On Thu, 13 Jan 2011 13:06:36 -0600, Danny Nicholas
da...@debsinc.com wrote:
Unless you need a canned app, this would be an easy program to develop on
your own. The easiest way (IMO) to do this would be to put a small
instance of Apache on your Asterisk server and run a CGI program that
interfaces
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Thursday, January 13, 2011 4:14 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CallerID and URL pop up for windows...
On Thu
PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CallerID and URL pop up for windows...
On Thu, 13 Jan 2011 13:06:36 -0600, Danny Nicholas
da...@debsinc.com wrote:
Unless you need a canned app, this would be an easy program to develop on
your own. The easiest way (IMO
- Non-Commercial Discussion'
Subject: [asterisk-users] callerid and user on voicemail
Hello,
There is a problem that i can not figure out how to solve.
I got users with 5 digit usernames for sip.
Some users has a callerid for outside calls.
I have such problems
When a user activates
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oguzhan Kayhan
Sent: Wednesday, December 22, 2010 4:11 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] callerid and user
2010/11/20 Olivier oza_4...@yahoo.fr
Depending on what telco Charlie is connected to would change the CallerId
presented to Charlie from being Alice's or Bob's Cid.
When a call is forwarded, Charlie's telco receives different ANI and CID :
some (seems to) favor ANI and some CID.
An
Hi all,
I've got 4 actors on my stage:
Alice calling from outside
Bob transferring incoming calls to Charlie
Charlie who has a mobile phone
My PBX which is connected to my ISDN line.
I want Charlie to see Alice's Callerid after Bob has transferred the
call as if Charlie is receiving the call
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Friday, November 19, 2010 9:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] callerid not forwarded when transferring
-users@lists.digium.com
Subject: [asterisk-users] callerid not forwarded when transferring call
from
ISDN line to mobile phone via Asterisk
Hi all,
I've got 4 actors on my stage:
Alice calling from outside
Bob transferring incoming calls to Charlie
Charlie who has a mobile phone
My PBX which
Nicholas da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Friday, November 19, 2010 9:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] callerid
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Friday, November 19, 2010 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] callerid
Of
Giorgio
Incantalupo
Sent: Friday, November 19, 2010 9:34 AM
To: asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
Subject: [asterisk-users] callerid not forwarded when
transferring call from
ISDN line
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Friday, November 19, 2010 9:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] callerid not forwarded when transferring call from
ISDN line to mobile phone via Asterisk
Hi all,
I've got
Incantalupo
Sent: Friday, November 19, 2010 9:34 AM
To: asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
Subject: [asterisk-users] callerid not forwarded when
transferring call from
ISDN line to mobile phone via Asterisk
Depending on what telco Charlie is connected to would change the CallerId
presented to Charlie from being Alice's or Bob's Cid.
When a call is forwarded, Charlie's telco receives different ANI and CID :
some (seems to) favor ANI and some CID.
An interesting thing to test is to let Bob issue a
Ira said at 13/11/2010 17:50:
At 05:56 AM 11/13/2010, you wrote:
John Novack said at 13/11/2010 12:58:
Ronny Adsetts wrote:
[...]
The problem I'm trying to solve at the moment is getting caller ID
info passed over to the SIP phones when calls are placed. The
exten = s,1,Verbose(1,Samsung
Hi,
I've now set up Asterisk to interface with our current Samsung iDCS 100 PBX via
an 8SLI analogue extension card in the Samsung and an Openvox A400P04 4-FXO
card in the Asterisk box. It all works in that I can place calls in both
directions from the office Samsung extensions and Asterisk
Ronny Adsetts wrote:
Hi,
I've now set up Asterisk to interface with our current Samsung iDCS 100 PBX
via an 8SLI analogue extension card in the Samsung and an Openvox A400P04
4-FXO card in the Asterisk box. It all works in that I can place calls in
both directions from the office
John Novack said at 13/11/2010 12:58:
Ronny Adsetts wrote:
[...]
The problem I'm trying to solve at the moment is getting caller ID
info passed over to the SIP phones when calls are placed. The
caller ID is coming through as 'asterisk' which I assume is the
default if nothing is present. So
At 05:56 AM 11/13/2010, you wrote:
John Novack said at 13/11/2010 12:58:
Ronny Adsetts wrote:
[...]
