How do I make a user dial a passcode to make calls through asterisk?
We would like to place a phone at a client's location for our employee but are
afraid it may get abused by the other workers.
This electronic message contains information from BOSH Global
I think it can be worth checking the authenticate function.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Authenticate
2013/5/20 Felix Vazquez felix.vazq...@theboshgroup.com
How do I make a user dial a passcode to make calls through asterisk?
We would like to place a phone at a client’s
Hello all,
I am getting a strange behaviour of IAX protocol in an IAX trunk set up for
one of our clients.
the calling presentation is equal to 0 : *Calling presentation: 0x00*
Wireshark presents the call as if the from (caller) is null.
It does not seem that there is any config in
Hey All,
I want to implement a conference calling scenario.
Conference Call Procedure:User1 dial the User2. When call is connected put the
current call on Hold and dial User3. When the call is connected between User1
and User3 join the User2 in a conference room!How I can implement this
Here is where to get you start with this.
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
-Tri
From: Faheem faheem_...@yahoo.com
To: asterisk-users@lists.digium.com
Sent: Sat, February 27, 2010 12:08:24 PM
Subject: [asterisk-users] Conference
Muhammad
It is not really your scenario but the scenario to setup a conference
call with three numbers could be to generate two call files that
points to a local channel/a context/extension that route the leg into
the conference room and have your own leg routed into the conference
room
Hi,
I am running a Asterisk 1.6.1.6 (soon to be upgraded) PBX for a client and they
are having a issue that they are unable to reach a TFN (Toll Free Number).
When they call a automated announcement is received that the number will not
accept calls from the originating area code.
It has been
Thx!
Worked as a clock!
I did modify it to:
exten =
977,1,ExecIf($[${CALLERID(num)}=733025975]?Set(CALLERID(all)=Magnus
Benngard))
Even better! :)
On Tue, 15 Dec 2009 11:32:44 -0600, Steve Johnson wrote:
How about:
exten = 977,1,ExecIf($[${CALLERID(num)} =
How about:
exten = 977,1,ExecIf($[${CALLERID(num)} =
733025975]?Set(CALLERID(num)=0317998975))
exten = 977,n,ExecIf($[${CALLERID(num)} = 1234]?Set(CALLERID(num)=317998977))
exten = 977,n,ExecIf($[${CALLERID(num)} = 5678]?Set(CALLERID(num)=317998978))
[..]
exten = 977,n,Dial(SIP/0317998977)
On
Hi!
Trying to figure out how to rewrite calling number of an incoming call...
A cell phone (0733025975) dials a X-Lite (977).
X-Lite shows 733025975 at the display, but I want it to be 0317998975.
I thought i could do something like:
exten = 977/733025975,1,Set(CALLERID(number)=0317998975)
Hi,
I have a weird issue that I hope someone can help me with. I have 2 test
computers and I've changed each the roles of each one with the same results.
I have one xlite client running across a VPN and another connecting directly
from the WAN via the external IP. The client connecting
extension. What
could the issue be?
David Wathen
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Wathen
Sent: Wednesday, November 11, 2009 9:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users
Since 2004 asterisk/libpri have been able to receive the Calling Sub Address
in the ISDN setup message, and the dialplan was able to use it if required.
It's support is limited to only NSAP, not BCD or user formatted.
At the time 25/06/04 the questioned was asked, wouldn't it be a good idea to
On Fri, Jun 12, 2009 at 8:43 AM, Christopher Stamper
christopherstam...@gmail.com wrote:
On Thu, Jun 11, 2009 at 2:00 PM, Danny Nicholas da...@debsinc.com wrote:
Nerdvittles.com has a nice example of this, when they are up. They
used it for Phone trees for a school or something like
On Thu, Jun 11, 2009 at 2:00 PM, Danny Nicholas da...@debsinc.com wrote:
Nerdvittles.com has a nice example of this, when they are up. They used
it for Phone trees for a school or something like that. Took less than 30
minutes to put in my dialplan and use
Sounds like exactly what I am
Right now, my organization is using a commercial service (OneCallNow.com),
that gives telephone notifications to all numbers in a predefined list.
Example:
-Admin records a voice message
-Service calls each number in the list, and plays the message back to them
It's a pretty handy service,
] On Behalf Of Christopher
Stamper
Sent: Thursday, June 11, 2009 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Automatic Calling Feature?
