adjust your rxgain up or down
Anton Frolov wrote:
Bruce Reeves wrote:
Nov 21 16:54:09 ERROR[6039] caller id.c: fsk_serie made mylen 0 (-9)
Nov 21 16:54:09 WARNING[6039] chan_zap.c: CallerID feed failed: Success
Nov 21 16:54:09 WARNING[6039] chan_zap.c: CallerID returned with error
on
How does the gain play into the callerid? And would the gain being to low
actually effect all 3 lines not just the first 2?
On 11/27/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
adjust your rxgain up or down
Anton Frolov wrote:
Bruce Reeves wrote:
Nov 21 16:54:09 ERROR[6039] caller
CallerID (in the USA and Canada) comes in as a 1200 or 2400 baud FSK
burst, just like a modem. If the volume is too high the audio gets
distorted, if it is too low it is distorted.
Bruce Reeves wrote:
How does the gain play into the callerid? And would the gain being to low
actually effect
Are there different error messages for too high or too low. The reason I ask
is I tried making some adjustments and running ztmonitor while calling a
line and what I saw was that when the rxgain was pretty high
Nov 27 16:24:07 ERROR[7877]: callerid.c:276 callerid_feed: fsk_serie made
mylen 0
The NOTICEs indicate that Caller*ID should be getting into the system.
I don't know what else to suggest.
Bruce Reeves wrote:
Are there different error messages for too high or too low. The reason I
ask
is I tried making some adjustments and running ztmonitor while calling a
line and what I
Dear all,
a newbie question...
I have two external lines (PSTN SIP) and two internal lines.
When receiving an incoming call, I correctly get the CID, but it's not
propagated to the internal lines. My analog phones shows External call
instead of the CID.
My analog device is a TDM400P (2 FXO + 2
Hi!The problem is that I have commercial Asterisk baste switch that it works wit.My trixbox do not. I guess it has to do with the setting of system for Caller ID.//Mattias
On 15/11/06, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Mittwoch, den 15.11.2006, 01:06 +0100 schrieb Mattias
Hi!The problem is that I have commercial Asterisk baste switch that it works wit.My trixbox do not. I guess it has to do with the setting of system for Caller ID.//Mattias
On 15/11/06, Mattias Andersson [EMAIL PROTECTED] wrote:
Hi!The problem is that I have commercial Asterisk baste switch that it
Hi!
The problem is that I have commercial Asterisk baste switch that it works wit.
My trixbox do not. I guess it has to do with the
setting of system for Caller ID.
//Mattias
I do not now way, but my posting are not coming trow. Or are the?
//Mattias
On 15/11/06, Mattias Andersson
Hi!I am getting inbound caller ID fine bout not out.I am in Sweden and suing Rixtelcom /POrt80 as provider.anyone knowing what is wrong?Also is anyone knowing about Swedish voices to trixbox/Asterisk? I have male now and am looking fro female voices.
RegardsMattias-- Mattias
Hi!I am getting inbound caller ID fine bout not out.I am in Sweden and suing Rixtelcom /POrt80 as provider.anyone knowing what is wrong?Also is anyone knowing about Swedish voices to trixbox/Asterisk? I have male now and am looking fro female voices.
Sorry if I have missed a previous answer on the
Am Mittwoch, den 15.11.2006, 01:06 +0100 schrieb Mattias Andersson:
Hi!
I am getting inbound caller ID fine bout not out.
I am in Sweden and suing Rixtelcom /POrt80 as provider.
anyone knowing what is wrong?
Assuming that is a SIP provider, it is not your job to set the callerid
but the
Hi All
Has there been problems with Caller ID and Asterisk 1.2.10 ???
I have a Phone Number from Teliax, I get this
chan_sip.c:10468 handle_request_invite: Failed to authenticate user
3523029577 sip:[EMAIL PROTECTED];tag=as03efd979
And the call does not go through.
I have a CG-410 FXO
Hello,
We are using asterisk with 6 POTS lines and Caller ID is not always
read from the lines properly. Is there a way to make asterisk
wait for the caller id before proceeding with the dial plan or is it
possible a setting is wrong in a conf file somewhere? Any
guidance would be helpful.
