2006/3/29, John Novack [EMAIL PROTECTED]:
The reality is, of course, that telephone systems have provided thisfunction for many years. A DSS/BLF is available on MANY so called legacysystems, so until this function is readily available , customers that
require a receptionist will continue to go
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 wonder
if there is a way to check if this (provider) line is taken already.
How can I do that?
Same is with the phone.
Title: bristuff does not work with TDM400P
Hi-
we are having issues with quadBRI card which does not work together with TDM400P. We've tried to hunt
the problem and here is the scenario:
1) starting asterisk with tdm400P and two FXS modules (two phones)
2) pickup first phone and dial the
I had conversation with Welltech support and I got this description (I
can't send attachment through the list):
The 380x has a routing table function. There are two default route
exist in the routing table, one is for IP incoming call another is for
FXO incoming call, the IP call will be routed
Lonnie Abelbeck wrote:
asterisk at anime.net writes:
On Thu, 30 Mar 2006, Giridhar Reddy Bandi wrote:
I am looking at purchasing some DID lines from Teliax to install it on my
asterisk.
i would like to know some feed back on Teliax before i purchase.
suggest me if there are better sevice
Hi all
I complile asterisk 1.2.4 successfully.I install
festival successfully and i configure asterisk to work
with festival.But When i call the festival extension
configured in extensions, the festival application is
executed well (i see it in the log) and must read the
text (hello world).But
I am trying to write a outgoing Macro which has some sort of failover
for failing SIP connections.
For example...
Try Outgoing SIP Provider 1
- No Route to Destination
Try Outgoing SIP Provider 2
- Congested
Try Outgoing SIP Provider 3
- Success and connect..
Everything I try doesnt work.
Even
Has anyone linked the Asterisk to the Quintum Tenor DX4060? If
yes I would appreciate any valuable information to do this in anyway.
Cheers
Stephen
___
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Asterisk-Users mailing list
To
On 08:54, Fri 31 Mar 06, Olivier Krief wrote:
Why don't everybody use chan-capi ?
All our E1 interface use the zaptel driver, so impossible to
use chan_capi for them.
We use Sangoma cards, and the wanpipe driver for those cards
is a zaptel interface for asterisk, not a capi one.
There is your
On 03/31/06 00:28 Stefan-Michael. Guenther (in-put GbR) said the following:
To make it clear: We don't want to compare the three system against each
other. The asterisk server is running on a completely different hardware. We
what are the hardware and OS specs for the asterisk server ? this
Hi,
we are still trying to properly connect a Tenovis PBX to an Asterisk server
(asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this
time with QSIG.
Calling from a Tenovis phone to a SIP phone (i.e. traditional phone -
Tenovis PBX - QSIG - Asterisk - SIP phone) works with
Dear Group;
I have a requirement to pass the ${SIPDOMAIN} variable from Server A to
Server B over IAX2. Basically Server A is an Internal (*) and Server B
is an External (*) in the DMZ.
On Server A I do the following;
[SIPOUT]
exten = _6.,1,SetVar(DS=${EXTEN}%${SIPDOMAIN})
exten =
Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't
get it to work.
-David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian J.
M.
Sent: Friday, March 31, 2006 1:44 AM
To: Chris Earle; Asterisk Users Mailing List -
I know that astcc has that feature built in. In it you
specify your diffrent routes and the order. Have a
look at it.
--- Steve Ducat [EMAIL PROTECTED] wrote:
I am trying to write a outgoing Macro which has some
sort of failover
for failing SIP connections.
For example...
Try Outgoing
I liked the music
--- Andrew Latham [EMAIL PROTECTED] wrote:
hint... - listen to the queue for a bit
On 3/30/06, Melcon Moraes [EMAIL PROTECTED]
wrote:
Are you gonna answer me? I'm the first in line and
no answer! :)
[]'s
MM
-Original Message-
From: Steve Totaro
Asterisk 1.2.4 + bristuff-0.3.0-PRE-1l + libpri-1.2.2 +
zapte-1.2.3 +
*CLI zap show status
Description Alarms IRQ
bpviol CRC4
quadBRI PCI ISDN Card 1 Span 1 [TE] (caˇ OK 0
0 0
quadBRI PCI ISDN Card 1 Span 2 [TE] (caˇ OK
Hi Johann,
Johann Hanne [EMAIL PROTECTED] writes:
Hi,
we are still trying to properly connect a Tenovis PBX to an Asterisk server
(asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this
time with QSIG.