The problem I'm trying to solve at the moment is getting caller ID
info passed over to the SIP phones when calls are placed. The
exten = s,1,Verbose(1,Samsung 209 ${CALLERID(all)})
And I
Ronny Adsetts wrote:
John Novack said at 13/11/2010 12:58:
Ronny Adsetts wrote:
[...]
The problem I'm trying to solve at the moment is getting caller ID
info passed over to the SIP phones when calls are placed. The
caller ID is coming through as 'asterisk' which I assume
Hi All,
We have a production system running 1.4.25.1 and yesterday we upgraded it to
1.4.36. Basically we use this system to generate scheduled calls via .call
files.
Sample .call file used:
Channel: local/11...@context-out
WaitTime: 30
CallerId: 3
Extension: 2
Context:
I've got 2 asterisk servers on the same box: ubuntu 10.04 lucid. I have not
been able to send useful callerid info between them (callerid becomes
serverB).
serverA register statement: (serverB has the exact opposite statement)
register = serverA:serverapassw...@ip_of_serverb_nic/serverB
I've got 2 asterisk servers on the same box: ubuntu 10.04 lucid. I have not
een able to send useful callerid info between them (callerid becomes
serverB).
serverA register statement: (serverB has the exact opposite statement)
egister = serverA:serverapassw...@ip_of_serverb_nic/serverB
...@lists.digium.com] On Behalf Of
unsero...@aol.com
Sent: Wednesday, August 04, 2010 8:53 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] callerid between 2 asterisk servers
I've got 2 asterisk servers on the same box: ubuntu 10.04 lucid. I have not
been able to send useful
Hi, i've some trouble with an * installation when the following scenario
happen.
1) Inbound call to SIP/ ;
2) Call is redirected to ring group 6xx
3) SIP extension 1xx answer.
4) caller want to speak with john doe on his mobile
5) assistant put caller on hold
6) assistant start a call
Hi!
7) if john doe want to speak with caller assistant bridge the two
lines using the transfer function of GXP2000 phone (REFER).
After the transfer in the CDR i can't see the callerid of the caller,
only data of the bridged call is reported.
Any idea on what i can do to keep it ?
Try to use local channel, and the pass the callerid of the caller to the
local channel, an the later put this in CDR using h extention.
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10
On Mon, May 17, 2010 at 10:26:18PM -0300, Daniel Bareiro wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I'm testing a telephone connected to FXS port of a Sangoma A200 card.
But I'm observing that callerid is not working. The configuration that
I'm using in chan_dahdi.conf
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi, Tzafrir.
On Thu, May 20, 2010 at 09:58:26 -0300, Tzafrir Cohen wrote:
I'm testing a telephone connected to FXS port of a Sangoma A200 card.
But I'm observing that callerid is not working. The configuration
that I'm using in chan_dahdi.conf is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I'm testing a telephone connected to FXS port of a Sangoma A200 card.
But I'm observing that callerid is not working. The configuration that
I'm using in chan_dahdi.conf is the following one:
-
Hi all... I'm sorry for repeating my message.
I have a problem with caller id on my asterisk server with xorcom astribank.
here is my configuration :
centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2
ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco
Hi Alyed,
Thank you for the response. I tried this solution, I got Unknown
displayed instead of 999. Also, I tried both 200 and 200 as the CID
number for the extension, but the results were the same.
On Sat, Apr 10, 2010 at 2:10 PM, Alyed al...@vivoxie.com wrote:
Don't have a system to test
Don't have a system to test this right now, but read somewhere this was a 2
steps solution:
1) Leave the callerid in your tunk definition blank (in your example the 999
username)
2) Use brakets around the callerid definition of your peers: callerid= 200
(extension 200 in your example)
Let us
Hello everyone,
I'm fairly new to asterisk and this list. Currently I'm working on IAX
trunks to send/receive calls between 2 asterisk boxes with asterisk
1.6.1.1+asterisk gui 2.0. After some work in the gui, two boxes can
send/receive calls to/from the other just fine, the only problem I
have is
-Commercial Discussion
Subject: Re: [asterisk-users] CallerID presented in Asterisk
On 10 Mar 2010, at 05:41, Gopalakrishnaiyer Venugopal-Q16770 wrote:
So is this a bug in Asterisk 1.6? Has anyone verified/reported this
issue?
Read what people send you. Are you using FreePBX? If yes
On Thu, Mar 11, 2010 at 04:18:21PM +0800, Gopalakrishnaiyer Venugopal-Q16770
wrote:
Hi,
I am not using FreePBX.I am using Asterisk 1.6.1.6 and TDM800P cards
for analog lines.
