hi guys,
i need to install the .tar.gz version of trixbox. i cant find any help files for installation in it and also there is no help for it on the website. can anybody please help? Thanx in advance--
RegardsRizwan HishamSoftware Engineer
___
http://www.trixbox.org/modules/smartsection/item.php?itemid=4 On 8/28/06, Rizwan Hisham
[EMAIL PROTECTED] wrote:
hi guys,
i need to install the .tar.gz version of trixbox. i cant find any help files for installation in it and also there is no help for it on the website. can anybody please help?
Hey guys,
I need some assistance in tracking down the cause of audio problems that
are occurring at two of my sites:
Both sites run Asterisk 1.2.10 and use Sangoma A101u PRI cards. Both
sites are reporting that audio in calls is dropping out during words,
so that the other caller (i.e. the
Hi Patrick,
thanks for your answer.
Unfortunately I cannot use misdn-init command because my distro has not
the lspci command misdn-init is based on. That's why I want to bypass
it. I'm doing all this mess because Debian Sarge installer does not work
with new asus motherboards, so I'm trying
Avi Miller wrote:
Does anyone have any suggestions on where to look next? My users are
getting increasingly annoyed and I'm quickly running out of ideas.
Replying to myself to note that this is now happening on outbound calls
via ISDN, i.e. calls that don't use IAX2 or the inter-office
Avi,
We need more info,
Through what means are both sides connected, 1:1 xDSL?
What bandwidth, are you using tunnels (pptp/gre/ipsec), how many concurrent
calls etc.
You could try analysing network delay/jitter/packetloss using Smokeping.
Note that on DSL 1 g729 calls uses about 45 kbit/s, alaw
-BEGIN PGP SIGNED MESSAGE-
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Avi Miller wrote:
Avi Miller wrote:
Does anyone have any suggestions on where to look next? My users are
getting increasingly annoyed and I'm quickly running out of ideas.
Replying to myself to note that this is now happening on outbound calls
On 2006-08-24 08:43:01 -0700, Ronald Wiplinger [EMAIL PROTECTED] said:
I want that each call from PSTN goes to Asterisk to the context for
this line. Within this context can be a menu or a dial command, ...
As more I read, as more I get confused, ... and each try is not working!
My sip.conf:
thanks Dovid, infact I just got things recorded from her.
Nitin
On 8/27/06, Dovid Bender [EMAIL PROTECTED] wrote:
snip
2) What are the best sources (cost effective) to get prompts recorded.
/snip
I would go with allison. She is the one that did all the voice files that
you currently have on
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Hash: SHA1
To a single extension?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brandon
Galbraith
Sent: Sunday, August 27, 2006 8:16 PM
To: Asterisk Users Mailing List - Non-Commercial
We have been lent a PORTech MV-370 mobile VOIP gateway, and can't seem
to get it to do outbound dialling properly.
Has anyone else used one of these successfully with Asterisk?
PaulH
AsteriskIT
___
--Bandwidth and Colocation provided by Easynews.com
On Mon, August 28, 2006 5:17 pm, Erik said:
Through what means are both sides connected, 1:1 xDSL?
All offices are connected via 512/512 SDSL.
What bandwidth, are you using tunnels (pptp/gre/ipsec), how many
concurrent calls etc.
No tunnels (that I'm aware of). Very few concurrent calls,
On Mon, August 28, 2006 5:21 pm, Matt Riddell (IT) said:
Are you using realtime?
No, the Asterisk boxes are managed by FreePBX which creates .conf files. I
have two boxes playing up (the ones with PRI connections). My other three
servers that use BRI are just fine. Calls between the other three
I would like to set up a simple PABX for use by IT students (1st year
univeristy level) to set up callcenter An suggestion for (low cost)
hardware configuration?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
I posted about this
a year or two ago:
Currently we can
find Cisco 801 ISDN Routers on ebay for about 40 euro. This equipment supports
RCAPI (CAPI over TCP) Think of it as a isdn-ethernet
adapter.
There are other
manufacturers who support this protocol in their ISDN Devices. (AVM, Bintec,
Avi Miller wrote:
Avi Miller wrote:
Does anyone have any suggestions on where to look next? My users are
getting increasingly annoyed and I'm quickly running out of ideas.