I tried with asterisk 1.2.2 against Alcactel 4400 on Monday. We had
Hi, sorry for this noop question,
but does anybody know how to cut the first digit of a variable?
example:
044612345
should be after cut operation:
44612345
My try in the extension.conf:
exten = _0[0-9].,2,Cut(mynum=EXTEN,/ ,1)
exten = _0[0-9].,3,Dial(Zap/g1/${mynum},90,T)
but it didn't work,
Hi, sorry for this noop question,
but does anybody know how to cut the first digit of a variable?
example:
044612345
should be after cut operation:
44612345
My try in the extension.conf:
exten = _0[0-9].,2,Cut(mynum=EXTEN,/ ,1)
exten = _0[0-9].,3,Dial(Zap/g1/${mynum},90,T)
but it didn't work,
Doug this is in no way an offense to you but I think
we need to start the asterisk booze fund. This will be
for all of us that have ups and downs in working on
getting asterisk set up. I for one have my friend
Johny Walker right by my side when ever it gets to
me.
--- Douglas Garstang [EMAIL
Christian Reelfs [EMAIL PROTECTED] writes:
Hi, sorry for this noop question,
but does anybody know how to cut the first digit of a variable?
example:
044612345
should be after cut operation:
44612345
Look at README.variables! It says:
,
| The format for removing characters from a
http://www.voip-info.org/wiki-Asterisk+variables
section: substrings
F.
Christian Reelfs [EMAIL PROTECTED] writes:
Hi, sorry for this noop question,
but does anybody know how to cut the first digit of a variable?
example:
044612345
should be after cut operation:
44612345
On Fri, 2006-03-31 at 14:03 +0200, Christian Reelfs wrote:
Hi, sorry for this noop question,
but does anybody know how to cut the first digit of a variable?
example:
044612345
should be after cut operation:
44612345
My try in the extension.conf:
exten = _0[0-9].,2,Cut(mynum=EXTEN,/
Thanks.
I think our problem ca be similar. Have
you tried to call from analog phone #1 to another analog phone #2? It works.
But when you try
to call vice versa from #2 to #1 it does
not work. When you restart asterisk it works again but only one
direction.
-David
From:
On Fri, 31 Mar 2006, Olivier Krief wrote:
2006/3/31, Armin Schindler [EMAIL PROTECTED]:
Yes, this is possible of course with the Eicon Diva Server PRI (T1) card.
This card provides a CAPI interface where you can connect Asterisk(with
chan-capi) and any other CAPI based application like
Armin,
Thanks a lot for the very detailed answer. I'll have to take a long look at
the CAPI interfaces and see how I can pull all this off, it's all very new
to me, but at least I understand that with an Eicon card, I could share a T1
between Asterisk and Hylafax. I'm not clear on whether I
Hello,
I bought a Grandstream Handytone 486 to forward
incoming calls from our old analogue PBX to the
asterisk server.
My first test was connecting an analogue phone to
the Handytone and calling a sip phone - worked.
Now I used the same cable to connect the line port
of the
Title: [Asterisk-Users] Howto cut the first digit
Christian Reelfs wrote:
example:
044612345
should be after cut operation:
44612345
My try in the extension.conf:
exten = _0[0-9].,2,Cut(mynum=EXTEN,/ ,1)
exten = _0[0-9].,3,Dial(Zap/g1/${mynum},90,T)
but it didn't work, my
snip
Can Asterisk serve as an access server/gateway to
the internet?
/snip
I have the same question. If I had a PRI coming in to
asterisk can I have users dial in and have asterisk
work as a gateway to the internet ?
Dovid
__
Do You Yahoo!?
We created out own. You should do the same.
--- Bob McDowell [EMAIL PROTECTED]
wrote:
The owner of my company just asked me for an
Asterisk brochure. Has
anyone seen such a creature? I know of some really
informative
websites, but I think a pdf would be priceless at
this point.
Hi,
I have a IAX2 trunk between two sites (connected with an high bandwidth
link) but sometime/often I get:
chan_iax2.c: Auto-congesting call due to slow response
and call is dropped (and routed on a PSTN link).
In iax.conf, I have:
[iax-out]
username=iax-in
type=peer
trunk=yes
It's look like this:
incomming connection
channel = Zap/7, dstchannel = SIP/200 (SIP hardphone)
when SIP/200 hangs up Asterisk do:
- Changing state for Zap/7 - state 1 (Not in use) and
- Changing state for SIP/200 - state 1 (Not in use)
but don't send ISDN-Q.931 DISCONNECT message to finally
Yes it will work but you have to set up the dial plan
to do what you want it to do. What exactly do you want
to happen when you use your soft phone ?