This is an analog line. Either caller ID is sent or it isn't. Even if it
is sent, you can choose to ignore it.
On 11 Mar 2010, at 08:18, Gopalakrishnaiyer Venugopal-Q16770 wrote:
I am not using FreePBX.I am using Asterisk 1.6.1.6 and TDM800P cards
for analog lines.
Right! Slowly getting the info we need now. How about some config?
S
--
*
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir
Cohen
Sent: Thursday, March 11, 2010 3:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CallerID presented in Asterisk
On Thu
On Thu, Mar 11, 2010 at 08:21:09PM +0800, Gopalakrishnaiyer Venugopal-Q16770
wrote:
Hi,
I am restricting the caller ID from the SONUS class 5 server towards
the asterisk.
What does (technically) a restricted caller ID mean? How should I be
able to tell a caller ID string is restricted?
hi folks,
I want to change the callerid= variable generated from php page.
Let me explain :
in /var/log/asterisk/cdr-csv/Master.csv we have the following line :
device 360
we want to change it we don't want the extension name but the displayed name
instead of 360 we want Poste 360 or something
Georghy a écrit :
hi folks,
I want to change the callerid= variable generated from php page.
Let me explain :
in /var/log/asterisk/cdr-csv/Master.csv we have the following line :
device 360
we want to change it we don't want the extension name but the displayed name
instead of 360 we want
On 10 Mar 2010, at 05:41, Gopalakrishnaiyer Venugopal-Q16770 wrote:
So is this a bug in Asterisk 1.6? Has anyone verified/reported this
issue?
Read what people send you. Are you using FreePBX? If yes, then that
ticket is a FreePBX bug report. If you read the words in the report it
will
Gopalakrishnaiyer Venugopal-Q16770 wrote:
Caller Identity restricted. The asterisk is displaying the caller id of
the caller eventhough they are not supposed to be shown.
core show application setcallerpres
hylafax*CLI
-= Info about application 'SetCallerPres' =-
[Synopsis]
Set
On 9 Mar 2010, at 12:21, Gopalakrishnaiyer Venugopal-Q16770 wrote:
My SIP server (SONUS) is making a call to Asterisk DAHDI line with
Caller Identity restricted. The asterisk is displaying the caller id
of
the caller eventhough they are not supposed to be shown.
Kindly throw some light on
Hai All,
My SIP server (SONUS) is making a call to Asterisk DAHDI line with
Caller Identity restricted. The asterisk is displaying the caller id of
the caller eventhough they are not supposed to be shown.
Kindly throw some light on this issue
Regards
Venugopal
--
-Commercial Discussion
Subject: Re: [asterisk-users] CallerID presented in Asterisk
On 9 Mar 2010, at 12:21, Gopalakrishnaiyer Venugopal-Q16770 wrote:
My SIP server (SONUS) is making a call to Asterisk DAHDI line with
Caller Identity restricted. The asterisk is displaying the caller id
of the caller
I am having a problem setting the caller ID that shows when I make an outbound
call over my PRI line. If I make a call from a SIP phone registered with the
Asterisk box the PRI is connected to the correct ID shows on my cell phone. If
I make a call from an IAX trunk connected asterisk box
Using distinctive ring detection with bell202 cid, is there any way to tell
DAHDI to sometimes expect the cid after the 2nd ring, other times after the
1st?
I just added RingMaster service (2nd DID w/ distinctive ring) to a TDM800P
FXO line. No problem setting dringcontext for the 2nd DID.
I have a strange callerid problem. All my SIP phones display the
correct name of the caller but the number is always the number of the
extension that was called. If I do a NoOp on the dialplan I can see
that both name and number are correct.
The call log in my phone records all
sean darcy wrote:
Calling from my home using Asterisk 1.6.2.1 to an office extension
(Asterisk 1.6.1.13) the callerid is not honored:
Home:
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [...@internal:1] Answer(DAHDI/1-1, ) in new stack
-- Executing
On 31 Jan 2010, at 16:24, sean darcy wrote:
-- Executing [...@internal:3] Set(DAHDI/1-1, CALLERID=Test
447) in new stack
Why isn't the office asterisk picking up the callerid from the home
asterisk?
