Jessee,
Thank you for your help.
I downloaded firmware and sample configuration files.
But the firmware was old version for MP118 and MP124.
Where can i download recent one?
Can i upload only ini file to change
countrycoefficient ?
Regards,
Jason.
--- Jessee J Holmes [EMAIL PROTECTED] wrote:
Hi guys!
I'm having one or two issues with queues hosted by an Asterisk machine
where the clients are on a legacy PABX - at least for the interim. I
fully expect most of these issues to be non-resolvable, but thought I'd
at least ask to find out if there is some way of working around the
issues.
I've tried to transfer a call using the Flash command, but with my
configuration it doesn't work.
I have a traditional PBX connected with a zap channel to Asterisk that acts
like an IVR:
TELCO line -- traditional PBX (FXS) -- (FXO) Asterisk
From the TELCO line I can make a call to the
,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2006-11-08 21:44:38','49761450','49761450','83','from-internal',
'Local/[EMAIL
PROTECTED],2','','AGI','recordingcheck|20061108-214438|1162975478.49',0,0,'NO
ANSWER',3,'','1162975478.49
Zap show channels shows only 2 or 3 channels to be in use ... others are
not. Resetinterval is set to 1200 (20 minutes).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Forrest
Beck
Sent: Thursday, November 02, 2006 4:50 AM
To: Asterisk Users Mailing List
,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode,uniqueid) VALUES
('2006-11-08 21:44:38','49761450','49761450','83','from-internal',
'Local/[EMAIL PROTECTED],2','','AGI','recordingcheck|20061108-214438
|1162975478.49',0,0,'NO ANSWER',3
Hi Lee,
On Wed, 08 Nov 2006 at 09:00:27 -, Lee Archer wrote:
Are you using freePBX by any chance?
Yes, version 2.1.1.
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Hi, have a look at http://www.freepbx.org/trac/ticket/1174 it's
currently in the bug list.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nigel
Roberts
Sent: 08 November 2006 09:14
To: Asterisk Users Mailing List - Non-Commercial
Title: Message
Hi
everybody,
I've got an Asterisk
configuration where an agent handles calls from multiple queues. At the moment
I'm using the default Queue application and I encountered the following problem:
When there are calls waiting in multiple queues the selection of which call is
My Asterisk server is working fine, although every time that
in the middle of
any call there is a reinvite, the user hears a glitch. Why is
this happening?
How can I solve this problem?
That's because a REINVITE is generally used to change from one
codec to another. For some reason this
Brian,
I should concur with all that Dean raised.
Given the experience level you describe and the clear business case for what
you want to do, had you considered a commerical solution ?
It would give you the peace of mind that all will work. It will also allow
you to do many of the smaller
Hi everybody
Does anyone know if exists some CTI software than is
certificated by SAP?
Thanks
Sílvia Gallego Gonzalez
[EMAIL PROTECTED]
Optisistem:
Optimización de Sistemas Empresariales S.L.
Telf. (+34) 902 500 388
Fax. (+34) 93 217 67 77
In fact as far as I know, Asterisk stands in the middle of calls,
breaking one transaction and initiating another to the other side, doing
the bridge between them... Although good in some cases like permitting
to start a new transaction to the next hop changing codecs, in my case I
don't need
Hi Eric,
i have replied but nobody seems to got a deeper knowledge of the problem.
I have searched for talkoff, i found a lot of stuff, like check IRQs
(checked, and good) and/or set relaxdtmf=no (it is set)
or check the dtmf modes to be the same or or.
But nothing of the things i found
Hello,
here is our layout:
asterisk-A --- WAN --- asterisk-HQ --- WAN --- asterisk-B
calls are routed with SIP between asterisk's (found IAX to unreliable).
When asterisk-HQ attempts to native-bridge OR simply forward calls
between A and B no sound is sent.
If either leg (A - HQ or
For the benefit of the archives, my problem was a simple one.
I hadn't forwarded the IAX port on the router of the remote * server
connection, so when voip provider was trying to connect directly to the
remote * server, it couldn't.
Hurray for wasting an entire day over a simple silly little
Hello ppl,
Reading all over the net. Learnt quite a lot, but that has left me
confused-a-lot as well.
Need answers to a few questions. Before that, I have an ISP(fax gateway)
which will help me send/recv faxes using the T.38 protocol. I am using
Asterisk 1.2.12.1.
Now to the few questions I
Hi,I need to play a peep tone(to warn that he is going to another network) before ringing tone, when user is calling to mobile network. But peep tone must work under certain conditions, when destination is available( if unavailable - hangup).