Replying to myself to note that this is now happening on outbound calls
via ISDN, i.e. calls that don't use IAX2 or the
Hi,
remote CAPI already exists for a specific protocol (Bintec). This protocol
is used by rcapidas well , a daemon which exports local CAPI ISDN hardware
via TCP using the bintec protocol. Using rcapid you can use the ISDN hardware
remotely within windows with the 'brickware' or from another
On Mon, August 28, 2006 8:10 pm, Rich Adamson said:
Is this a new installation, or, were the boxes working okay for a while
and they just now started having problems?
Its not a new installation: Calls have been fine for at least a month on
one server and about 4 months on another. Both servers
On Mon, 2006-08-28 at 09:08 +0200, Giorgio Incantalupo wrote:
Hi Patrick,
thanks for your answer.
Unfortunately I cannot use misdn-init command because my distro has not
the lspci command misdn-init is based on. That's why I want to bypass
it. I'm doing all this mess because Debian Sarge
We have a machine with a TE410P in it acting as a client to route calls
via iax2 to our central server,
caller -- ( zap - iax ) --- ( iax - whatever ) -- called
client server
often the called can't hear the caller (both machines on public ip)
'iax2 show
Hello Sergio,
please download and install the vmxml scripts again, there was a problem when
php was configured with register_globals=off. This is fixed now.
Please report success.
best regards,
Arnd
___
--Bandwidth and Colocation provided by
Hello,I have a call queue with ringall strategy. Users are IAX2 users. I would like to allow only one parallel call at all. I tried setting incominglimit=1 in iax.conf, but this did not help. I want queue to ring only when operator is not on line already with someone.
I tried creating Local
Simone Cittadini wrote:
We have a machine with a TE410P in it acting as a client to route calls
via iax2 to our central server,
caller -- ( zap - iax ) --- ( iax - whatever ) -- called
client server
often the called can't hear the caller (both machines on
Earl
Thanks for the answer, I am currently looking at it, and it looks like it does not solve all my problems (Asterisk-wise, that is), but it is surely a big step on the way.
Michael
2006/8/24, Earl Terwilliger [EMAIL PROTECTED]:
Michael,you might take a look at this, as it does most of that +
1
On 8/27/06, Brandon Galbraith [EMAIL PROTECTED] wrote:
Is there a maximum number of SIP devices that can be registered to an
extension?
-brandon
--
Brandon Galbraith
Email: [EMAIL PROTECTED]
AIM: brandong00
Voice: 630.400.6992
A true pirate starts drinking before the sun hits the yard-arm.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Actually, isn't there SLA work being done in the trunk right now?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Monday, August 28, 2006 9:16 AM
To: Asterisk Users Mailing List -
Your are right, I dont have to invent the wheel again, and I'm getting cleverer by looking at other peoples code.
But this does not solve my problems, I have worked in the PABX business as a software developer for about 8 years, and coming to * is not all that easy.
For instance, * does not
IIRC, you'll want to look at 'hint' extensions, and
possibly subscriptions to get status updates
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
MirSent: Monday, August 28, 2006 9:34 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
I'm trying to set up a system so I can do something like this.
Dial a feature code like *34
Get prompted for an ext.
Enter an ext.
Push 1 to set the MWI on that ext.
Push 2 to clear the MWI on that ext.
I need to know how to set and clear the MWI for a given ext.
I searched through the list,
As far as I can see on this web page
http://www.voip-info.org/wiki-Asterisk+video Asterisk doesn't support h264
codec. I can see the same on this pages http://www.asterisk.org/features
Question is, can I somehow enable H264 codec support in Asterisk? I have
Grandstream GXV-3000 video IP phone
remote CAPI already exists for a specific protocol (Bintec). This
protocol is used by rcapidas well , a daemon which exports local CAPI
ISDN hardware via TCP using the bintec protocol. Using rcapid you can
use the ISDN hardware remotely within windows with the 'brickware' or
from another linux
I'm trying to patch the Asterisk 1.2 source to support AEL2 as follows:
svn checkout http://svn.digium.com/svn/asterisk/branches/1.2
cd 1.2
svn diff http://svn.digium.com/svn/asterisk/branches/1.2
http://svn.digium.com/svn/asterisk/team/murf/AEL2-1.2 AEL.patch
patch -p0 AEL.patch
make install
Is there a way for me to grab the value of the authenticated mailbox from the
VoicemailMain() app?
If a user calls in to the main extension, enters in a mailbox and password and
authenticates, I want to know what mailbox number was authenticated for use in
another app. For instance, when the
Hi
I wanted to know if somebody solves this problem...
We have a commercial PBX and attached Asterisk as VoiceMail Systems.