--- Kiffin Gish [EMAIL PROTECTED] wrote:
I have a Digium TDM400P card with 1 FXS and 1 FXO
module running on my
FreeBSD 6.0 server.
While I
Did you forget to nsert shamelessPlug ?
--- Melcon Moraes [EMAIL PROTECTED] wrote:
Sure!
In fact, there's a nice GUI for setting up all this,
called Phonecall.
Check it out in http://www.vecsector.com/phonecall/
You can do it all by your hands as well. :)
[]'s
MM
-Original
Hi!
I'm trying to enable echo cancelation with my Eicon Diva Server 4 Bri card.
I've enabled it from zapata.conf, (as I read from www.voip-info.org)
---
; Enable echo cancellation
echocancel=64
echocancelwhenbridged=yes
echotraining=2000
if the attachment has something else, please forward it to
eaperezh (at) gmail (dot) com
thanks for the info.
On 3/31/06, artifex maximus [EMAIL PROTECTED] wrote:
I had conversation with Welltech support and I got this description (I
can't send attachment through the list):
The 380x has a
I have same problem, do you have asterisk box behind nat?
PJ
Mimmus wrote:
Hi,
I have a IAX2 trunk between two sites (connected with an high bandwidth
link) but sometime/often I get:
chan_iax2.c: Auto-congesting call due to slow response
and call is dropped (and routed on a PSTN link).
In
[EMAIL PROTECTED] wrote:
Lonnie Abelbeck wrote:
asterisk at anime.net writes:
On Thu, 30 Mar 2006, Giridhar Reddy Bandi wrote:
I am looking at purchasing some DID lines from Teliax to install it
on my
asterisk.
i would like to know some feed back on Teliax before i purchase.
suggest me if
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Pavel Jezek
I have same problem, do you have asterisk box behind nat?
No, they are not behind NAT, peraphs there is a Checkpoint firewall.
DV
___
On Fri, 31 Mar 2006, Mike wrote:
Armin,
Thanks a lot for the very detailed answer. I'll have to take a long look at
the CAPI interfaces and see how I can pull all this off, it's all very new
to me, but at least I understand that with an Eicon card, I could share a T1
between Asterisk and
Don Pobanz wrote:
Adolfo R. Brandes wrote:
Lee Howard wrote:
However, based on the comments you give I'd suspect that you're
having what people seem to be calling frame slipping. There seem
to be some motherboards that react poorly with Zap cards (or their
respective drivers) and cause
Appreciate the replies everyone -- really
I'm wondering if I should be using zapHFC with my Junghanns card instead of
qozap? Everyone always mentions zaphfc -- mostly I guessed because they are
using a zaphfc-compatible card - but *maybe* I should try that instead
of qozap???
And yep --
That's not entirely correct :)
Fax and voice on the same DID is not possible when using a second
application like hylafax. Because how should the two applications
decide
which one accepts the call?
With the help of iaxmodem (which works really well) its easily done!
Just detect the incoming
When using Eicon Diva Server, you use chan-capi!
And the zapata.conf is not read by chan-capi. You need to setup
capi.conf. See the example and the README provided by chan-capi package from
chan-capi.org
Armin
On Fri, 31 Mar 2006, Giuseppe wrote:
Hi!
I'm trying to enable echo cancelation
Hi,
I have debug off (debug level 0) why are the following lines showing up
in '/var/log/asterisk/full'
Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match Found
Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on
Hi,
I want to build a PBX base on Asterisk using an embedded device.
Can anyone please recommend an embedded device I can use for doing so?
I will install linux or freebsd in the device.
Thanks
A
___
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Hi,
I want to build a PBX base on Asterisk using an embedded device.
Can anyone please recommend an embedded device I can use for doing so?
I will install linux or freebsd in the device.
Thanks
A
___
--Bandwidth and Colocation provided by Easynews.com
However, based on the comments you give I'd suspect that you're
having
what people seem to be calling frame slipping. There seem to be
some motherboards that react poorly with Zap cards (or their
respective drivers) and cause that. Your zttest results should be
Look what you have in /etc/asterisk/logger.conf
find:
console =
message =
full =
Hi,
I have debug off (debug level 0) why are the following lines showing
up in '/var/log/asterisk/full'
Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102:
Nathan Alberti wrote:
As i am using a central asterisk box with multiple stub sites I don't
wish every call put on hold to be wasting WAN bandwidth, I am wondering
if it is possible to create a multicast stream to each site and rather
than asterisk sending its address and the media
On 31/03/06, sam [EMAIL PROTECTED] wrote:
Hi,
I want to build a PBX base on Asterisk using an embedded device.