Is it possible to do with asterisk?
HiCan you verify whether your PBX expects a hook flash for transfer or if it uses the Recall (or Flash) button on a telephone? Not an expert but I'm told by the real experts that they're different and my investigations
http://lists.digium.com/pipermail/asterisk-users/2006-November/171749.htmlshow
There is an option on the list server membership configuration screen
that will disable receiving your own posts to the list. Maybe the OP
accidentally disabled this feature.
Bob...
On Wed, 2006-11-08 at 06:09 +0200, Dovid B wrote:
I have seen this mainly with gmail. the logic is why do you
Hi Rick,
Well, if I told you I'd have to kill you
:-)
Seriously, taken from a very hidden Wiki page: http://www.voip-info.org/wiki/view/Polycom+reboot+hardphone+script
From the CLI: sip notify
polycom-check-cfg
xxx being the registered name in SIP.conf. It is meant for
the phones to
I recently ran fxotune against our incoming PSTN lines to try and help
with some echo problems.
It produced the following fxotune.conf file:
2=8,253,2,244,255,10,244,3,253
3=4,0,0,0,0,0,0,0,0
4=4,0,0,0,0,0,0,0,0
I'm a bit surprised by all of the '0's for channels 34, esp. given that
it's
Hi all,
I would like to use the Agent Login feature with
real-time queues it is not possible with asterisk 1.2, as described
here :
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue
The mantis bug describing the implementation of
realtime queue is bug 4037. This bug includes
Michael Sampson wrote ..
Say I have agents using a softphone like eyebeam that has 6 lines.
They
log in to the queue. Say there are 3 agents in my queue. 3 calls come
in
and all three agents are on a call. Now a fourth call comes in. Is it
possible to have it setup so that the 4 call rings
Stefan Agethen wrote:
Hi Eric,
i have replied but nobody seems to got a deeper knowledge of the problem.
I have searched for talkoff, i found a lot of stuff, like check IRQs
(checked, and good) and/or set relaxdtmf=no (it is set)
or check the dtmf modes to be the same or or.
But nothing of
This is not necessary - unless you are setting call-limit in sip.conf, and
don't have any patches on 1.2 to prevent app_queue from sending multiple
calls to the same member, they will automatically receive the call on the
second line appearance. (And third, and forth, and so on.)
Wes Baehr
It only happens when you go from IAX/SIP -- asterisk box -- aastra phone.
Doesn't happen PSTN -- asterisk box -- aastra phone.
The aastra people have said they believe it is a codec negotiation
issue... but the newest firmware didn't fix it send them packet
dumps.
On 11/7/06, shadowym
You'll need to use another key, instead of *. The * key is hard coded
for that hangup feature in queues.
On 11/7/06, Stefan Agethen [EMAIL PROTECTED] wrote:
I got an up2date Asterisk with SNOM360 as SIP and mISDN with 2 ISDN
Cards, if i press in a call the * Asterisk, Asterisk destroys the
Hello,
I need (some) help in configuring PAP2.
Best regards,
Twanny Azzopardi.
Mob: ( 356 ) 79713618
Email: [EMAIL PROTECTED]
Web: http://line.sytes.net
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Hello Alex,
I also apologize. I have now changed email on this list since I thought there
were problems with the previous one i was using. I actually got a copy of one
of my messages to my previous email and I thought why not get the others to. I
also checked the list archives but they weren't
Hi,
I have a system that connects to the PSTN.What do I need to do so
that when a call comes in, the system will start ringing the hunt
group I have setup but not actually answer the call? The problem is
the system is answering the call, and then passing 'ringing tones'
back to the caller,
On Mon, Nov 06, 2006 at 06:47:01PM -0600, Eric ManxPower Wieling wrote:
Louis-David Mitterrand wrote:
Hello,
Using a PRI (E1) with the euroisdn protocol, I don't seem to get any
specific message from the telco when attempting to dial a non-existent
number. Asterisk returns a
Hi,
I would like to know which is the lasted Asterisk 1.2.x version (branch
or trunk) for video support with h264 codec, and where I can downloaded it.
Thank You
Jorge Mendoza
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Talk to the folks at Asteria. The have a product called Reign. It
looks just like your old interface, runs off .NET as a client on the
machine.
http://www.asteriasgi.com/pbx/reign
On 11/7/06, Stephen Wingfield [EMAIL PROTECTED] wrote:
Andres,
The Bicom Systems Operator Panel is probably
Hello,
On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an
unalocated number? I always get 3 (no route) which is less than helpful.