Now in Switzerland all Voicemail start with Prefix 860 so if a customer dials
such a number, it can be either a mailbox on the asterisk, or a mailbox on a
foreign system
Michael Sampson wrote:
I'm trying to set up a system so I can do something like this.
Dial a feature code like *34
Get prompted for an ext.
Enter an ext.
Push 1 to set the MWI on that ext.
Push 2 to clear the MWI on that ext.
What are you actually trying to accomplish?
Doug
--
Ben Franklin
On Mon, 28 Aug 2006, Daniel Matos wrote:
remote CAPI already exists for a specific protocol (Bintec). This
protocol is used by rcapidas well , a daemon which exports local CAPI
ISDN hardware via TCP using the bintec protocol. Using rcapid you can
use the ISDN hardware remotely within windows
I'm trying to evaluate my path to several voip providers, so I set
qualify=400 in iax.conf. But, I'm not seeing any
REACHABLE/UNREACHABLE or LAG messages in the logs. Is there a logging
option to set so these will show up?
these messages are logged with verbosity LOG_NOTICE, so, in the
Dear All,
We need to do the following crazy scenario which is really stupid but
wanted :-((, I need to make the sip server initiate a call on zap
channels and once the phone answers, it should play an IVR and according
to the choice of the called he will be moved to other extensions, we plan
to
We use our own CDR, but as I understand, the C option resets the CDR,
that does not means is not going to save cdr, but is going to restart
the CDR. So, a simple NoCDR() before dialing should work, or ForkCDR()
and then NoCDR() if you want to save previous data.
Regards
On 8/27/06, Master Abi
I am new to asterisk. After studying the book and
the tutorial on the Internet, I am still confued about how to use context for
incoming calls.
Here is my question:
If I create s extensions in two different contexts
for incoming calls which one will be used? When a call comes in, which
On 8/28/06, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote:
Dear All,
We need to do the following crazy scenario which is really stupid but wanted
:-((, I need to make the sip server initiate a call on zap channels and
once the phone answers, it should play an IVR and according to the choice of
Does anybody know if the Cisco implementation is different from
Bintec?
I cannot tell for sure, but I doubt that since the Bintec protocol is
proprietary.
Historically which one came first? I would be very happy to know that
Cisco used the (apparently) open protocol from Bintec to
another phone system is being used for voicemail. This system is hooked
up to asterisk via a pri. it will dial into asterisk and set the code
when there is a voicemail.
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000
Doug Lytle wrote:
Just Curious,
Is all the messages or just a message now and then?
There was a bug fix in 1.2.10 for orphaned .txt files in the
/var/spool/asterisk/voicemail/.../INBOX directory.
How old is your asterisk core?
Like I said just curious and might be way off base. But I am trying to
track down the
Michael Sampson wrote:
another phone system is being used for voicemail. This system is
hooked up to asterisk via a pri. it will dial into asterisk and set
the code when there is a voicemail.
To turn MWI on:
touch /var/lib/voicemail/YourVMContextHere/${EXTEN}/INBOX/msg.txt
I'm not sure
Dear Friends,
One customer of mine has a line from Vonage connected to his Asterisk
box, he receive the following messages each 30 seconds in the CLI:
REGISTER attempt 1 to [EMAIL PROTECTED]
May you help me to understand what it means and how can I avoid this messages?
Thank you in advance,
Tomislav Parčina a écrit :
As far as I can see on this web page
http://www.voip-info.org/wiki-Asterisk+video Asterisk doesn't support h264
codec. I can see the same on this pages http://www.asterisk.org/features
Question is, can I somehow enable H264 codec support in Asterisk? I have
On Mon, 2006-08-28 at 19:17 +0400, Jean-Michel Hiver wrote:
Tomislav Parčina a écrit :
As far as I can see on this web page
http://www.voip-info.org/wiki-Asterisk+video Asterisk doesn't support h264
codec. I can see the same on this pages http://www.asterisk.org/features
Question is,
gc wrote:
I am new to asterisk. After studying the book and the tutorial on the
Internet, I am still confued about how to use context for incoming calls.
Here is my question:
If I create s extensions in two different contexts for incoming calls
which one will be used? When a call comes in,
Rushowr wrote:
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Hash: SHA1
Actually, isn't there SLA work being done in the trunk right now?
It doesn't work how you think it does, you can still only have 1 SIP
device registered to a SIP peer at a time.