Right now, my organization is using a commercial service (OneCallNow.com),
that gives telephone notifications
Not too hard to do,
you can have a script generate a list of call files which automatically
ring the callers in the list and play a message
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Cheers Duncan
Christopher Stamper wrote:
Right now, my organization is using a
Greetings listers.
I'm running asterisk 1.4.21.2 on SUSE 11.0 using
Polycom 501 phones. My outgoing connections are Zapata using a TDM401P.
For the most part I can make and receive calls fine except for these 3
issues:
1. When I call an external conference, the
Turn off callprogres=yes or have it configured properly.
It should fix your problem.
regards
Martin
On Fri, Apr 3, 2009 at 2:42 PM, Danny Nicholas da...@debsinc.com wrote:
Greetings listers.
I’m running asterisk 1.4.21.2 on SUSE 11.0 using
Polycom 501 phones. My
Dear all
Does anyone know where I can find some good quality Russian language voice
file for calling card?
Thanks in advanced?
Sam
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix,
I am writing an AGI script that needs to check on the idle/busy status
of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and
Snoms thrown in for fun).
I ended up grabbing this info from the manager interface, within an AGI
script. A little back-asswards, but it works.
I
On Wed, 12 Mar 2008, Kevin DeGraaf wrote:
I am writing an AGI script that needs to check on the idle/busy status
of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and
Snoms thrown in for fun).
I ended up grabbing this info from the manager interface, within an AGI
script.
Kevin DeGraaf wrote:
I am writing an AGI script that needs to check on the idle/busy status
of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and
Snoms thrown in for fun).
I ended up grabbing this info from the manager interface, within an AGI
script. A little
This worked for me on * 1.6 where 1223 is the sip peer I wanted to get
status from.
use Asterisk::AGI;
my $AGI = new Asterisk::AGI;
my $peer = $AGI-get_variable(SIPPEER(1223,status));
It didn't work (for me) on 1.4.18. An empty string was returned, even
though I gave a it a valid peer
Solved:
use Asterisk::AGI;
my $AGI = new Asterisk::AGI;
my $peerst = $AGI-get_variable(SIPPEER(123|status));
my $peercc = $AGI-get_variable(SIPPEER(123|curcalls));
This works fine in 1.4.18.
Thanks.
--
Kevin DeGraaf
___
-- Bandwidth and
Greetings,
I am writing an AGI script that needs to check on the idle/busy status
of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and
Snoms thrown in for fun).
Is it possible to call Asterisk functions (e.g. SIPPEER) from AGI
scripts? Based on my Googling, I would guess
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Kevin DeGraaf wrote:
$AGI-verbose(Test using Set(): $cc[0] $cc[1] $cc[2]);
$AGI-verbose(Status of 200: . $AGI-channel_status('SIP/200'));
$AGI-verbose(Status of 221: . $AGI-channel_status('SIP/221'));
$AGI-verbose(Status of 231: .
We have the following weird issue.
When we call an unallocated number from asterisk through an E1/PRI
euroisdn, the call disconnect with cause 31 (unspecified), This produce
an Asterisk congestion message.
If the same E1/PRI trunk is now connected to a Nortel BCM400, the call
disconnect with cause
Hi. I am using the 'get_data' function from an AGI, and i find that
sometimes when users call in, it won't play the full gsm soundfile, and when
i try to press a number (or pound, or star), nothing will happen - it just
hangs there...
anyone else experience this?
- Dominic Son
It is not the
I've had slew of problems with my Bell Canada Single Number Reach (SNR)
dropping in the past couple of months. Another outage Monday for
several hours has me wondering if there's a way to
1. Make a call out of my system via a PSTN back to my SNR line, say
every 30 minutes (this I'm sure is
I see three parts to this if I was doing it.
1) set up an extension that, when dialed, requests a huge pin number.
upon successfull pin number entry, it 'touch'es a file on the server to
update its modification time
[internal]
; could be extension to update heartbeat, asks for pin next
I must be not understanding your question very well, because it seems
like an easy answer :)
In the following Dial event, we have Source and Destination. Like Eric
said, Destination can contain multiple devices, so can't be trusted.
But Source should only contain one device Does it have
Thanks for your answer, I've now investigated this further and it was
really easy, it was right in front of my eyes...