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Hello everybody,
I'd like to hear some success stories about the use of Asterisk
Realtime in medium-large contexts, like 50 extensions.
Don't you think that in those contexts the system could be overloaded
from the excessive number of queries to the DB?
So.. is anybody using ARA in those kind
You did not mention what your FXO (connection to PSTN) hardware is???
Depending on what it is there may be configuration options for things like
'ring thru' and wether the fxo answers or passes the call to *
Doug
On Wed, 8 Nov 2006, Matt wrote:
Hi,
I have a system that connects to the PSTN.
I just installed the Digium asterisk-gui from svn on to an asterisk 1.4
beta3 configuration.
I can get to the main page, cfgbasic.html, and then log in OK, however
after I log in and then
each time I click on a new menu item I receive Stack overflow at line:
0. None of the data
Fields on the
Hi all
How much does configuring a network with VLANs improve or effect quality ?
Is there much reason to justify the configuration of VLANs ( I know
networking, but not VLANs at all)
Would it not be better to find high traffic users and determine why?
Your Thoughts
Thanks
Barry
Apologies.. we are using a sangom 4 port FXO card. It used to work
(or so the company claims that has the PBX), but they are saying it
stopped.. yet nothing has changed on the PBX system. I have verified
it IS picking up and then passing the call onto the ringgroup (hence
taking it out of the
Hello,
When i try to install the sfftobmp3.1, the tribbox box give me the
following error:
...
checking for TIFFOpen in -ltiff... yes
checking jpeglib.h usability... no
checking jpeglib.h presence... no
checking for jpeglib.h... no
configure: error: jpeglib.h not found
I try to find packages
What sort of interface are you using? Is there any way you could get diagnostics from the pbx? On 11/8/06, Andrea Giuliani
[EMAIL PROTECTED] wrote:I've tried to transfer a call using the Flash command, but with my
configuration it doesn't work.I have a traditional PBX connected with a zap channel
Why don't you post your configuration?On 11/8/06, Matt [EMAIL PROTECTED] wrote:
Apologies.. we are using a sangom 4 port FXO card. It used to work(or so the company claims that has the PBX), but they are saying itstopped.. yet nothing has changed on the PBX system.I have verifiedit IS picking up
So if I have notransfer=yes, why is it 'returning from native bridge'?
Nov 8 10:07:51 VERBOSE[21620] logger.c: -- Attempting native
bridge of IAX2/peer1-iax-10 and IAX2/peer2-test-14
Nov 8 10:13:06 DEBUG[21620] channel.c: Returning from native bridge,
channels: IAX2/peer1-iax-10,
Yet.. I am getting CDR records.. or am I misunderstanding what a
native bridge is?
On 11/8/06, Matt [EMAIL PROTECTED] wrote:
So if I have notransfer=yes, why is it 'returning from native bridge'?
Nov 8 10:07:51 VERBOSE[21620] logger.c: -- Attempting native
bridge of IAX2/peer1-iax-10 and
I took a look at this slide show and saw a lot that I like ... this level of
integration between voice and email has been along time coming and I think
it will eventually sell very well ... my initial reaction is, I WANT IT NOW
...
we use an Exchange 2003 server in our own shop ... I have
On Wed, 2006-11-08 at 10:15 -0500, Barry Fawthrop wrote:
Hi all
How much does configuring a network with VLANs improve or effect quality ?
Is there much reason to justify the configuration of VLANs ( I know
networking, but not VLANs at all)
Would it not be better to find high traffic
Anyone on the asterisk list have any thoughts about the new Open
Moko linux mobile?
http://www.theinquirer.net/default.aspx?article=35590
http://www.linuxdevices.com/news/NS2986976174.html
http://linux.slashdot.org/linux/06/11/08/004230.shtml
Is there any integration into
Jason,I think it's something only supported in the newer firmware. Get in contact with the place you bought the unit from, they should be able to get the latest firmware for you.MP118_SIP_F4.80A.034.004.cmp should work for MP114, this is what we used and all that we could get from Audiocodes, I
Hi All.
I'm using asterisk 1.2.10, 1.2.13 and 1.4.0-beta3 wth softphone eyebeam 15
I do the following:
eyebeam call to PSTN phone 911234567 and asterisk can't create a zap channel
sends CANCEL to eyebeam.
The log of eyebeam shows this:
[06-11-08]16:52:07.415 | Info (debug) RESIP:TRANSACTION |
What codec are you currently using for voice?