--
Joshua Colp
Software Developer
Digium, Inc.
Juanjo Portela wrote:
Dear Friends,
One customer of mine has a line from Vonage connected to his Asterisk
box, he receive the following messages each 30 seconds in the CLI:
REGISTER attempt 1 to [EMAIL PROTECTED]
May you help me to understand what it means and how can I avoid this
messages?
Hi,
a few weeks ago someone mentioned a menu point 2 in the advanced options of
the voicemail menu, which allows a call back to the caller who left the
message.
I have two asterisk servers running but none has this second menu point.
Is this a feature which has to be enabled or did I
Race Vanderdecken wrote:
Is all the messages or just a message now and then?
Sergio mailed me and said he cant listen to any voicemail. That was
a stupid bug in our phpvoicemail script and not related to the infamous
orphant .txt bug.
There was a bug fix in 1.2.10 for orphaned .txt files in
I'm attempting to have multiple phones (geographically seperated) register to a single extension, so when the extension is dialed, any phone can pick up the call. Is this better handled by having each phone have a seperate extension, and handle the call routing in a dial plan?
-brandonOn 8/28/06,
Brandon Galbraith wrote:
I'm attempting to have multiple phones (geographically seperated)
register to a single extension, so when the extension is dialed, any
phone can pick up the call. Is this better handled by having each phone
have a seperate extension, and handle the call routing in a
Hi Hans
There is a SIP image now
But boy am I having probs with it..
Has ANYONE managed to get it to work with the latest SP 8.03 iamge ?
If so what did you use for the Loandinformation line (An example would be
great)
Thanks
Paul
- Original Message -
From:
Well, since you can technically only have one phone
registered to an extension, you'll need to do a simultaneous ring setup in your
dial:
Dial(SIP/1SIP/2SIP/3.)
I may be having a momentary brain freeze about the ''
but I believe that's right...
From: [EMAIL PROTECTED]
Hi list,
after too much time of googling and trial and error, I need some help.
In older Asterisk Versions 0.9 - 1.0 (Asterisk CVS-HEAD-02/13/05-15:26:28)
we used this setup:
extensions.conf
exten = 876779,1,AGI,reverse.agi| ${CALLERIDNUM}
exten = 876779,2,SetCIDName(Privat ${LONGNAME})
exten
On 8/25/06, Colin Anderson [EMAIL PROTECTED] wrote:
I don't see anything in there thatI'm not doing already (and have been for over a year, with 200 users)with Asterisk 1.0.9, HylaFAX, and Exchange 5.5, with the exception of the text-to-speech stuff which is do-able with Cepstral / Festival and
Hi, I'm using Asterisk in a Call Center.I put this patch http://bugs.digium.com/view.php?id=5577because if I have 2 calls and 2 free agents, I need that every call ring in differents agent a the same time. (Different from RINGALL)
If I put only this patch
You set which context will be used in you sip.conf, iax.conf, or zapata.conf
for the specific channel you are using. For example, my zapata.conf for channel
25 has context=incoming which means it should go to the context labeled
incoming in my extensions.conf. My PRI channels (1-23) have
How does it work?
Joshua Colp wrote:
Rushowr wrote:
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Hash: SHA1
Actually, isn't there SLA work being done in the trunk right now?
It doesn't work how you think it does, you can still only have 1 SIP
device registered to a SIP peer at a time.
Hi,
I'm using IAX2 to connect remote users to my asterisk server. Both
server and user are behind a nat. But sometimes the user registrates
correctly but sometimes doesn't.
Doing a debug i got:
Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13)
Acking anyway
Sending
Peder @ NetworkOblivion wrote:
How does it work?
Joshua Colp wrote:
Rushowr wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Actually, isn't there SLA work being done in the trunk right now?
It doesn't work how you think it does, you can still only have 1 SIP
device registered to a
This is why we set the SIP user ID to be the MAC of the device. It
helps us remember that EXTENSION != DEVICE.
Joshua Colp wrote:
Brandon Galbraith wrote:
I'm attempting to have multiple phones (geographically seperated)
register to a single extension, so when the extension is dialed, any
One more: be sure to set your webserver/php to register_globals off and
safe_mode off, example from apaches httpd.conf:
--
php_admin_flag register_globals off
php_admin_flag safe_mode off
--
Set this globally or in the virtual server config section. The scripts wont
work with most
On Sat, 26 Aug 2006, Kelvin Williams wrote:
If Asterisk was used to set up and tear down calls, and using canreinvite
allowing the RTP to pass from end-point to end-point, how many calls could
Asterisk handle at once?