Using the java library from asterisk-java.org it was extreamly easy,
start monitoring on a new channel, check that to is ment for me and add
a property listener for
Hi,
Is it possible to get the remote channelname that will be bridged when
the call is answered, only having the channel that is in the Ring(ing)
state? As far as I can see no variable seems to fit when doing the show
channel command.
I want to be able to redirect/manipulate an incoming
Marcus Carlson wrote:
Is it possible to get the remote channelname that will be bridged when
the call is answered, only having the channel that is in the Ring(ing)
state? As far as I can see no variable seems to fit when doing the show
channel command.
I want to be able to
Hey users,
i've got a question about calling line id in libpri - zaptel with
switchtype q.sig. My Q.Sig partner is a Siemens F900 (HiPoint). If I
enable
span debug i see messages from type CONNECT with some kind of bit field:
Protocol Discriminator: Q.931 (8) len=87
Call Ref: len= 2
: [asterisk-users] Hold calling channel and ask called
channelbeforeconnect???
you can find an example on the wiki here:
http://www.voip-info.org/wiki/view/Asterisk+cmd+dial
On 12/1/06, Nigel J. Terry [EMAIL PROTECTED] wrote:
I posted this a week ago and have had no response. Can someone tell me if
I
am
or
impossible?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nigel J. Terry
Sent: Wednesday, November 22, 2006 10:27 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hold calling channel and ask called channel
beforeconnect???
I am
either obvious or
impossible?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nigel J.
Terry
Sent: Wednesday, November 22, 2006 10:27 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hold calling channel and ask called channel
, 2006 10:27 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hold calling channel and ask called channel
beforeconnect???
I am a newbie. Just got my Asterisk working and I love it.
I want to do the following, believe it should be possible, but can't work
out how:
When I get
J. Terry
Sent: Wednesday, November 22, 2006 10:27 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hold calling channel and ask called channel
beforeconnect???
I am a newbie. Just got my Asterisk working and I love it.
I want to do the following, believe it should be possible
I am a newbie. Just got my Asterisk working and I love it.
I want to do the following, believe it should be possible, but can't work
out how:
When I get an incoming call, I want to answer and just send ringing to the
calling channel.
Then I want to call the destination channel, send a message
Right -
I get the error on the console - I just can't tell how many
transcodes are occuring at any given point in time...
On 10/18/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Mr. Jones wrote:
Is there some way I can tell?
On 10/16/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Mr. Jones wrote:
Mr. Jones wrote:
Is there some way I can tell?
On 10/16/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Mr. Jones wrote:
I'm having problems with conference calls (3-way) when I have my codec
forced to g729 in sip.conf.
I'm using Grandstream 2000s.
If enable both g711 and g729 then 3 way
Mr. Jones wrote:
I'm having problems with conference calls (3-way) when I have my codec
forced to g729 in sip.conf.
I'm using Grandstream 2000s.
If enable both g711 and g729 then 3 way calling and transfers work.
I'm not sure why this would matter?
Here's the error:
Oct 13 13:54:45
Is there some way I can tell?
On 10/16/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Mr. Jones wrote:
I'm having problems with conference calls (3-way) when I have my codec
forced to g729 in sip.conf.
I'm using Grandstream 2000s.
If enable both g711 and g729 then 3 way calling and transfers
I'm having problems with conference calls (3-way) when I have my codec
forced to g729 in sip.conf.
I'm using Grandstream 2000s.
If enable both g711 and g729 then 3 way calling and transfers work.
I'm not sure why this would matter?
Here's the error:
Oct 13 13:54:45 NOTICE[31184] chan_sip.c:
On Wed, 2006-10-04 at 15:14 -0700, [EMAIL PROTECTED] wrote:
I am trying to call the DUNDILOOKUP dialplan function from
ael2, like this:
context route {
Set(PATH=${DUNDILOOKUP(${EXTEN},DUNDIRegistr)});
}
The
Thanks for the quick reply Steve. It turned out to be user error. D'oh.
