Roger De Salis wrote:
A further question...
How would you support multiple 12SP's on an asterisk install
if only one can be hardcoded into the source/binary?
Is this a mini programming project to add the magic number
to the /etc/asterisk/skinny.conf, and then have chan_skinny.c
search the config
Hi,
I have a basic dial plan question;
Here is the scenario.
Call comes through IAX and my * authenticate, then collect the digits and
dials out, simple :).
Here is the dial plan;
[did-in]
;for did callers
exten => 866219,1,Ringing
exten => 866219,2,Wait,4
exten => 866219,3,Answe
A further question...
How would you support multiple 12SP's on an asterisk install
if only one can be hardcoded into the source/binary?
Is this a mini programming project to add the magic number
to the /etc/asterisk/skinny.conf, and then have chan_skinny.c
search the config file?
Ta muchly.
Roger
Is app_meetme the only way to conference calls on a 7960 with * ? It looks
as if the 7960 would allow it, but * reports 'no compatible codecs'
- Chris Clifton
- Original Message -
From: "Brancaleoni Matteo" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, February 24, 2004
In my parking.conf, I have -
;
; Sample Parking configuration
;
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls; Which context parked calls are
i
I just placed a call to an 800 number via IAXtel and it went through fine. However, calling my FWD number does not. The console reports back "All circuits busy" Actually, I've seen this a lot with the bridge between FWD and IAXtel.
Michael
--Original Message Text---
From: Matt McIntyre
I am using following setup to dialout, I can take calls through sip-out
which is defined in sip.conf. My issue is cdr records will have a "s" for
destination. What Can be wrong and any suggestions to fix? Can this be a
bug I am using Version 0.7.2.
instead of macro-dialout if I directly dialed
By calling (248) 724-0700 and entering
the full IAXtel number at the prompt.
Matt
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop
Sent: Tuesday, February 24, 2004
10:58 PM
To:
[EMAIL PROTECTED]
Subject: Re: Spam Alert:
[Aster
On Tue, 24 Feb 2004, Greg Boehnlein wrote:
> Hello all,
> I have an application where I am attempting to use Agents and
> CallQueues to distribute inbound calls to remote users on cell phones. The
> system works quite well, except for one annoying thing that I cannot
> figure out. I have
Hi Matt
How do you reach a Iaxtel 1-700 from PSTN?
didn't think
this was possible ?
Barry
For the past few days I have not been able to reach my
asterisk server from the IAXtel PSTN gateway and just assumed there was a
configuration error on my part. I have however not been able to find the
problem and according to my asterisk console, I am not registered due to
timeout. I can n
VOIP services are illegal I thought in the Bahamas according batelco
:)
What island are you on?
Have fun finding some.
Dave P
>>> [EMAIL PROTECTED] 2/24/2004 4:16:59 PM >>>
Need SIP or H.323 origination from Bahamas ASAP. Can someone
provide an
access number or Toll Free number origin
On Tue, Feb 24, 2004 at 04:33:41PM -0500, mattf wrote:
> Take a look at my GUI app:
>
> http://sourceforge.net/projects/astguiclient/
>
> It'll run on Linux and Windows, it's written in perl and it'll list every
> channel(Zap/SIP/Local) that is active on your system updated every second.
>
> You
Can anyone give me a high level technical summary of how the Avaya IP phones
function with the Avaya IP pbx?
Need to implement QoS in the infrastructure, and later attempt interconnecting
the Avaya system to *.
I've got a sniffer trace of the phone bootup, an actual phone to phone call,
etc.
On Tue, 24 Feb 2004, Clif Jones wrote:
> Gee, maybe I'm missing something, but the spec does not say that.
I copied&pasted the wording bellow from RFC3261.
> The
> RFC actually says that
> when you send a final response, you are required to store that final
> response for 64*T1 seconds
> and ret
Hello Klauss:
Could you please let us know the price of your Quad-BRI card?
Thanks,
Carlos
Klaus-Peter Junghanns wrote:
Am Fr, 2004-02-20 um 15.22 schrieb Armand A. Verstappen:
On Fri, 2004-02-20 at 09:17, Klaus-Peter Junghanns wrote:
to clear things up again, the problem is a wrong synt
Hello everyone,
I'm trying to setup the T1 line with T100P card but
could not get it accepts inbound calls. I only got busy signal.
Anyone know what would be the dialplan (extension.conf) to accept T1 line
calls.
Thanks.
-Tri.