I've pushed over 1,000 concurrent calls this way using the SIPP program
501s, 601s running 1.6.5
Asterisk 1.2.10
NAT
Logs at the bottom of the email
Using AMP or FreePBX for the config files
Heres whats happening:
Call comes in
Answer the call
On the Polycom
Hit Transfer (person calling in hears MOH just fine)
Enter park extension (my case
Hi,
It would like to know if it is possible to establish connection
asterisk (IVR) with traditional PABX.
My company possesss a common structure of PABX currently and is
needing to implement IVR with ASTERISK, but for the time being she
would like to keep the structure of the normal PABX.
It
On Mon, 2006-08-28 at 08:57 -0700,
[EMAIL PROTECTED] wrote:
I'm trying to patch the Asterisk 1.2 source to support AEL2 as
follows:
svn checkout http://svn.digium.com/svn/asterisk/branches/1.2
cd 1.2
svn diff
On Monday 28 August 2006 13:02, Greg Boehnlein wrote:
I've pushed over 1,000 concurrent calls this way using the SIPP program
for SIP performance testing. There was some tuning that needed to be done,
but it worked. Never went that far in production, though.
May you share some of your tuning
Yes it is possible.
May I suggest you spend more time with www.voip-info.org
Or even better download www.trixbox.org on an old server to get an idea of how
configs work.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
Dean Collins wrote:
Yes it is possible.
May I suggest you spend more time with www.voip-info.org
Or even better download www.trixbox.org on an old server to get an idea of how configs work.
Getting Trixbox would help him understand how Trixbox configs work, not
how Asterisk configs work.
Dear Dean Collins,
I am learning asterisk in the hand, without use of front-end. But
obliged for the tip.You it affirmed that it is possible, but as
serial? as I make it Asterisk to direct a linking for one definitive
common branch.
For example:I make a linking of the PTSN for mine I
I'm setting up my first (and very simple) Asterisk PBX and running
into problems with the FXO module I have on a TDM400P - I'm trying to
connect to a standard UK, BT, POT.
The problem is that when I plug the FXO module into a functioning BT
line, it seems to make the line become engaged - ie if
Arnd Vehling wrote:
Set this globally or in the virtual server config section. The scripts
wont work with most installations when safe_mode is off
^
wont work if safe_mode is ON!
Damn, not my day today.
___
--Bandwidth and Colocation provided by
This is what I have so far
[app-set-mwi]
exten = *35,1,Answer
exten = *35,n,Wait(1)
exten = *35,n,Playback(please-enter-yourextension)
exten = *35,n,Read(fromext,then-press-pound,,)
exten = *35,n,Wait(1)
exten = *35,n,system(touch
Actually Eric I disagree with you.
Through the use of config edit it allows you to look into each of the
conf folders to understand the layout of a multi channel in/out asterisk
server.
Using the voip-info wiki while possible is incredulous and difficult
(and some of it alarmingly out of date).
- Original Message -
From: Eric ManxPower Wieling [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, August 28, 2006 1:47 PM
Subject: Re: [asterisk-users] Asterisk with PABX
Dean Collins wrote:
Yes it is
Hi,
a few weeks ago someone mentioned a menu point 2 in the advanced
options of the voicemail menu, which allows a call back to the caller who
left the message.
Feature needs to be enabled in the voicemail.conf
callback=context
I've personally never used it.
well, this hint brought
kissing up some more
FYI, you both are right! Getting started with AAH/Trixbox can be very
valuable, but relying upon it can be very limiting.
/kissing up some more
I started w/ AAH, then went back and learned the dialplan apps,
scripting, etc. For some guys like me, it's easier to start with a
On Mon, 28 Aug 2006, Andrew Kohlsmith wrote:
On Monday 28 August 2006 13:02, Greg Boehnlein wrote:
I've pushed over 1,000 concurrent calls this way using the SIPP program
for SIP performance testing. There was some tuning that needed to be done,
but it worked. Never went that far in
This post spurred off of the comment of Michael Collins on
the Asterisk with PABX thread. I am going to post the relevant information
here:
I started w/ AAH,
then went back and learned the dialplan apps, scripting, etc. For some guys
like me, it's easier to start with a working (if
Michael Sampson wrote:
[app-set-mwi]
For some reason I get a busy signal when I dial *35 from an ext.