Do you have sendcalleridafter=2 in your [channels] section of
/etc/asterisk/zapata.conf? (I had to change it for mine to work)
Nick
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Tue, 3 Oct 2006, [EMAIL PROTECTED] wrote:
Subject
03/10/2006 16:52 Re: [asterisk-users] Caller ID on
Zap not always working
Hello (o;
Ist there a way to remove the trailing @domain from
the displayed caller id on the Cisco 7970G?
No problem dialing a number from the missed call
directory with the domain attached...just looks
weird (o;
cheers
rick
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I was hoping someone might have the answer to this:
As an update: even though you can modify the Caller ID in
extensions.conf for call handling and use CallerID(number) in you
script, asterisk does not honor the modified number in CDR and
VoiceMail.- I need to fix the number at answer.
Do you use the AnI II digits for anything? If not, call the telco and
tell them to just send ten digits. When I used to have some T1s with
UCN, they sent the ANI II digits in a separate field specifically for
that. I could see them in PRI debug on the console. Now with Global
Crossing, I
Well, currently I do not use them, but I hate to give it up either -
nice to be able to ID the call type and it might be useful some day.
Now it's a pain to deal with. What you get is the same here from GBX,
but with MCI, it the same with ANI II added to ANI
Bart
Steve Totaro wrote:
Do you
About 70% of the time, my Local DID provider sends me ANI II digits
(see http://www.nanpa.com/number_resource_info/ani_ii_assignments.html)
where there will be an extra 2 digits
added to the Caller ID - For example 62714222 where '62' = Cell
Phone for example..
The problem is, I have not
At 03:37 PM 9/2/2006, you wrote:
What can I do to strip these digits from Caller ID before answering
the call so CDR and Voice Mail Caller ID announcement show correct number?
I probably mis-typed something here, but something like this should do:
exten =
You could create a function that uses GotoIf() to detect the extra digits.
The line it points to could strip the extra digits.
What version of asterisk are you using? (the functions are different
pre-1.2.1)
On September 2, 2006 18:37, Bart Fisher wrote:
About 70% of the time, my Local DID
Hi,
Does anyone can tell me how to set the caller id shown in the callee
phone? When I use hard IP phone to make a PSTN call, the number
displayed in PSTN phone correctly using set(callerid(num)). However,
the caller id won't be displayed when I use software IP phone to PSTN.
Does any
You cannot set callerid on POTs lines. You my have more luck if you
place your call via a T1 - but it's still up to your carrier. Some
VoIP providrs also allow you to set callerid on SIP calls, but you
need to check. I fear you'll have a hard time finding a carrier that
will allow you to set
On my TDM400 FXO interface, I get the following error
whenever I call in:
Aug 7 19:45:41 ERROR[3515]: callerid.c:276
callerid_feed: fsk_serie made mylen 0 (-9)
Aug 7 19:45:41 WARNING[3515]: chan_zap.c:6087
ss_thread: CallerID feed failed: Success
Aug 7 19:45:41 WARNING[3515]: chan_zap.c:6131
- Douglas Garstang [EMAIL PROTECTED] wrote:
is not set. Part of the problem may be that 3254103 STILL HAS CONTROL
of the call. I have not pressed transfer a second time yet to release
the call, and Asterisk still think that it is attended at this point.
Small clarification: at this
I have
three phones here with extensions 2944093, 3254103 and
9220371.
2944093 calls 3254103. 3254103 transfers 2944093 to 9220371. We want the
caller id of 2944093 to be presented on the display of
9220371.
However, the caller id of the transferer, 3254103, is appearing. This
doesn't
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Tue, 25 Jul 2006 15:25:15 -0300
Subject: [asterisk-users] Caller ID on
Transfers
I have three phones here with extensions
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 8:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID on Transfers
- Original Message -
From: Douglas Garstang
[mailto
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Tue, 25 Jul 2006 15:37:10 -0300
Subject: RE: [asterisk-users] Caller ID on
Transfers
What type of transfer? blind
On Tuesday 25 July 2006 14:37, Douglas Garstang wrote:
What type of transfer? blind or attended?
Does it matter? Both...
Yes it does matter. On any KSU or PBX I have used, attended transfers show
the name/extension of the transferer (presumably because it is THEM you are
talking to).