I believe this is a PPP issue. Either make sure you have
ppp support in your kernel (even as a module) or look in the
Makefile for 'PPP'. It should be obvious when you see it.
Tim
On Tue, Feb 24, 2004 at 05:47:24PM -0500, Mark Phillips wrote:
> Hi folks,
>
> I'm trying to compile the zaptel driv
We are using IAXphone in a production environment with hardphone backup
(BRI).
Most things work nicely, but we have found that we can only transfer each
call once.
I.E. Incoming call comes in on the [incoming] context, receptionist
transfers it to a local extension in the [international] context.
> > Then when the receptionist is on the phone, they could hit the services
> > button, scroll through the list of extensions; see what the persons
> > status is, and even transfer the call right through by pressing the Dial
> > button underneath the extension if they hit Transfer before the Servic
> I am positive I saw add-on modules for Cisco phones to show line/extension
> status, though.
Your did
http://cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008
008883d.html
alas no one has done any integration work with * yet
___
Ast
Hi!
> I am positive I saw add-on modules for Cisco phones to show
> line/extension status, though.
It appears that for the SNOM 220 there is a 20 key extension pad, and you
can use up to 3 of those per phone --> maximum of 60 lines to monitor?
Cheers, Philipp
__
> Then when the receptionist is on the phone, they could hit the services
> button, scroll through the list of extensions; see what the persons
> status is, and even transfer the call right through by pressing the Dial
> button underneath the extension if they hit Transfer before the Services
> but
Just a thought on this topic, if you're using Cisco 7940/7960 IP handsets it
should be possible to write a Perl script on the Asterisk box that updates
the XML directory to show the current status of extensions.
Then when the receptionist is on the phone, they could hit the services
button, scroll
On Tuesday 24 February 2004 16:47, Mark Phillips wrote:
> When I run the zaptel Makefile I get the following unresolved
> symbols;
>
> /sbin/depmod -a
> depmod: *** Unresolved symbols in
> /lib/modules/2.4.22-10mdk-p3-smp-64GB/misc/tor2.o
What's the output of '/sbin/depmod -ae' ? That should gi
Hi
I have the same setup as You. Here are my sip.conf , extensions.conf and
modules.conf
that is currently on my system.
Just copy and paste, change numbers and reload..
This is my sip.conf which I have on my system and it is working. It is a
modified eksample file
; 2003-04-24 05:06 GM
Jean-Denis,
Great! Why didn't I think of that! Merci beaucoup, I'll try it out
tomorrow (it's a little too late now ;-)).
I did some tests on my own, and came up with another method that seems
to work, based on GoToIf() and the DNID variable. It goes something like
this:
[isnd-in]
exten => s,1,G
Is there a way to change the standard codes (US) to something else (EU)
codes.
Regards
Jan Larsen
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http
Need SIP or H.323 origination from Bahamas ASAP. Can someone provide an
access number or Toll Free number origination from Bahamas?
Thanks,
Aram
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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Brian Capouch a écrit :
Sales wrote:
Curious if anyone has any feedback on Nufone voip pbx.
Perfectly happy customer. Most of my customers use NuFone as
well--perhaps a dozen of us all told.
Excellent uptime, reasonable rates. Email-based customer service for
the most part. Others are bot
Robert Sprockeels a écrit :
Hello,
I looked for this in the archives and docs, but could not find anything,
so here goes...
I have * running with an Eicon.Diehl Diva (passive) ISDN BRI modem and
SIP and IAX2 phones. For the modem I used chan_modem. It is not really
clear to me if chan_capi will wo
Hi folks,
I'm trying to compile the zaptel driver and am getting nowhere.
I've got 2.4.22-10mdk sources loaded.
I've edited the Makefile to reflect my multiprocessor kernel (although I'm
only running 1 P4 - Mandrake 9.2 installed it this way) and I've run a
make menuconfig in the linux source ar
I hope, you do not mind few questions from the newbie:
As a first after installing the Asterisk I try to make it working with two
PCs and Xten-Lite installed.
Both PCs have fixed IP addresses but they log in to Asterisk only if
configured in sip.conf with host=dynamic. Why?
I have them defined in
Hello,
I looked for this in the archives and docs, but could not find anything,
so here goes...
I have * running with an Eicon.Diehl Diva (passive) ISDN BRI modem and
SIP and IAX2 phones. For the modem I used chan_modem. It is not really
clear to me if chan_capi will work with my hardware. Tried
Do you know which one is the good VoIP termination? Is it Nufone or
VoicePulse Connect? Any other suggestion for business plan.