You need to do a:
include = app-set-mwi
In the context your phone is coming in from.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
At 11:16 AM 8/28/2006, you wrote:
Actually Eric I disagree with you.
Through the use of config edit it allows you to look into each of the
conf folders to understand the layout of a multi channel in/out asterisk
server.
IMHO: I started with AAH pre TrixBox and soon thereafter moved to a
You'll want to put them in the _additional.conf files,
because AAH/TB/FPBX doesn't always play nice with changes to the configuration
files that it modifies directly.
Rushowr / SKM
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Curt
ShafferSent: Monday, August
Hi Wederson,
- start reading about ivr and see the examples at voip-info.org.
- If you plans to use the analog extension on you pbx you need to use a fxo
adapter or a pc card (for example tdm400p) on * side, then you could finish
your ivr script with Dial application and call your old pbx. (very
My question to everyone is this..This is where I am
at now. I have been using FreePBX for about a year, after moving from [EMAIL PROTECTED] I am
starting to need some manual changes and modules. My question is can anyone
point me in a direction on how to learn how to create these. I
I believe that he meant _custom.conf
files.
I also use freepbx for my staff to create and edit
extensions.
But I have heavily modified the _custom.conf files
for anything that I "add" to the system functionality.
-- -- Steven
http://www.glimasoutheast.org
"Rushowr" [EMAIL PROTECTED]
Stefan-Michael. Guenther (in-put GbR) wrote:
Why does Asterisk strip all digits except 4498 and why doesn't _X. match
That I can't answer, I've never used the option.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve
I remember the config edit from [EMAIL PROTECTED] but I
do not have it on my freePBX now. I dont mind using vi, Im very
comfortable in Linux. Thanks for the answers!
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins
Sent: Monday, August 28, 2006 3:29
On 13:12, Mon 28 Aug 06, Michael Sampson wrote:
This is what I have so far
[app-set-mwi]
exten = *35,1,Answer
exten = *35,n,Wait(1)
exten = *35,n,Playback(please-enter-yourextension)
exten = *35,n,Read(fromext,then-press-pound,,)
exten = *35,n,Wait(1)
exten = *35,n,system(touch
We have been using Mandrake linux 9.2 with Asterisk for several years. We havejust loaded Debian both 2.4 and 2.6 kernels on a server for testing.After just a few hours or so, the asterisk(1.2.7.1) IAX2 driver locks up.Has anybody here that is using Debian seen this happen?Don
Hi,
Can anyone recommend a large button/type sip phone (VOIP) that an older
person could use. I have a client that needs to have large button phones
for elderly residents in her facility.
Thanks
___
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When using the # key identified in
features.conf this issue goes away.
Stilloddand so is the lack
of documentation on the built in Park button
Bill
From: Bill Gibbs
Sent: Monday, August 28, 2006 1:02
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject:
Chuck - Grandstream GXP-2000, while not designed expressly for the elderly,
does have relatively large keypad size. A better fit might be to simply
purchase a specialized, analog phone and pair it with an ATA to SIP enable
it. Companies such as the one below design phones specifically for this
Hi,
Say I have 2 queuesI have a bit of a problem.
I have 4 agents.
All 4 are on calls.
QUEUE1 has 0 calls
QUEUE2 has 1 call waiting for 24 minutes.
QUEUE1 now gets a caller.
Agent 3 gets off the phone.
After the wrap period.. Agent 3 gets the call from QUEUE1 which has
been waiting
When I use
exten = _70XX,1,NoCDR()
exten = _70XX,2,Dial(SIP/${EXTEN}|20|tr)
I get
Executing NoCDR(SIP/7002-081ac898, ) in new stack
Aug 28 15:27:18 WARNING[4670]: cdr.c:443 ast_cdr_free: CDR on channel
'SIP/7002-081ac898' not posted
Aug 28 15:27:18 WARNING[4670]: cdr.c:445 ast_cdr_free: CDR
Doug Lytle wrote:
Stefan-Michael. Guenther (in-put GbR) wrote:
Why does Asterisk strip all digits except 4498 and why doesn't _X.
match
That I can't answer, I've never used the option.
My VM works just fine by sending the callback through the same context
as what your sip phones use.
Chuck Bunn wrote:
Hi,
Can anyone recommend a large button/type sip phone (VOIP) that an older
person could use. I have a client that needs to have large button phones
for elderly residents in her facility.
How about the old Grandstream BT100?
Large buttons, requires a firm press (no
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