Can anyone please recommend an embedded device I can use for doing so?
I will install linux or freebsd in the device.
Depends what horsepower you'll need - many people have had good
On 03/31/06 19:49 Wolfgang Zweimueller said the following:
My conclusion with Q.SIG: do not use it at this implementation
level. YMMV.
i'll beg to differ. we've used Q.SIG successfully with an Ericsson MD110
for a customer in thailand.
--
Regards, /\_/\ All
I'm and [EMAIL PROTECTED] user - been so now for almost a year.
Lately, I've upgraded to the latest greatest.. (which is built on 1.2.5)
and am unable to install oh323.
I've already asked over at the ([EMAIL PROTECTED]) Sourceforge forum but no one seems to
think it worth answering.
The
On 00:01, Sat 01 Apr 06, sam wrote:
Hi,
I want to build a PBX base on Asterisk using an embedded device.
Can anyone please recommend an embedded device I can use for doing so?
I will install linux or freebsd in the device.
http://www.soekris.com
--
Michiel van Baak
http://gumstix.com/waysmalls.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of sam
Sent: Friday, March 31, 2006 8:01 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Building Asterisk embedded device
Hi,
I want to build a PBX base
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
wonder if there is a way to check if this
You cannot criticize Teliax until you investigate how your calls are
getting to them.
I have a customer on 17th St. in downtown Denver who use Qwest.net as
their ISP. They use Teliax (on 16th St.) as their ITSP. Piece of cake,
right?
This may have changed recently, but Qwest doesn't have
We use them for origination over IAX. At first we had callee's reporting
that our voice was choppy to them, while the callee has always sounded fine
on our end. I made that problem go away by introducing traffic shaping at
our firewall.
They have a bug, whereby on their website you can set the
http://voip-info.org/wiki/view/Easy+PABX
With Easy PABX you can create your own virtual PABX online in just minutes.
Easy PABX is based on Asterisk and best of all - it's completely free.
Regards
thorben.dk
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Hi,
Thank you that was it, I had 'debug' listed under 'full' in logger.conf.
Not sure how I missed that...
Thanks Again
Filip Drągowski wrote:
Look what you have in /etc/asterisk/logger.conf
find:
console =
message =
full =
Hi,
I have debug off (debug level 0) why are the following lines
Hi..
I have to setup an extension in a remote location that will use a
cordless analog telephone.. I am looking at the IAXY to do this for
me..Basically the data path will be as follows...
[Asterisk] == (NAT) == {Internet} == (NAT) == ATA -- Handset
Since there are two NAT boxes in the path
Hi,
Option 'e' is for
selecting an empty conference to join. My question is. How do I know what the
conference number is for the next party to join? Does it set it to a
variable?
___
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Hello Dinesh
I got a Panasonic KX-TDA100, can you tell me please how can you
configure the PBX side? Qsig slave? master? and the other side of the
asterisk? I got TE100P
Regards,
Daniel
Dinesh Nair wrote:
On 03/31/06 19:49 Wolfgang Zweimueller said the following:
My conclusion with
WipeOut wrote:
To save bandwidth I would like to stay away from using the G.711
codecs.. Does the IAXY support GSM or iLBC? I couldn't find anything in
the docs..
No. The IAXy only supports G.711 ulaw/alaw and ADPCM.
I don't know what 'docs' you were looking in, but this page:
[11747] logger.c: -- Executing AGI(SIP/451-0e31,
recordingcheck|20060331-165356|1143816836.643) in new stack
Mar 31 16:53:56 VERBOSE[11747] logger.c: -- Launched AGI
Script /var/lib/asterisk/agi-bin/recordingcheck
Mar 31 16:53:57 VERBOSE[11747] logger.c: recordingcheck|20060331-165356
Hi all
How can i decrease the speed of festival? It appear
that in festival, the text is read too fast for me
___
Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les
: -- Executing
GotoIf(SIP/451-0e31, 0 0?2:4) in new stack
Mar 31 16:53:56 VERBOSE[11747] logger.c: -- Goto (macro-record-enable,s,4)
Mar 31 16:53:56 DEBUG[11747] pbx.c: Launching 'AGI'
Mar 31 16:53:56 VERBOSE[11747] logger.c: -- Executing AGI(SIP/451-0e31,
recordingcheck|20060331-165356
Looking at the TE100P I don't see it listed Q.SIG as supported. We have it
working great as PRI. Am I wrong about the Q.SIG support?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Sent: Friday, March 31, 2006 9:04 AM
To: Asterisk Users Mailing
Hi Sebastian -
sorry for the long debug output below. I configured Asterisk with AMP to send
the whole number including the extensions of the callers to the called party.