Thanks.
-Tri.
- Original Message -
From: "Brian Capouch" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, February 24, 2004 10:59 AM
Subject: Re: [
Does anyone know of a good-quality (Vonage, Net2Phone, etc) service that
would partner with us to resell and support residential customers for our
ISP? We need someone who will handle the 911 locator, provide the
equipment, etc. But we will provide support, billing and will [seriously
consider]
Take a look at my GUI app:
http://sourceforge.net/projects/astguiclient/
It'll run on Linux and Windows, it's written in perl and it'll list every
channel(Zap/SIP/Local) that is active on your system updated every second.
You can also do a lot of other things with it too.
We've been using it he
Thanks Brent,
You will never believe, it works.
I copied a config from a friend (ZAPATA.CONF) and now it works fine.
I can call to * from pstn and can call from * to pstn.
I will take a look tomorrow what was wrong in my config and let you know.
Wim
- Original Message -
From: "Brent Fr
On Tue, Feb 24, 2004 at 03:49:36PM -0500, Chris Clifton wrote:
> That's fine for outbound lines, but what if I want to call the guy in the
> next office ? I have to call him and get redirected to his busy vm just to
> know that he's on the phone.
>
> This is a huge issue with the recepetionist wit
The best thing I can recommend is giving Digium a call. That's part of
the reason to buy the cards from Digium, as the support is included free
with the hardware you purchase from them. Well worth the money. They
have an excellent support staff that should be able to see where the
problem is cro
does sombody has an example config for a vegastrem 50 BRI and asterisk would help
me a lott
thanks
Michael
DISCLAIMER: The content of this e-mail message does not constitute a commitment of
DKMA bvba This e-mail and any attachments thereto may contain information which is
confidential a
Chris Clifton wrote:
That's fine for outbound lines, but what if I want to call the guy in the
next office ? I have to call him and get redirected to his busy vm just to
know that he's on the phone.
This is a huge issue with the recepetionist with the 'master console'. How
does he/she know whether
I wonder if there is a possibility to access the
"InfoElement" (provided by chan_capi debugging)
in context - maybe thru a variable - It would be a perfect
way of getting our HiPath to deliver Display-Names to
SIP(Soft)Phones.
Does anybody know about a solution?
Greetings,
Markus.
That's fine for outbound lines, but what if I want to call the guy in the
next office ? I have to call him and get redirected to his busy vm just to
know that he's on the phone.
This is a huge issue with the recepetionist with the 'master console'. How
does he/she know whether a user is busy or no
I have a similar situation,
I have an office with 4 lines that are answered in different names and
wanted to give users the option to see the status of each line etc.
I have some cisco vip30 with the select buttons and appropriate leds, I
was going to set these up on a * box and see how develope
I would put it in a totally different light. IE; depending on who they use
as an IAX/SIP carrier, they may have potentially unlimited outbound and
inbound lines with the limit only imposed by the total number of indications
on the phones in the office and even then, new inbound calls can still
I'd like to see if anyone out there might have some ideas on this.
I have a customer who wants to move to VoIP, but who has an office full
of people who are very conservative about their telephones.
They would like the asterisk system that I am proposing to have
something analogous to what they
Stephen R. Besch wrote:
Sean Rodger wrote:
I'm using a grandstream phone with asterisk.
Everything seems to be working fine, but every once in a while talking to
someone, the call is dropped.
A loud busy signal immediately interrupts the call for the grandstream
user,
while the other person (co
I have firmware 1.0.3.81
I have a Cisco ata186, and that seems to work fine. Infact, I've had that
up for about 3 months without rebooting, and it is still working great.
The dropped call problem only happens on the grandstreams.
Sean Rodger
___
Ast
I see that in queues.conf there is a maxlen variable to control the maximum
size of the queue. So...if you set the queue to a maxlen = 3...my test
caller gets dead air if they are queued to a queue with 3 calls already in
the queue.
I thought I could increment a variable each time a call is qu
Above says all - I have echocancel=yes in capi.conf's [interfaces], but
this week someone called (analog cordless handset on ISDN TA through
ISDN PSTN to Diva card in * box, then to my Grandstream) and I had a
terrible echo.