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 12:48 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Caller ID on Transfers
On Tuesday 25 July 2006 14:37, Douglas Garstang wrote:
What type of transfer
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 8:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID on Transfers
- Original Message -
From: Douglas Garstang
[mailto
Douglas Garstang wrote:
talking to). Blind transfers show the original caller.
Doesn't seem to be happening that way with Polycom phones and blind/attended
transfers.
Both are showing the original calling party caller id.
___
I can
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Tue, 25 Jul 2006 16:31:13 -0300
Subject: RE: [asterisk-users] Caller ID on
Transfers
I thought the new SIP invite had
that's odd, our Polycom phones show the original caller id on blind transfers but the callerid of the person doing the attended transfer, in our case a receptionist. On 7/25/06,
Doug Lytle [EMAIL PROTECTED] wrote:
Douglas Garstang wrote: talking to).Blind transfers show the original caller.
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID on Transfers
Douglas Garstang wrote:
talking to). Blind transfers show
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 9:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID on Transfers
- Original Message -
From: Douglas Garstang
[mailto
Users Mailing List - Non-Commercial
DiscussionSubject: Re: [asterisk-users] Caller ID on
Transfersthat's odd, our Polycom phones show the original
caller id on blind transfers but the callerid of the person doing the attended
transfer, in our case a receptionist.
On 7/25/06, Doug
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Tue, 25 Jul 2006 17:03:00 -0300
Subject: RE: [asterisk-users] Caller ID on
Transfers
If the new invite looks like a regular
Douglas Garstang wrote:
DOug
Hi Doug. Your doing your transfers with the transfer keys on the Polycom, right? So am I. I think a distinction needs to be made here. I get the impression that most people, and certainly the others are using #1 and #2 to do Asterisk assisted transfers.
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 2:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID on Transfers
Douglas Garstang wrote:
DOug
Hi Doug. Your doing your
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 10:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID on Transfers
- Original Message -
From: Douglas Garstang
[mailto
Douglas Garstang wrote:
Doug,
Thanks, but this isn't the same scenario. Can you try transferring from one
Polycom to another Polycom?
Not until the end of the week. I have an order for more phones.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a
In my experience (although I didn't test this as I type now) using the
Transfer button on the Polycom if you do a blind will show the
original CID, and doing an attended will show the transferees CID.
RDNIS should not come up, but ${BLINDTRANSFER} on a blindtransfer
should. On a non blind
.
Regards,
- Brad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
GarstangSent: Tuesday, July 25, 2006 4:05 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: RE:
[asterisk-users] Caller ID on Transfers
Bruce,
I bet your doing Asterisk assisted
This behavior Bruce mentioned is confirmed with our installation
(krisk.org cfg files)
Bruce Reeves wrote:
that's odd, our Polycom phones show the original caller id on blind
transfers but the callerid of the person doing the attended transfer, in
our case a receptionist.
On 7/25/06, * Doug
The 'o' option to the Dial() command, along with using blind transfers,
fixed this problem for us.
A.
On Jul 25, 2006, at 11:25 AM, Douglas Garstang wrote:
I have three phones here with extensions 2944093, 3254103 and 9220371.
2944093 calls 3254103. 3254103 transfers 2944093 to 9220371. We
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 4:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID on Transfers
In my experience (although I didn't test this as I type now) using
that is a new call initially.
Regards,
- Brad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
GarstangSent: Tuesday, July 25, 2006 4:05 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: RE:
[asterisk-users] Caller ID
1.5.x
On 7/25/06, Douglas Garstang [EMAIL PROTECTED] wrote:
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 4:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID on Transfers
In my experience
-Original Message-
From: Anthony Rodgers [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 4:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID on Transfers
The 'o' option to the Dial() command, along with using blind
AFTER the destination number has started to ring.
Doug.