You're making up the syntax?
http://www.voip-info.org/wiki/view/Setting+Callerid
S
--
Steve Howes wrote:
On 31 Jan 2010, at 16:24, sean darcy wrote:
-- Executing [...@internal:3] Set(DAHDI/1-1, CALLERID=Test
447) in new stack
Why isn't the office asterisk picking up the callerid from the home
asterisk?
You're making up the syntax?
On 31 Jan 2010, at 23:17, sean darcy wrote:
Doh. It appears I was making it up.
Thanks.
No problem. If that doesn't work, try a sip debug and see what's in
that.
S
--
_
-- Bandwidth and Colocation Provided by
Calling from my home using Asterisk 1.6.2.1 to an office extension
(Asterisk 1.6.1.13) the callerid is not honored:
Home:
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [...@internal:1] Answer(DAHDI/1-1, ) in new stack
-- Executing [...@internal:2] NoOp(DAHDI/1-1, Context:
Hi,
I got a solution for this problem from Freepbx
forumhttp://www.freepbx.org/forum/freepbx/users/caller-id-not-working#comment-23520.
Is anybody know about this DTMF to FSK converter? Is this solution solve my
problem?
Any way I will try it and get back.
--
Thanks,
Arun S
System
Hi,
But the caller ID function is still not working my system.
Please Help.
Thanks,
Arun S
On Wed, Jan 6, 2010 at 11:13 AM, Kyle Kienapfel doctor.w...@gmail.comwrote:
On Tue, Jan 5, 2010 at 5:24 AM, Arun Sasidhar
arun.sasid...@cabotsolutions.com wrote:
Hi,
I am using asterisknow
On Tue, Jan 05, 2010 at 06:54:18PM +0530, Arun Sasidhar wrote:
Hi,
I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
working fine except the caller ID of incoming call from PSTN line. The phone
display is showing Unknown when there is an incoming call. I think the
Hi,
I dont know the type of caller ID. What you mean by this?. I am from
India. I don't know more about this.
*
Thanks,
Arun S*
On Wed, Jan 6, 2010 at 4:40 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Tue, Jan 05, 2010 at 06:54:18PM +0530, Arun Sasidhar wrote:
Hi,
I am
On Wed, 6 Jan 2010, Arun Sasidhar wrote:
Hi,
I dont know the type of caller ID. What you mean by this?. I am from
India. I don't know more about this.
*
Thanks,
Arun S*
Hi Arun,
Just for fun I read over the bug id you quoted below, and it seems there
are a number of settings you may
] On Behalf Of John Novack
Sent: Wednesday, January 06, 2010 8:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID on Indian PSTN is not working.
Are you even paying for the service?
Here in the US, on PSTN lines from the ILEC's, CallerID is a pay
Are you even paying for the service?
Here in the US, on PSTN lines from the ILEC's, CallerID is a pay
service, with 2 tiers. Number only, and number with name.
Some CLEC's include this without extra charge, as do most/all VOIP
providers.
Do you have a box or phone, independent of the Asterisk
Hi,
Its a free service here and My ordinary phone displaying the Caller ID
without any problem.
I have done some modifications in zapata.conf
Now it looks like this
*[channels]
language=en
hanguponpolarityswitch=yes
answeronpolarityswitch=yes
busydetect=yes
busycount=6
callprogress=yes
At 07:00 AM 1/6/2010, you wrote:
Its a free service here and My ordinary phone displaying the
Caller ID without any problem.
I have done some modifications in zapata.conf
Now it looks like this
Make sure that there is between 1 and 5 seconds after the first ring
before you answer the call.
On Wed, Jan 06, 2010 at 08:30:48PM +0530, Arun Sasidhar wrote:
Hi,
Its a free service here and My ordinary phone displaying the Caller ID
without any problem.
I have done some modifications in zapata.conf
Now it looks like this
*[channels]
language=en
hanguponpolarityswitch=yes
hi,
I made changes in zapata.conf but no result.
I tried different settings. I am getting differnt logs But no result
when i use cidstart=ring
I am getting this in my asterisk log
[Jan 7 09:31:13] VERBOSE[7129] logger.c: -- Starting simple switch on
'DAHDI/1-1'
[Jan 7 09:31:14] ERROR[7129]
On Thu, Jan 07, 2010 at 09:54:09AM +0530, Arun Sasidhar wrote:
hi,
I made changes in zapata.conf but no result.
You use zapata.conf . I suppose you use asterisk 1.4 . Give asterisk
1.6.0 or newer a shot.