AFAIK, echosquelch/echotail are for the built-in chan_capi echo
suppressi
Dave,
I have fxsks = 1
(fxsls is for loopstart prot. and it doesn't work for me)
Wim
- Original Message -
From: "David J Carter" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, February 24, 2004 8:33 PM
Subject: RE: [Asterisk-Users] Unable to create channel of type 'Zap'
> I
Brent,
Zaptel configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channel configured.
Registered tone zone 0 (United States / North America)
Wim
- Original Message -
From: "Brent Franks" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesda
Sean Rodger wrote:
I'm using a grandstream phone with asterisk.
Everything seems to be working fine, but every once in a while talking to
someone, the call is dropped.
A loud busy signal immediately interrupts the call for the grandstream user,
while the other person (connected through an X100P) i
I had this after my last CVS update.
A line in Zaptel.conf was set to fxsls=1 instaead of fxsks=1
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Wim Venneman
Sent: 24 February 2004 19:17
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Unable to cr
Wim,
What happens when you do a ztcfg -vv
- Brent
On Tue, 24 Feb 2004, Wim Venneman wrote:
> Thanks Derek,
>
> Changed the channel = 1 to channel => 1, makes no difference.
>
> Wim
>
> - Original Message -
> From: "Derek Samford" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Cc: <
Thanks Derek,
Changed the channel = 1 to channel => 1, makes no difference.
Wim
- Original Message -
From: "Derek Samford" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Cc: <[EMAIL PROTECTED]>
Sent: Tuesday, February 24, 2004 6:38 PM
Subject: RE: [Asterisk-Users] Unable to create channem
You are best off emailing [EMAIL PROTECTED] I just got their rate chart
1 week ago and they are charging 2.9c/min for continental US calls other
than Alaska.
Darren Wiebe
[EMAIL PROTECTED]
Tri Tu wrote:
Hi Brian,
By looking at the www.nufone.net, it doesn't have any much details of the
servic
Tri Tu wrote:
Hi Brian,
By looking at the www.nufone.net, it doesn't have any much details of the
services. What is the current rate that you have for domestic long distance
rate?
2.9c/min domestic.
International varies by country, of course, but is very competitive
according to those of my cust
Hi Brian,
By looking at the www.nufone.net, it doesn't have any much details of the
services. What is the current rate that you have for domestic long distance
rate?
-Tri.
- Original Message -
From: "Brian Capouch" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, February 24,
Sales wrote:
Curious if anyone has any feedback on Nufone voip pbx.
Perfectly happy customer. Most of my customers use NuFone as
well--perhaps a dozen of us all told.
Excellent uptime, reasonable rates. Email-based customer service for
the most part. Others are bothered by that; I am not.
B
Thomas M. Schaefer wrote:
Hi all, I have a strange problem. Whenever I plug in the base cord connected
to the X100, my DSL service goes down. I DO have a Cisco filter (the one
that comes with the product) installed.
Has anyone else seen this problem?
There was a similar entry in the archives, but
hi
>
> 1) Does Asterisk support a consulting transfer? E.g., call comes in, Mary
> answers, Mary presses Transfer and dials Joe, verifies that Joe answers and
> informs him who is calling, and then presses Transfer to complete the
> transfer?
on zap channels yes
on sip channels yes, depending if
Two newbie questions:
1) Does Asterisk support a consulting transfer? E.g., call comes in, Mary
answers, Mary presses Transfer and dials Joe, verifies that Joe answers and
informs him who is calling, and then presses Transfer to complete the
transfer?
2) How does one set up a 3-party conference?
Hi all, I have a strange problem. Whenever I plug in the base cord connected
to the X100, my DSL service goes down. I DO have a Cisco filter (the one
that comes with the product) installed.
Has anyone else seen this problem?
There was a similar entry in the archives, but it was without a filter.
please change WiMax to be WiMAX
Steven Schroedl
President VeriLAN, Inc.
VeriLAN, Inc.
13500 SW Pacific HWY # 241
Tigard, OR 97223
Tel: 503 224-8822
Fax: 503 224-8833
Cell: 503 869-0276
[EMAIL PROTECTED]
www.verilan.com
This e-mail contains proprietary information and may be confidential. If yo
Wim,
Made one more change below in Zapata.conf
It should be channel => 1
-Original Message-
From: Wim Venneman [mailto:[EMAIL PROTECTED]
Sent: Monday, February 23, 2004 4:46 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap'
Thanks for
http://www.chron.com/cs/CDA/ssistory.mpl/business/2416385
Personally I think it's a good thing, though I don't know that
regulation was needed. It appears that 911 services were going to be
provided either way. This just prevents smaller players for skimping on
the service. I'll gladly pay an e
Try this: (You'll have to change the dial portion of course...)
exten => _1800NXX,1,Dial(${TRUNK}/[EMAIL PROTECTED])
exten => _1800NXX,2,Congestion
exten => _1888NXX,1,Dial(${TRUNK}/[EMAIL PROTECTED])
exten => _1888NXX,2,Congestion
exten => _1877NXX,1,Dial(${TRUNK}/[EMAIL PROTE
Oh...and here is the error message from my error_log...fyi.