-Original Message-
From: Steve Murphy [mailto:[EMAIL PROTECTED]
Sent: Wed 10/4/2006 10:32 PM
To: asterisk-users@lists.digium.com
Cc:
Subject: [asterisk-users] Re
Hi all,Let me add to my query.I would prefer to have an Asterisk GUI that has billing calling card solution all in one.Got any suggestions.?Thanks
On 12/09/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi Users,Im looking for recommendations on softwares for calling card implementation and post
You can try for Trixbox"[EMAIL PROTECTED]" [EMAIL PROTECTED] wrote: Hi all,Let me add to my query.I would prefer to have an Asterisk GUI that has billing calling card solution all in one.Got any suggestions.?Thanks On 12/09/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Users,Im looking for
I thought maybe my configs would have been a good idea to post:
iax.conf:
[general]
bindport=4569
bindaddr=10.0.0.20
bandwidth=medium
disallow=lpc10
allow=gsm
jitterbuffer=no
forcejitterbuffer=no
register = 776754:snipped@iax2.fwdnet.net
allow=ulaw
tos=lowdelay
autokill=yes
[iaxfwd]
Hi Michael,
I tried what you had said and then tried calling you, and it worked. Then
I called my brother and while I did not get the error, I still got the
busy message i was getting before I borked my config trying too many
ideas ;)
So, any other 6 digit FWD users willing to take a call
Nick Ellson wrote:
Hi Michael,
I tried what you had said and then tried calling you, and it worked.
Then I called my brother and while I did not get the error, I still got
the busy message i was getting before I borked my config trying too
many ideas ;)
So, any other 6 digit FWD users
-- Call accepted by 192.246.69.186 (format ulaw)
-- Format for call is ulaw
-- IAX2/192.246.69.186:4569-3 is busy
-- Hungup 'IAX2/192.246.69.186:4569-3'
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing Congestion(SIP/4003-5d5e, ) in new stack
== Spawn
Hi all,
I have been researching a dialing problem I am having with FWD. I followed
their IAX2 config notes, and I can receive calls from my brother from FWD,
and all the echo tests, call me services work. But I cannot call him.
-- Executing SetCallerID(SIP/4003-9de6, Nick Ellson) in new
Hello Jonathan,I tried in quintum to route my server with any dialed number. but i am not agble to get in quintum FXO line configuration, so i can route the call to my asterisk.do u have any about quintum how i can route calls to server once FXO line will be called?Abdul
Do you Yahoo!? Everyone
Ive only used a Quintum a few
times,sorry.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abdul
Sent: Friday, August 25, 2006 6:49
AM
To:
Asterisk-Users@lists.digium.com
Subject: RE: [asterisk-users]
quintum Calling Card
Hello Jonathan,
I tried
Hi all,Could anyone provide me some usefull link or some idea, how to configure quintum as calling card purpose with Asterisk.Already i created AGI script which working with SIPURA well. But i do not have the idea about quintum how to configure so quintum will dial our asterisk calling card
PROTECTED] On Behalf Of Abdul
Sent: Thursday, August 24, 2006
8:12 AM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users] quintum
Calling Card
Hi all,
Could anyone provide me some usefull link or some idea, how to configure
quintum as calling card purpose with Asterisk.
Already i
Hi,
I'm trying to call an exten from inside extensions.ael, as below, ddi calls
ael and then ael needs to call the extensions.conf (8000 exten) for the call
queue.
Is this possible? Or is there an easier way to combine the exten 8000 to the
ael?
Thanks,
Dean.
ddi.conf
exten =
Hi there!
I'm setting up an E1 with a new Telco and they are asking me to add the
extension number into an "Additional calling party number". Guess it
refeers to a part of the E1 trace they are getting. I've been
playing around with the callerid and in zapata.conf and sip.conf but
have
Hi All,
I installed Asterisk recently and it was working from 2 weeks without a problem
until today. Today it started showing strange error
Feb 28 03:14:08 WARNING[31430]: chan_sip.c:4826 check_auth: Stale nonce received
from 'sip:[EMAIL PROTECTED]'
Whatever number I call it displays this,
I see these from time to time, I think it means that packets got lost,
or received out of sequence. It looks to me like asterisk manages to
deal with this, so unless your calls have also stopped working, I
wouldn't worry. (If we should be worrying, I expect someone will let us
know).
I am using asterisk CVS 10842 and a TDM 400p withanfxs and an fxo
module and when I dial the fxs channel it rings for a second and then
says no answer after 20 seconds. I also have the latest Zaptel drivers.
Here is a log snippet.