Whatever I configure in AMP it looks like it is used, In my eyes it is ok, but
doesn't seem to work.
033811234451 is the
Kevin P. Fleming wrote:
WipeOut wrote:
To save bandwidth I would like to stay away from using the G.711
codecs.. Does the IAXY support GSM or iLBC? I couldn't find anything in
the docs..
No. The IAXy only supports G.711 ulaw/alaw and ADPCM.
I don't know what 'docs' you were looking in,
So, if you are absolutely sure that you've specified the correct T1synchronization parameters in your /etc/zaptel.conf and you still have
fax reliability issues, look elsewhere in your implementation for theroot cause.So, would you conclude that it's possible for a given T1/E1 to have incorrect
Hi,
Sometimes my Asterisk displays the following error
message...
Mar 31 12:53:04 WARNING[17170]: app_queue.c:519
statechange_queue: Failed to create update thread!
Has anybody seen it before?
Thank you
Dov
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Kai,
Thank you for the reply.
I didn't want to bother the list too much. However, after reading I discover
I dont have a clear cut way of doing transcoding.
Can somebody direct me to where I can get document to get this transcoding
done.
My set up
From [cisco (g729)] [asterisk (sip
Dinesh Nair [EMAIL PROTECTED] writes:
On 03/31/06 19:49 Wolfgang Zweimueller said the following:
My conclusion with Q.SIG: do not use it at this implementation
level. YMMV.
i'll beg to differ. we've used Q.SIG successfully with an Ericsson
MD110 for a customer in thailand.
Well, that's
Hi Avi -
I know this is off-topic for Asterisk, but I don't know where else to
ask: I've setup a central directory.xml file for my Polycom IP501 phones
with a list of all the internal extensions. None of them have sd1/sd
as I don't want to enable any speed dials, just have a list in each
Hi!
I'm here again with echo canceller problem... :-(
I think I've done everything to enable echo canceller feature, but it
still doesn't work...
Can anybody tell me if there is some error or something missing in this
configuration please?
I'm using Eicon Diva Server 4Bri.
On Fri, 31 Mar 2006, Boris Bakchiev wrote:
That's not entirely correct :)
Fax and voice on the same DID is not possible when using a second
application like hylafax. Because how should the two applications
decide
which one accepts the call?
With the help of iaxmodem (which works
NICE!
On 3/30/06, Joe Dennick [EMAIL PROTECTED] wrote:
I see (and like) the demo, but where can we get it?
Doug Lytle wrote:
Nicolás Gudiño wrote:
shameless plug Something like this perhaps?
http://www.asternic.org/stats/demo
O
VERY cool!
Doug
can anyone recommend a asterisk turn key company.
we will need the hardware as well as tech. support 24/7.
we'll want all the goodies, voice mail, auto attendant.
we have 6 incoming pot lines (all the same number), and 40 normal
telephones.
we have no interest in changing to ip phones or the
Doug this is in no way an offense to you but I think
we need to start the asterisk booze fund. This will be
for all of us that have ups and downs in working on
getting asterisk set up. I for one have my friend
Johny Walker right by my side when ever it gets to
me.
I'll second that. Maybe
Olivier Krief wrote:
2006/3/29, John Novack [EMAIL PROTECTED]:
The
reality is, of course, that telephone systems have provided this
function for many years. A DSS/BLF is available on MANY so called legacy
systems, so until this function is readily available , customers that
require a
Dovid Bender wrote:
snip
Can Asterisk serve as an access server/gateway to
the internet?
/snip
I have the same question. If I had a PRI coming in to
asterisk can I have users dial in and have asterisk
work as a gateway to the internet ?