I have found that when nothing else works, playing with the gains on the
Zap channel helped. Usually lowering them.
I use rfc2833 for dtmf, alaw as codec.
Yes, a lowering could be a idea, but the problem is logged on any kind of
channels in my
Apologies.. we are using a sangom 4 port FXO card. It used to work
(or so the company claims that has the PBX), but they are saying it
stopped.. yet nothing has changed on the PBX system. I have verified
it IS picking up and then passing the call onto the ringgroup (hence
taking it out of the
I just found a link to this presentation
that has some more information
http://www.linuxdevices.com/files/article072/sld002.html
Cheers,
Dean
From: Dean Collins
Sent: Wednesday, 8 November 2006
11:16 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
The config is pretty simple.. when a call comes in it does an
Answer(), which obviously is going to stop the phone companies
no-answer-call-forward from working. My question, better perhaps,
is.. is there a way to cause asterisk to push the ringing through to
my ring group, without actually
Good Morning,
I've recently gotten Asterisk installed and configured our IVR using
FreePBX. Things seem to be going well except a few of our inbound
callers are ending up in the wrong place when trying to connect to a
specific extension. The example I had this morning was someone trying to
call
Hello everybody,
I'd like to hear some success stories about the use of Asterisk
Realtime in medium-large contexts, like 50 extensions.
Don't you think that in those contexts the system could be overloaded
from the excessive number of queries to the DB?
So.. is anybody using ARA in those kind
Title: Message
What
about creating _two_ appearances on the phone, one for each
queue?
-Original Message-From: Ardjan Zwartjes
[mailto:[EMAIL PROTECTED]Sent: Wednesday, November 08, 2006
2:20 AMTo: asterisk-users@lists.digium.comSubject:
[asterisk-users] Agents that handle
what is the sip.conf for 1239
which I'm going to assume is a extension on the TNT
Barry
JR Richardson wrote:
Hi All,
I have a lab setup with two asterisk servers and a MAX TNT in the
middle like this:
asterisk sip sip TNT pri pri asterisk
exten 1239 is the CID Number from the
When all else fails I resort to adding this in the sip.conf peer config:
Insecure=invite,port
It took me a while to figure out they can be used together.
Regards,
Scott
Thanks Scott, i have it set to that, but that has no effect. The
incoming call still requires proxy authentication.
On Nov 6, 2006, at 8:06 AM, Steven wrote:
Matt, How does one check for this??
You would probably know from the dmesg output card, just make sure
it's using the Octasic based echo canceler. I think it says something
about a VPM450M in the dmesg logs if it's the version I'm thinking of.
I get the same thing using inband -- funny thing I am the only one who
hears the random tones -- other party does not hear them and they are
not recorded with the monitor app.
on Wednesday 11/08/2006 Eric \ManxPower\ Wieling([EMAIL PROTECTED]) wrote
Stefan Agethen wrote:
Hi Eric,
i
You should get 3 if the number is not valid for any routing database.
You should get 1 if there is an athorative switch for that number, but it is
not assigned.
With DIDs.
you get 3 if the number has not been assigned to any telco.
you get 1 if it is assigned to a telco, but not an und user.
Ma
Hi Matt -
The config is pretty simple.. when a call comes in it does an
Answer(), which obviously is going to stop the phone companies
no-answer-call-forward from working. My question, better perhaps,
is.. is there a way to cause asterisk to push the ringing through to
my ring group, without
Matt wrote:
The config is pretty simple.. when a call comes in it does an
Answer(), which obviously is going to stop the phone companies
no-answer-call-forward from working. My question, better perhaps,
is.. is there a way to cause asterisk to push the ringing through to
my ring group, without
Ahh ok.. thanks.
On 11/8/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Matt wrote:
The config is pretty simple.. when a call comes in it does an
Answer(), which obviously is going to stop the phone companies
no-answer-call-forward from working. My question, better perhaps,
is.. is
I (and some others) are having an issue with placing calls on hold.
Our setup is as follows:
IAX2 or SIP terminator/originator --- asterisk box --- SIP Phones
I have tried asterisk verison 1.0.9, 1.2.6 and 1.2.12, and all have
the same issue.
When I place a call on hold that has come in a
On 11/8/06, Thomas Blanchin [EMAIL PROTECTED] wrote:
Hi David.