Feb 25 00:53:11 VERBOSE[2015] logger.c: -- Executing
On Wed, 2006-02-22 at 21:44 +0200, [EMAIL PROTECTED] wrote:
Hi, i'm using an elesign voip gateway esc1700 to call to one iax
sofphone (idefisk) and sip device (sipura spa-2002), the trouble es when
I make the call using the esc1700 the communication is dropped, this is
the log portion:
Hi, i'm using an elesign voip gateway esc1700 to call to one iax
sofphone (idefisk) and sip device (sipura spa-2002), the trouble es when
I make the call using the esc1700 the communication is dropped, this is
the log portion:
Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Call accepted by
Hi, i'm using an elesign voip gateway esc1700 to call to one iax
sofphone (idefisk) and sip device (sipura spa-2002), the trouble es when
I make the call using the esc1700 the communication is dropped, this is
the log portion:
Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Call accepted by
Sorry for the late response, but the w for wait ONLY works with DTMF.
Not well documented, but asterisk doesn't detect dialtone, therefore it
can start to dial numbers before the CO is ready, and I don't know how
you can wait for a second dialtone if it doesn't even wait for the first
one!.
Hi,
I'm having a problem calling international numbers with debian's
Asterisk 1.0.7 w/ zaptel 1.0.9 in Moscow. Russia doesn't seem to have
touchtone dialing, so pulsedial is enabled on my TDM400P interface.
Local numbers work fine, but when it comes to long distance or
international, I'm lost.
Hello,
There're few POTS supporting touchtone, others - just pulse. In
Russia you need to dial 8, wait for tone and only then continue
dialing 10 (for intl. plan), country code, area code and number.
bkmc Hi,
bkmc I'm having a problem calling international numbers with debian's
bkmc Asterisk
Hi Grigoriy,
Thanks for the reply. I have tried to implement this dial pattern by
dialing from 8w10 to 8ww10, 8p10 (which should be the same as 8ww10)
and even just dialing 8 and sending the rest as DTMF, but it doesn't
seem to work, all I hear in the line is dead air with occasional
In article [EMAIL PROTECTED],
Maxim Litnitsky [EMAIL PROTECTED] wrote:
Hello all.
I am trying to use app_mysql.
It works for selects and functions, but does not want to work with
procedures.
Pls have a look:
Please could you post the relevant sections of your dialplan?
Calling function:
I am having an issue using a Polycom 501 and VoIP for outgoing calls where if I call say my credit card company and try to follow their PBX menu, the key presses never register with their PBX. It's as if every key press I make absolutely nothing is being sent to them. Is there some setting in the
set dtmf mode to inband and use g711
On Jan 11, 2006, at 12:21 PM, Andrew Berman wrote:
I am having an issue using a Polycom 501 and VoIP for outgoing
calls where if I call say my credit card company and try to follow
their PBX menu, the key presses never register with their PBX.
It's
I made the change for the dtmf mode (I was already using g711) and it is still having issues. However, it seems to only be when I call American Express's PBX system. I tried a different company and it works. Very bizzarre.
On 1/11/06, Jerry Jones [EMAIL PROTECTED] wrote:
set dtmf mode to inband
Hi
I terminated a call through SIP to a landphone i have the following
problems.
1.) asterisk gives a fake riming tone, it does not give the real tone from
the phone company.
2.) when I put the call on hold the on hold music is not very clear.
but when I talk the call quality is very clear.
I terminated a call through SIP to a landphone i have the following
problems.
1.) asterisk gives a fake riming tone, it does not give the real tone from
the phone company.
2.) when I put the call on hold the on hold music is not very clear.
but when I talk the call quality is very
If you haven't seen it already, this will be a lot of help to you.
http://www.voip-info.org/tiki-index.php?page=AreskiCC+CallingCard+Application+The+idiots+guideV2
You should now be on step 12. :)
G
Omar McKenzie wrote:
Hi
I have gone thru the steps of installing AreskiCC, I
-Users] Areski Calling Card GUI
If you haven't seen it already, this will be a lot of help to you.
http://www.voip-info.org/tiki-index.php?page=AreskiCC+CallingCard+Applicatio
n+The+idiots+guideV2
You should now be on step 12. :)
G
Omar McKenzie wrote:
Hi
I have gone thru
Hi
I have gone thru the steps of installing
AreskiCC, I would like to know how to get access to the GUI interface of this
application.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Can anyone tell me if there is a Calling Card Platform in which I can use
in conjuction with Asterisk that can give me Authentication via the caller
id of the user. I don't want a PIN based Calling Card system, but the
software to be able to recognize the caller ID information and
authenticate the
On Tue, 2005-10-04 at 08:35 -0500, [EMAIL PROTECTED] wrote:
Can anyone tell me if there is a Calling Card Platform in which I can use
in conjuction with Asterisk that can give me Authentication via the caller
id of the user. I don't want a PIN based Calling Card system, but the
software to be
I just copied the *98 extension to the extension of one of our DID numbers.