[Tue Feb 24 11:52:59 20 04] qview.pl: Invalid argument at
/var/www/cgi-bin/qview.pl line 23.
Mark
Has anyone ever had the following error when using qview.pl?
"Invalid argument at /var/www/cgi-bin/qview.pl line 23."
Just wond
Quick and dirty would be to use "sample.call" method to send all of the
intercom extensions to a meetme conference. I know - it's nasty but, it
will do what you are trying to do - kinda.
Asterisk Jones wrote:
I have made a Cisco 7960 auto answer a call on an extension to act like
an Intercom.
Has anyone ever had the following error when using qview.pl?
"Invalid argument at /var/www/cgi-bin/qview.pl line 23."
Just wondering.
Thanks in Advance.
Mark
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Hi All,
I have 2 FXO Gateways devices. Both under sip.conf are set to
"canreinvite=yes". My dailplan is below.
exten => _X.,1,Absolutetimeout(40)
exten => _X.,2,dial(SIP/[EMAIL PROTECTED])
exten => T,1,BackGround(tt-weasels)
exten => T,2,Hangup()
With this setup the absolute timeout is never tr
Hi!
> ; 1st rule
> exten => _1800.,1,Dial(SIP/..)
> exten => _1800.,2,Congestion
>
> ; 2nd rule
> exten => _1.,1,Dial(SIP/..)
> exten => _1.,2,Congestion
>
> The problem is that some 1800 calls are still going to the second rule.
> What is the best way to accomplish that?
I would really
I wasted a lot of time on this issue with an Audiocodes FXO gateway I am
currently exploring possible
workarounds. Of course, I would really like to see a patch on the
asterisk-cvs mail list. :)
Gerard O'Rourke wrote:
Hi,
We are using Asterisk for a h323 / SIP converter.
We are having problem
Put them in different [context] and use include => context in
your dial plan.
-Heison
On Wed, Feb 25, 2004 at 01:43:37AM +0900, Isamar Maia wrote:
>
> I am trying to make some call routing...
>
> I have the following rules in the same context:
>
> ; 1st rule
> exten => _1800.,1,Dial(SIP/.
I am trying to make some call routing...
I have the following rules in the same context:
; 1st rule
exten => _1800.,1,Dial(SIP/..)
exten => _1800.,2,Congestion
; 2nd rule
exten => _1.,1,Dial(SIP/..)
exten => _1.,2,Congestion
The problem is that some 1800 calls are still going to the se
Gee, maybe I'm missing something, but the spec does not say that. The
RFC actually says that
when you send a final response, you are required to store that final
response for 64*T1 seconds
and retransmit the final response each time you receive the
retransmitted request. (T1 = 500ms)
Otherwise
Hi,
Please can somebody explain how to setup the cisco ip phones to make
them work with SIP?
With SCCP they were working fine but now I'm not even able to get them
to talk with asteriks, apparently.
With Sip Debug ON I get some feedback but nothing gives errors or something.
this is my sip.conf
The latest firmware definitely helped my BT-101. I'm not sure if you're
hearing congestion/reorder from the Asterisk box or from the Grandstream
though. I loaned my BT-101 to a friend when I got my 7960 (night and day
difference I'm telling you!) so, I don't have it here compare notes.
If you
I have made a Cisco 7960 auto answer a call on an extension to act like
an Intercom. While interesting, this is isn't very useful in a large
office environment. Is it possible to make multiple phones answer the
intercom call? It seems to me that if you made all the phones ring,
only the firs
What firmware do you have on your GS?
-Heison
On Tue, Feb 24, 2004 at 10:17:25AM -0500, Sean Rodger wrote:
> I'm using a grandstream phone with asterisk.
>
> Everything seems to be working fine, but every once in a while talking to
> someone, the call is dropped.
