10% of the calls, the
others go through properly.. and yes, for the rest both parties can talk
and hear each other..
asterisk version:
Asterisk 1.2.13 built by root @ [hostname] on a i686
You login to your vonage account on the web and set the bandwidth saver
option. That is the most you can do with a locked ATA.
Vijay Gandhi wrote:
Thanks for all the feedback on the message, if i do
the vonage integration using FXo card, is there any possibility of
working on G729 or GSM
Cisco PIX501 devices, The call center in India is running 5 agents using
PolyCom phones, and we're using G729 to save bandwith. And yes, we
purchused 5 licenses of G729 codec.
What firmware version on the Polycoms?
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Attended transfer is really four functions
1. Put the caller on Hold while you dial another number
2. Speak to the dialed number (announce the call)
3. Patch the call on hold to the other party using transfer button.
4. Disconnect (otherwise this would be a 3 party conference)
How these functions
This 24/7 mantra that companies keep promoting to us is often just the
ability to subject us to endless hours of their lame MOH while you wait
for the one service specialist to answer the phone from Tinbuckto.
My apologies if you live in Tinbukto.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Has anyone found a way to monitor a meetme conference for only
the first user? I find have one recording per user is pretty hard on the
server performance wise... Suggestions?
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010100|so |3|00|Platform: Model=SoundPoint IP 301,
Assembly=2345-11300-010 Rev=B
010100|so |3|00|Platform: Board=2345-11300-010 A
010100|so |3|00|Platform: MAC=0004f204e6de, IP=10.93.94.103,
Subnet Mask=255.255.255.0
010100|so |3|00|Platform: BootBlock=2.5.0 (11300_010)
A little more RTFM'ing and voila!
Using MeetMeCount I should be able to record only the first user.
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMeCount
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Connolly
Sent: Tuesday, December 05,
Hi...
I will send you the firmware for the 410 economy tomorrow... This
firmare can only be used for the 410 economy, not for any other 410,
400 or 420
2006/12/5, Brad Templeton [EMAIL PROTECTED]:
I have not seen anybody on the web to have found this so I thought
I would check here. Anybody
Hi All:
I'd like to use Realtime Static in terms of the performance concern
about dynamic realtime. Assume that I create a table:
as following:
CREATE TABLE `extensions_table` (
`id` int(11) NOT NULL auto_increment,
`context` varchar(20) NOT NULL default '',
`exten` varchar(20) NOT NULL
This is more of a MySQL question.. But its going to look something like:
ALTER TABLE `extensions_table` ADD `variable_name` type DEFAULT '0'
NOT NULL ;
From the specs page:
http://dev.mysql.com/doc/refman/5.0/en/alter-table.html
-Original Message-
From: [EMAIL PROTECTED]
Does there seem to be a popular Linux distro folks use specifically for
Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with
Linux distros. In particular, I'm looking for a free, stable, well
supported distro that has a friendly community. Any advice appreciated.
Sorry for
snip
I have tried setting another variable as a counter with some logic
tests to see the number of attempts to call the agent, but this is
failing as the variable appears to be lost when the call goes back to
the queue.
Local variables are destroyed once the call terminates. You'll have
I recommend debian, been using it for years now, it was a no brainer
to choose this for my asterisk deployments.. A few other people I know
have used debian with asterisk with no problems either.
On 12/5/06, Phil Finkler [EMAIL PROTECTED] wrote:
Does there seem to be a popular Linux distro
Hi List:
Can someone hlep to provide one or two examples to data entry for
sip.conf under the table structure?
CREATE TABLE `sip_conf` (
`id` int(11) NOT NULL auto_increment,
`cat_metric` int(11) NOT NULL default '0',
`var_metric` int(11) NOT NULL default '0',
`commented` int(11) NOT
On Tue, 2006-12-05 at 18:47 -0500, Mike Garey wrote:
I recommend debian, been using it for years now, it was a no brainer
to choose this for my asterisk deployments.. A few other people I know
have used debian with asterisk with no problems either.
Choose Debian, is easy to maintain..
On Tuesday 05 December 2006 15:36, Phil Finkler wrote:
Does there seem to be a popular Linux distro folks use specifically for
Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with
Linux distros. In particular, I'm looking for a free, stable, well
supported distro that has
I stand corrected!
However you do get my point ...