So now if I dial 5686 from inside or 1-xxx-xxx-5686 from outside, I get the
same prompts as dialing *98.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---
Yup that's what I was going to suggest you do.. we've been using that
and it works great.
On 9/29/05, Steven [EMAIL PROTECTED] wrote:
I just copied the *98 extension to the extension of one of our DID numbers.
So now if I dial 5686 from inside or 1-xxx-xxx-5686 from outside, I get the
same
I'm running Asterisk (stable branch downloaded 2-Sep-05) on RedHat 9
and I have a TDM22B installed (TDM400 w/ 2 FXO and 2 FXS ports).
Everything seems to work except threeway calling. I can establish
a threeway call, but it uses up BOTH FXO lines. Note that I DO
have threeway calling active with
This is the second time that I've seen questions about this in the past few months. My own impression is that it's not worth the trouble. Get yourself an account with a decent ITSP, perhaps IAX2 based, and get a DID via IP. In this way you can have multiple simultaneous incomming calls ( to the
Hi
I have configured sip accounts and they work some times. when i make a call
to another SIP account it works right
but some times i get the following error
Jul 29 07:17:00 WARNING[802]: chan_sip.c:694 retrans_pkt: Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno
102 (Critical
Thank you Michiel.
I tried to remove m and use r , but still not working, after I change r
to R , it is working. Anybody know why?
Michiel van Baak wrote:
On 11:12, Wed 13 Jul 05, Bill Wong wrote:
Can you show me the example, i am newbie.NOt sure whether the code i
modified is correct
On 14:12, Thu 14 Jul 05, Bill Wong wrote:
Thank you Michiel.
I tried to remove m and use r , but still not working, after I change r
to R , it is working. Anybody know why?
This is in the 'show application dial'
'r' -- indicate ringing to the calling party, pass no audio until answered.
Make sure you have /etc/asterisk/indications.conf If that fixes it,
let me know.
Michiel van Baak wrote:
On 14:12, Thu 14 Jul 05, Bill Wong wrote:
Thank you Michiel.
I tried to remove m and use r , but still not working, after I change r
to R , it is working. Anybody know why?
This is
On 11:12, Wed 13 Jul 05, Bill Wong wrote:
Can you show me the example, i am newbie.NOt sure whether the code i
modified is correct or not..
my code as below..
exten = 671042,1,Dial(${PHONES1},20,Ttmr)
loose the m.
m = provide music while ringing
r = provide ring sound while ringing.
Hi,
When I make a call by using sip phone or softphone, there is no calling
sound, how do I get the calling sound ?
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Add the r parameter to the end of the Dial() statement.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Wong
Sent: Tuesday, July 12, 2005 10:38 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] NO calling tone
Hi,
When I make a call
]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Wong
Sent: Tuesday, July 12, 2005 10:38 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] NO calling tone
Hi,
When I make a call by using sip phone or softphone, there is no calling
sound, how do I get the calling sound
I have a question about contexts calling each other. We have one * box
that is setup for multiple companies. Calls come into the default
context and that hands them out to the context for each company. For
example, 1x goes to context1, 2x goes to context2, etc. Each context
includes
From: Ryan Stark [EMAIL PROTECTED]
Subject: [Asterisk-Users] Calling on all Polycom Experts
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
Hey all, I'll give my reseller a call for support in the morning, but I
usually have
better
Hello There,
I *think* i've setuped the AreskiCC2 Calling Card system right , but
i've yet to make any calls out of it , i added a rate card , trunk
and defined some rates , generated some users , added 10 dollars in
them , okay , now i call any number , it asks me to enter my pin , i
do , it
in one of the two defines configs (where you set the database up)
(sorry cant recall which one and im out of the office) there is a min
call value, its set by default around the 10 unit mark. if the cards
credit is below this it stops you going any further. I can only assume
this was to end the
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