>
> A loud busy signal immediat
Rana Dutt wrote:
I am now wondering whether my lack of RAM may be the problem. I am running
Asterisk on a Dell Dimension XPS R450 with only 128 Mb of RAM. When I run
the top command, I see my RAM consumption is above 120 Mb, but I always see
about 4 Mb of RAM available. Should I get more RAM? Has a
Cory J Andrews wrote:
> Curious if anyone has any feedback on Nufone voip pbx.
>
>
> Cory J Andrews
Well, they finally called me back after over a week on Monday. Asked me
what I was interested in doing. I left a message telling them I wanted to
get IAX termination services from them. I told t
I wrote to the list a couple weeks back
about my voice mail messages sounding garbled. This happens no matter what
phone I use to record the message, I’ve tried IpDialog, Grandstream and SJ-Phone.
Obviously, no one else is having this problem, since I’ve never seen it
discussed here.
Curious if anyone has any feedback on Nufone
voip pbx.
Cory J Andrews
**
b2 Technologies
454 Sonwil Drive
Buffalo, NY 14225
**
866.44.B2TECH X22
local 716.630.1555 X22
fax 716.630.1548
***
[EMAIL PROTECTED]
web http://www.V
I captured a load spike graphically with ttyload in case anyone want s to
see what it looks like:
After hanging around at 0.50 load, it spiked up to 2.52 in less than 10
seconds. The only active processes before and during the spike were
asterisk-related.
below is a 10 second interval 12 minute s
> Also, have you tried doing a asterisk -vgcr to see what exactly
> is happening at this disconnect?
What is also help full is to send all the logger.conf output to a single
messages file
messages => notice,warning,error,debug,verbose
___
Aster
Have you tried using a soft-phone just to see if it's the phone or
something with your * system?
Also, have you tried doing a asterisk -vgcr to see what exactly
is happening at this disconnect?
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Hello I have an incoming DID configured with
voicepulse. I have iax.conf configured just like voicepulse shows you how when
you sign up.
Every time I take in a call, the IVR stops
responding to DTMF tones after about 1 minute or so. If I use the same
configuration, but in sip.conf
the syste
I'm using a grandstream phone with asterisk.
Everything seems to be working fine, but every once in a while talking to
someone, the call is dropped.
A loud busy signal immediately interrupts the call for the grandstream user,
while the other person (connected through an X100P) is just cut off.
D
You can have it accept "unauthenticated" calls from anywhere, if you
wish, or you can have it require a password. The password is the
"secret" field. The username is the context heading "othermachine-1".
If no "secret" is defined, it will accept incoming calls from anywhere.
Besides iax.conf
I noted that I have to put a load=res_parking.so before chan_capi.so in modules.conf,
since
chan_capi 3.1 uses some parking group stuff. Otherwise startup failed with error on
symbol
ast_get_group
Worth to notice in the README!
/O
___
Asterisk-Users ma
Hi,
You dont have to reboot your machine, you only have to restart
asterisk, "restart when convenient" is a safe way to do this.
Same thing as with the zaptel channel driver.
best regards
Klaus
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon:
Hi,
We are using Asterisk for a h323 / SIP converter.
We are having problems when users enter DTMF quickly.
The reason for the problem is that the Timestamp Header field in the RTP
stream
for the 2 separate DTMF events are set at the same number.
Has anyone else had this problem and does anyone
Senad Jordanovic wrote:
>> Yes, I have a SPA2000 as well, and noticed this on CVS from 2-3
>> months ago. I have pulled the newest CVS a week or so ago, but not
>> tested this scenario since then.
>>
>> I will pull a new CVS tonight and test again. I've been meaning trace
>> and see if I can watc
I have notised that when ever I nake a change in capi.conf (from junghanns)
I have to
reboot the maschine before the thanges is activated. A reload does not do
the thing.
Is this behavior right ??
Regards.
Jan Larsen
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Hello all,
I have an application where I am attempting to use Agents and
CallQueues to distribute inbound calls to remote users on cell phones. The
system works quite well, except for one annoying thing that I cannot
figure out. I have read just about everything that I can find about the
Hi!
> I've inquired about this before, but it seeems to me that most
> business class pbx systems allow the receptionist to see the status of
> all connected lines at a glance from their phone
There just is no hardware solution for this (at least not yet). However
you can use Asterisk mana
I've read about a 'flash' application for Zaptel that could allow to make
Flash(somewhere)
SendDMTF(remaining digits)
Is there anyway to implement that with H.323 ? and SIP ?
That's
Dial(H.323FXO)
SendDMTF(remaining digits)
The main issue is that Dial doesn't retu
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