The bigger the company the worse it is. Having to deal with these guys is
a nightmare. The company that brings me out in spots is Rogers Cable
(24/7). They have this electronic air-head called Gertrude or something,
(an android) who can't
Henry.L.Coleman wrote:
Attended transfer is really four functions
1. Put the caller on Hold while you dial another number
2. Speak to the dialed number (announce the call)
3. Patch the call on hold to the other party using transfer button.
4. Disconnect (otherwise this would be a 3 party
Singer Wang wrote:
010100|so |3|00|Platform: Model=SoundPoint IP 301,
Assembly=2345-11300-010 Rev=B
010100|so |3|00|Platform: Board=2345-11300-010 A
010100|so |3|00|Platform: MAC=0004f204e6de, IP=10.93.94.103,
Subnet Mask=255.255.255.0
010100|so |3|00|Platform:
Question:
I'm using a .call file to make some test calls. The call file works
great. When I try the same thing with the manager 'originate' action I
get something weird - the originate action looks for the 's' extension
in my context, regardless of what I supply as the 'extension' argument.
The
Henry, according with voip-info.org, attended transfer is
While on conversation with another party, you dial the atxfer key
sequence. Asterisk says Transfer then gives you a dial tone, while
putting the other party on hold. You dial the transferee number and
talk with the transferee to introduce
The manager interface expects Exten NOT Extension argument header.
On 12/5/06, Michael Collins [EMAIL PROTECTED] wrote:
Question:
I'm using a .call file to make some test calls. The call file works
great. When I try the same thing with the manager 'originate' action I
get something weird -
Arlen Nascimento wrote:
Henry, according with voip-info.org, attended transfer is
While on conversation with another party, you dial the atxfer key
sequence. Asterisk says Transfer then gives you a dial tone, while
putting the other party on hold. You dial the transferee number and
talk with the
I have just installed Asterisk wit a TE110P card. I have configured 30
channels which seems to be recognised by staff and zap show channels.
I can make outbound calls with exceptional call quality but inbound
(receiving) calls the caller get a message saying Your call could not be
connected,
Thanks, Henry. It is very helpful for me.
I also deleted the DIAL option r in our dial out trunk which fixed the
problem.
Dial command option r: Generate a ringing tone for the calling party,
passing no audio from the called channel(s) until one answers. Without
this option, Asterisk will
Once again, it's time for the Melbourne VOIP get together.
Thursday the 7th is the December meeting of the Melbourne VOIP club.
We will be meeting at the usual place which is Pint on Punt, 42 Punt
Road Windsor.
It's near the big messy intersection of Dandenong Road, King's Way, Punt
Road,
But I never said ATA.
I said you call Vonage and tell Vonage that you want to B.Y.O.D. there
is a KEY you need Vonage to get you and install into Asterick for Vonage
service to work.
Buy like Brad said there are easier ways than Vonage.
I am not downing Vonage I have and still use Vonage and
Please hold :-)
Now you will listen to MOH for 4 days :-D
By the way you forgot one thing.. The person you get can't speak
English. :-(
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VOIP PBX) 1-866-638-1254
(Voip PBX) Free World DialUp: 780-217
Hi Guys
i've trying to set a mfcr2 systembut i can't find the working
combination between
-asterisk version
-spandsp version
-unicall version
i can compile but loading the pbx shows
[chan_unicall.so]Dec 5 23:04:42 WARNING[27865]: loader.c:325
__load_resource:
All,
Is there a way of rejecting a call using SIP in the Asterisk Dialplan?
Essentially, I want to look at the called number and if it matches
something I don't like I want to send back a SIP response which will not
cause the other end to 'hunt'. The response codes that will achieve
this
30 Channels on Verizon? Is this in the US? T1 (24 channels) or E1(30
channels)? Are you dialing from the top (g1) of the group or bottom
(G1)?
On 12/5/06, Klaverstyn, David C [EMAIL PROTECTED] wrote:
I have just installed Asterisk wit a TE110P card. I have configured 30
channels which
That kernel-devel fix is just for ZAPTEL. The bug has been solved in 4.4
On 12/5/06, Vicky [EMAIL PROTECTED] wrote:
I am not sure but i think that fix is for compiling zaptel not asterisk .
Asterisk runs on centos with 0 problems :)
On 05/12/06, varun [EMAIL PROTECTED] wrote:
Thanks Karl.
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