I read your post on :
http://lists.digium.com/pipermail/asterisk-users/2006-September/167456.html
I am in the same situation as you are. I'm looking for a way to
monitor iax2 connexions on asterisk. I'm using sipsak for sip
My iax.conf is:
[general]
bindport = 4569 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
delayreject=yes
disallow=all
allow=ulaw
allow=gsm
jitterbuffer=yes
forcejitterbuffer=yes
mailboxdetail=yes
dropcount=3
minexcessbuffer=100
Mine is a VPM400 on a TE410P (2nd Gen)
Purchased as a TE411P
--
--
Steven
http://www.glimasoutheast.org
Matthew Fredrickson [EMAIL PROTECTED] wrote in message news:[EMAIL
PROTECTED]
On Nov 6, 2006, at 8:06 AM, Steven wrote:
Matt, How does one check for this??
You would probably
Hi all,
Lets say I have my incoming calls transfered to my mobile phone. When a call
comes in, Asterisk will answer the call and ask the caller to hold the line
while the call is being transfered.
I know how to do this, but i dont want the caller to hear me answer the mobile
phone. They can
Seems like it is the IAX jitterbuffer. Can anyone offer any insight
as to why? If I turn jitterbuffer=no or disabled (comment it) then my
one way audio after hold issue goes away.
On 11/8/06, Matt [EMAIL PROTECTED] wrote:
My iax.conf is:
[general]
bindport = 4569 ; Port to bind to
Has anybody tried running Asterisk on Solaris on a SUN SparcStation ?
Or maybe the alternative of running Asterisk on a Linux Distro on a SUN
SparcStation?
I am asked to do this but I think it's almost impossible work to make it
happen.
Regards,
Jorge A.
On 11:06, Wed 08 Nov 06, Gary G. Hendershot wrote:
I think that all the individual pieces of what I saw in this slide show are
available NOW as open source components ... it looks to me like the big
challenge here is integrating all the pieces so they all work together to
provide the desired
Asterisk People,
I'm currently using Asterisk and with a SIP voip provider and I'm
having problems where DTMF input in my IVR app is getting corrupted
intermittently.
For example, if someone enters 1025, it may come though correctly as
1025, or it may come trough as 10025, or 100255. DTMF
After mocking up an unauthenticated call from a different device, a
spa942 phone, I found something strange in the SIP debug between the
phone and the TNT.
Asterisk is accepting unauthenticated calls as long as there is not a
user in the SIP header from the calling device.
Invite from the MAX:
Thanks Curt, thats too cool for school, any idea on when this is coming to the MS SBS platform?I use SBS for myself at home and would love that level of functionality included.Does Asterisk therefore handoff voicemail storage etc to Exchange for this level of
Justin Tunney wrote:
Asterisk People,
I'm currently using Asterisk and with a SIP voip provider and I'm
having problems where DTMF input in my IVR app is getting corrupted
intermittently.
For example, if someone enters 1025, it may come though correctly as
1025, or it may come trough as 10025,
Gang,
I'm having this error pop up when I do a ForkCDR, and I'm not sure how
to get around it. Here are a few log lines:
Nov 8 10:37:08 VERBOSE[28079] logger.c: -- Executing
ForkCDR(Zap/49-1, ) in new stack
Nov 8 10:37:08 WARNING[28079] app_forkcdr.c: Channel does not have a
CDR
The
I do not know when they plan on SBS
deployment of this. I wouldnt imagine it would not be soon because they
just released 2003 R2.
The biggest hurdle to this working with
Asterisk from what I understand is that it requires SIP over TCP. I havent
read the docs fully for 1.4 version of
Migrating to 1.4 is not an option. I don't know what that is, but I
doubt my voip provider supports it.
On 11/8/06, Kristian Kielhofner [EMAIL PROTECTED] wrote:
Have you tried 1.4 with vldtmf?
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Hello, list!
I'd like to create an extension that points to an offsite location (a
number on the PSTN), the purpose of which would be to see if that
offsite location is still on a call forwarded to it by Asterisk. This
way a receptionist could choose to transfer calls to a mobile phone
only if
Have a look at http://www.solarisvoip.comOn 11/8/06, Jorge Alayon
[EMAIL PROTECTED] wrote:Has anybody tried running Asterisk on Solaris on a SUN SparcStation ?
Or maybe the alternative of running Asterisk on a Linux Distro on a SUNSparcStation?I am asked to do this but I think it's almost
What is your queues.conf? Can you dial the user outside of a queue after they transfer the call?On 11/8/06, Rob Hillis
[EMAIL PROTECTED] wrote:Hi guys!I'm having one or two issues with queues hosted by an Asterisk machine
where the clients are on a legacy PABX - at least for the interim.Ifully
I am now scheduled to replace this with a TE412P with the VPM450M EC module.