--
Tzafrir Cohen
icq#16849755
hi arun can you paste a dialplan here
and version of asterisk
regards
dhaval
On Thu, Jan 7, 2010 at 11:51 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Thu, Jan 07, 2010 at 09:54:09AM +0530, Arun Sasidhar wrote:
hi,
I made changes in zapata.conf but no result.
You use
Hi,
I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
working fine except the caller ID of incoming call from PSTN line. The phone
display is showing Unknown when there is an incoming call. I think the
same problem listed here: https://issues.asterisk.org/view.php?id=6683
Please respond.
Hi,
I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
working fine except the caller ID of incoming call from PSTN line. The phone
display is showing Unknown when there is an incoming call. I think the
same problem listed here:
On Tue, Jan 5, 2010 at 5:24 AM, Arun Sasidhar
arun.sasid...@cabotsolutions.com wrote:
Hi,
I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
working fine except the caller ID of incoming call from PSTN line. The phone
display is showing Unknown when there is an incoming
we're using a GSM-Gateway on asterisk to forward incoming calls to
the cellphones, but, of course, the cellphones always display the
callerid from the gateway. Does anyone know a symbian app that could
(on an incoming call) connect via grps/3G to a database behind the
asterisk and fetch the real
Hi,
we're using a GSM-Gateway on asterisk to forward incoming calls to the
cellphones, but, of course, the cellphones always display the callerid from
the gateway. Does anyone know a symbian app that could (on an incoming call)
connect via grps/3G to a database behind the asterisk and fetch the
On 8 Sep 2009, at 10:22, Jay R. Worthington wrote:
we're using a GSM-Gateway on asterisk to forward incoming calls to
the cellphones, but, of course, the cellphones always display the
callerid from the gateway. Does anyone know a symbian app that could
(on an incoming call) connect via
Hello all,
What is the recommended way to remove spaces in the name of the caller ID?
Justin.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On Thursday 30 April 2009 10:45:48 Justin Piszcz wrote:
What is the recommended way to remove spaces in the name of the caller ID?
You could use the FILTER function, something along the lines of
Set(CALLERID(name)=${FILTER(ABCDEFGHIJKLMNOPQRSTUVWXYZ,${CALLERID(name)})})
In 1.6.0 and later, you
Hi,
I'm having problems when the callerid of a user defined in the
sip.conf contains special characters such as: ñ, á, é, í, ó , etc. The
strange thing is that these characters are displayed correctly in the
dialplan by using the sip show peer command, but if this user makes a
call, these
Does anyone else have any ideas about this short of changing the
voicemail service (can't do that).
Most voicemail/answering service dont' care about callerid or ani,
they instead use the DID that the call comes in on to decide how to
answer the call.
Get a different voicemail/answering
Hello,
We're having some issues with CallerID and I thought someone here might
be able to shed some light as none of our carriers seem to know what I'm
talking about.
The issues is this:
A client of ours uses an after-hours voicemail service as mandated by
their corporate office. We have a
Most voicemail/answering service dont' care about callerid or ani,
they instead use the DID that the call comes in on to decide how to
answer the call.
Get a different voicemail/answering service.
On Tue, Jan 20, 2009 at 9:32 AM, mail-lists mail-li...@peachnet.com wrote:
Hello,
We're having
Not a possibility I'm afraid. Our client is an insurance agent and the
voicemail/answering service is mandated by corporate.
There also are not various DID's to call in on. All voicemail calls go
to an 800 number
Thanks for your advice though.
Most voicemail/answering service dont' care
I am trying to setup one time caller id block on my system(activated
when an incoming call matches *811XX), and I have had little to
no luck. Could you take a look at my context/macro definition and help
me figure out what I am missing?
Here is my context for my dialplan:
John Koenig wrote:
exten=s,1,set(CALLERID(all)= null)
exten=s,n,Dial(${ARG1})
Just a guess.
exten = s,1,Set(CALLERID(all)= null 0)
exten = s,n,SetCallerPres(prohib)
exten = s,n,Dial(${ARG1})
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Doug Lytle wrote:
John Koenig wrote:
exten=s,1,set(CALLERID(all)= null)
exten=s,n,Dial(${ARG1})
Just a guess.
exten = s,1,Set(CALLERID(all)= null 0)
exten = s,n,SetCallerPres(prohib)
exten = s,n,Dial(${ARG1})
Doug
I believe you need to use:
exten = s,1,Set(CALLERID(all)=)
To
I tried all of the suggestions, and still the callerid remains intact.