Dovid
Very interesting question.
if this feature
Hi everyone,
I have been having some problems lately with our PRI and Asterisk, or
maybe it is just me. It happens once maybe twice a day, but when some of
our customers are calling in, the phone just drops on them. I pulled the
information below from the log from one that happened. I notice
I have four tenorAX boxes and there were way too many options that I would
never use. Quintum has great support so use them. Only really tricky parts on
the AX box was the dialplan section needed to be blank except min and max and
the unit ships with g729 enabled which I changed to ulaw.
Assuming your definition of 'auto attendant' is the same as my own, then
you betcha!
If I were building such a beast, I would use a different context for
each tenant that wanted a customized IVR and a public/generic one for
everyone else. You could use DID to route the incoming calls to the
Very true, but I want to 'sell' him on the idea, not drive him screaming
to Cisco...
Not my cup of tea, I'm afraid.
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid
Bender
Sent: Friday, March 31, 2006 6:43 AM
To: Asterisk Users
It's been a while, but I didn't think those two terms were necessarily
exclusive. Checkpoint firewalls can provide NAT, can they not?
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mimmus
Sent: Friday, March 31, 2006 7:35 AM
To:
I agree, this does work well. My 'fax' extension is right off of the
docs:
-
[faxin]
exten = fax,1,UserEvent(Incoming Fax...)
exten = fax,n,Dial(IAX2/ttyIAX)
exten = fax,n,Dial(IAX2/ttyIAX2)
exten = fax,n,Dial(IAX2/ttyIAX3)
exten = fax,n,Dial(IAX2/ttyIAX4)
exten = fax,n,Busy
exten =
Post your 'cat /proc/interrupts' for us.
Kevin Smith wrote:
Hi everyone,
I have been having some problems lately with our PRI and Asterisk, or
maybe it is just me. It happens once maybe twice a day, but when some of
our customers are calling in, the phone just drops on them. I pulled the
Kevin Smith wrote:
Hi everyone,
I have been having some problems lately with our PRI and Asterisk, or
maybe it is just me. It happens once maybe twice a day, but when some
of our customers are calling in, the phone just drops on them. I
pulled the information below from the log from one
Mar
Not a lot to go on sam. What do you want to do? If you just want to play
or have very minimal requirements then get a soekris NET4801 board, CF and
install Astlinux.
http://www.soekris.com/net4801.htm
-Original Message-
From: sam [mailto:[EMAIL PROTECTED]
Sent: Friday, March 31,
We're assuming you will use a T1 (or E1) for your PSTN interface. If
you're using POTS lines then there will be no information about which
number was called--you'll need a separate POTS line(s) for each tenant.
We have multiple tenants on our hosted PBX without problem.
I was given the
Hi Domenico -
I have a IAX2 trunk between two sites (connected with an high bandwidth
link) but sometime/often I get:
chan_iax2.c: Auto-congesting call due to slow response
and call is dropped (and routed on a PSTN link).
In iax.conf, I have:
[iax-out]
username=iax-in
type=peer
Hi,
I'm confused with agents and queues in Asterisk. If I use
AddQueueMember() then show queues shows the agents that I have
logged into the queue... however the agent ID has to be the extension
the agent is sitting at ... kinda useless for stats tracking.
If I use AgentCallbackLogin() then show
Just got a call from a company in Warren, MI
. They recently had an Asterisk system put in by a vendor, and are having
issues which need analysis and correction. They have a tremendous sense of
urgency. They have about (40) users, and need DIDs assigned to
extensions and are having some
Craig,
Please correct the date on your machine. Your emails stick to the top
of the list because they have a date of 6/30/2006.
Thank you,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
___
--Bandwidth and
seems that if you get that log you didn't use jitetr buffer at all.
In my opinion the latest jitter 1.2-branch is not working, the last working
seems 1.2.1 patched.
Hope Zoa could lead us to fix it.
Regards
Rosario
- Original Message -
From: Adam Moffett [EMAIL PROTECTED]
To:
Hey,
I would like in the course of dial plan logic, to trigger a separate
outbound call. If that outbound call is answered, and if that certain
key response is detected then it will bridge the incoming call to the
newly dialed outbound call.
What I want to accomplish is that when a caller
I have been evaluating the Iaxy and Asterisk for the company I currently
work for, and am rather impressed with them both. Once configured, the
Iaxy is a solid device--it's pretty much an appliance at that point
(plug it in, turn it on, and leave it alone).
My only gripe is the initial
Hi all,
Thank you for the reply.
I didn't want to bother the list too much. However, after reading I
discover I dont have a clear cut way of doing transcoding.
Can somebody direct me to where I can get document to get this
transcoding done.
My set up
From [cisco (g729)]
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