Thanks for the heads up.
--
--
Steven
http://www.glimasoutheast.org
Steven [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Mine is a VPM400 on a TE410P (2nd Gen)
Purchased as a TE411P
--
--
the documentation for the sms app mentions a script that sends on the sms by e-mail once it's arrived
exten = _XX/_8005875290,n,System(/usr/lib/asterisk/smsin ${EXTEN:3})
Does anyone have any samples of such a script (it might be a good addition to the documentation) or does anyone have
Hello
I'm following instructions on how to install Asterisk on Fedora 5, but I'm
having a problem:
- the host is an older i686 athlon i386 GNU/Linux
- /etc/rpm/platform says athlon-redhat-linux
- running yum update kernel downloaded kernel i686 2.6.18-1.2200.fc5
- running yum update
Early dial is a feature on the phone that makes use of the 484 (Address
Incomplete) response.
This is desired for in-office, local (PSTN), and long distance dialing.
I'm really hoping to find a best-of-both-worlds solution to this.
Andrew Joakimsen wrote:
Does the GXP-2000 not have its own
After about one
weeks time I've gone from no VoIP to a completely configured system for two of
our offices to be able to page/communicate interoffice as well as handle
existing PSTN communications (okay, waiting onf hardware for the PSTN side and
I've likely jinxed myself now).
I was
Hi all,
I have now reinstalled my whole system because I had to change a few things
wiht my drives. Here is what happens. I have done apt-get build-dep asterisk
apt-get install linux-headers-2.6.17-2-686 which works just fine now.
Downloaded the latest files from digiums ftp.
First I unpacked
Three months ago, I was experiencing all sorts of issues with my Asterisk box maintaining a connection to multiple trunks, etc. I also experienced various timing issues as well. In addition, Asterisk would sometimes take almost a minute to fully load and register its SIP and IAX trunks.
Puzzled, I
The motherboard's capacitor? What is that? Since there are probably a
hundred or more caps on the MB, how did you determine that? Was it burned?
Other than that, without making either capacitance or noise tests I can't
imagine how you would make that assumption.
Doug
On Wed, 8 Nov 2006, Ronald
Hi,
I'm still pulling my hair out getting my queues setup in 1.2.13.
I went in to implement my custom roundrobinreset strategy (mentioned
in a post by me here:
http://lists.digium.com/pipermail/asterisk-users/2006-October/170713.html
and a similar issue is addressed by the developers back
Hi Alls,In Asterisk-1.4 there is new config file, users.conf, but i don't know how mechanism between users.conf and sip/iax.conf, usually i add new user in sip.conf, but when i try use asterisk-gui, it write to users.conf
and when i type sip list peer on asterisk console, there is no user that i
I have a debugging scenario where I wish to record the entire call. The
call is establish via a .call file. I can't seem to get Monitor to do
anything. My dialplan looks like this:
[dialout]
exten = s,1,DigitTimeout,1
exten = s,n,ResponseTimeout,10
exten = s,n,Answer
exten =
Always take your wedding ring off when working inside the box!!
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Steven
http://www.glimasoutheast.org
Doug Crompton [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
The motherboard's capacitor? What is that? Since there are probably a
hundred or more caps on the MB, how did
Hi
I have a * box with TE110P. When call comes in via ISDN without caller
id information, asterisk sets the caller id as Unknown. Is there any
way to change this? I've tried below but only works for calls with
caller id.
$AGI-set_callerid('74442932');
Thanks
Jay
Can anyone tell me what I have to do to get DID billing to word with
a2billing.
I am thing it may be context
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Are you outside of the US?
Do you need to call US Toll Free Numbers?
We can help you save money on
On 2006-11-08 06:26:45 -0800, [EMAIL PROTECTED] said:
Hello,
I need (some) help in configuring PAP2.
Try looking in sip.conf
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Christian wrote:<[EMAIL PROTECTED]>From: "Christian" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Wed, 08 Nov 2006 20:10:02 +0100Subject: [asterisk-users] talking caller ID Hi all,Lets say I have my incoming calls transfered to my mobile phone. When a call comes in, Asterisk will
Hey, Justin:
Justin Tunney wrote:
Asterisk People,
I'm currently using Asterisk and with a SIP voip provider and I'm
having problems where DTMF input in my IVR app is getting corrupted
intermittently.
For example, if someone enters 1025, it may come though correctly as
1025, or it may
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