I guess at this point I am starting to wonder what bit of logic is being
run when I dial *8111XX...
Is there a way I can trace how a call is being processed within
asterisk? Or even see what I am sending to my
John Koenig wrote:
I tried all of the suggestions, and still the callerid remains intact.
I guess at this point I am starting to wonder what bit of logic is being
run when I dial *8111XX...
Is there a way I can trace how a call is being processed within
asterisk? Or even see what
John Koenig wrote:
I tried all of the suggestions, and still the callerid remains intact.
I guess at this point I am starting to wonder what bit of logic is being
run when I dial *8111XX...
Is there a way I can trace how a call is being processed within
asterisk? Or even see
Thanks for the tip about sip set debug peer. I was able to capture
some information about the call in progress, but I am confused as to
what I see. When I pick up my sip phone I dial
*811area_codeprefixnumber, and the first invite I see is going to
1area_codeprefixnumber@my_asterisk_ip.
John Koenig wrote:
Thanks for the tip about sip set debug peer. I was able to capture
some information about the call in progress, but I am confused as to
what I see. When I pick up my sip phone I dial
*811area_codeprefixnumber, and the first invite I see is going to
Hello,
Asterisk 1.4.21.1
Well it seems like my month for questions. I have a situation where the
CallerID num shows as [EMAIL PROTECTED](the ip of the asterisk
box) on calls to any of the internal phones. This prevents the
ability to dial out from the missed call list. I have not been able to
For quite a long time already my CallerID stopped working (maybe even when
i upgraded from Asterisk 1.2 to Asterisk 1.4). I am using a TDM400P card
(in TDM11B config) with one FXO and one FXS port.
Tried googling for some more recent examples of Asterisk config files for
use in The
Hi,
I use destar, and when is setup in trunk config the
Outgoing calls to IAX trunk
Change Caller*Id Name to: ${CALLERIDNUM}
Its does not adding the calleridnum.
What can i do for adding for the base telnummer the calleridnum
value (this is a 3 character long number)
When i add manualy
Asterisk Users,
I am running a Debian Etch system with Asterisk 1.4.11 with a TDM03B card.
Once in awhile, I get this error on the Asterisk, which causes my channels to
be busy/congested, leaving me with just one channel to recieve and make calls:
NOTICE[31454]: chan_zap.c:6367 ss_thread:
Hi There,
We have a Asterisk 1.4 box with a X100P card connected to a analog
line with Caller ID serrvices enabled on it. When an incoming call
appears we get the following in the log:
-- Starting simple switch on 'Zap/1-1'
-- Detecting post-CID distinctive ring
[Apr 15 10:38:07]
hi, all
i am using zma800p card( from zapmicro).
i create small ivrs.
when i call on fxs channel calls lended and ivrs start on that channel.
but when i use callerid app. from asterisk , doesn't displayed any
callerid on asterisk.
any suggestion.
thanks in advance.
Bhrugu mehta
Hi,
I just encountered a simple but strange problem. I am using 2 sip phones to
call each other. Whenever i make a call, using softphone or ata, ali only
shows the CallerID(name) and not the number. I have no idea why it does not
show the number. I have tried various things but none have worked.
Asterisk Users,
I am running Asterisk-1.4.11 on a Debian
Etch system. On an occasion, when customer calls into my Asterisk Box, I get
this error messagefrom Asterisk
CallerID returned with error on channel Zap/3-1 , causing all my zap
channels to be busy. So, I cannot make any calls in,
John Meksavan wrote:
Asterisk Users,
I am running Asterisk-1.4.11 on a Debian Etch system. On an
occasion, when customer calls into my Asterisk Box, I get this error
messagefrom Asterisk CallerID returned with error on channel Zap/3-1 ,
causing all my zap channels to be busy. So, I
Heres a weird one...
Call comes in on mISDN channel. Little bit of dialcode (in a macro) looks
up the number in the astdb and puts an name to it. No real magic there,
and it works well.
Same macro also has parameter passed in to put a prefix on the name - this
is set in the DDI handling and
Gordon Henderson wrote:
Then no amount of Set(CALLERID(name)=somethin) will work. Even if I
explicitly do a Set before the dial, it seems to get ignored.
Trying doing that while using:
SetCallerPres(allowed)
Within your dial plan
Doug
--
Ben Franklin quote:
Those who would give up
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