-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID on Transfers
I can say
, 2006 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID on Transfers
I can say with
absolute certainty that in our installations using the blind transfer of
the Polycom (NOT the Asterisk transfers) will show the original caller ID
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 4:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID on Transfers
what version of asterisk you using? I use the ${BLINDTRANSFER} a lot
in my
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID on Transfers
so you are telling me that all the time you have been bitching you
have
DiscussionSubject: RE:
[asterisk-users] Caller ID on Transfers
I
don't understand how that's possible. When you press the 'transfer' button on
the polycom, enter a number, and press send, the SIP messaging and setup at that
point is exactly the same for both an attended and unattanded call
: [asterisk-users] Caller ID on Transfers
so you are telling me that all the time you have been bitching you
have been doing an attended transfer.
anyhow, once you hit transfer, another soft button shows up which says
blind hit that and dial the number.
Oh Crud. Well, here's what I was doing wrong
Hi,
Has anybody configured caller id on a Sangoma analog FXO card?
Does it support both DTMF and FSK based caller id?
Thanks
Mun
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I am not sure, I will check. If I dont', and get it started, will it just start working? If not, what do I need to do?ThanksJoshua West [EMAIL PROTECTED] wrote: Do you have the Caller ID feature with your telephone service package?sdgesa gaeharth wrote: How can I get the external caller id to
I have a single DID number from sellvoip.net.
This works well and has been mostly reliable (HEH). I am wondering if
caller id info is likely be provided for calls to my DID #?
If so, should they show up in the CLI(they don't)? Or how do I go
about looking at them (the caller ID infos that
I'm running 1.2.9.1, and I can't get caller id dialplan matching to work.
When calling from 9220370 to 1234, the following does not match.
exten = 9220370/1234,1,NoOp(${CALLERIDNUM})
exten = 9220370/1234,2,Answer
exten = 9220370/1234,3,Playback(tt-weasels)
However, when calling from 9220370 to
-Users] Caller ID Matching in extensions.conf
I'm running 1.2.9.1, and I can't get caller id dialplan matching to work.
When calling from 9220370 to 1234, the following does not match.
exten = 9220370/1234,1,NoOp(${CALLERIDNUM})
exten = 9220370/1234,2,Answer
exten = 9220370/1234,3,Playback(tt
Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, June 23, 2006 8:28 PM
Subject: [Asterisk-Users] Caller ID Matching in extensions.conf
I'm running 1.2.9.1, and I can't get caller id dialplan matching to work
D'oh... Might have just answered the wrong question here...
Also, if the channel you are using to get the caller ID from is analog
(FXO
or FXS), I believe you may have to answer the channel, then wait 1 sec to
get the correct caller id info.
Tim
On 23/06/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I'm running 1.2.9.1, and I can't get caller id dialplan matching to work.
When calling from 9220370 to 1234, the following does not match.
exten = 9220370/1234,1,NoOp(${CALLERIDNUM})
exten = 9220370/1234,2,Answer
exten =
Oops. You are correct. My bad.
-Original Message-
From: Kevin Collins [mailto:[EMAIL PROTECTED]
Sent: Friday, June 23, 2006 1:35 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Caller ID Matching in extensions.conf
Callerid
On Friday 23 June 2006 15:28, Douglas Garstang wrote:
I'm running 1.2.9.1, and I can't get caller id dialplan matching to work.
When calling from 9220370 to 1234, the following does not match.
exten = 9220370/1234,1,NoOp(${CALLERIDNUM})
exten = 9220370/1234,2,Answer
exten =
How can I get the external caller id to show on the polycom 501 phones. Currently, when someone calls our office, we only see the word "asterisk" in the caller id.This is our set up:VOIP(polycom)---Asterisk 1.2.4---PSTNThanks
Yahoo! Groups gets better.
Do you have the Caller ID feature with your telephone service package?
sdgesa gaeharth wrote:
How can I get the external caller id to show on the polycom 501
phones. Currently, when someone calls our office, we only see the word
asterisk in the caller id.
This is our set up:
We recently switched my wife's business over to an Asterisk setup
using Cisco IP phones (7940s and 7960s) with chan_sccp. They didn't
use any kind of office-style phone system before, they had one
phone in the office with a built in answering machine that would
display the Caller ID of
Message: 21Date: Tue, 20 Jun 2006 10:12:38 -0500From: Brian Swan
[EMAIL PROTECTED]Subject: [Asterisk-Users] Caller-ID Info with Voice Mail -- Can itdisplay to the phone?To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Content
Hi all, just upgraded to 1.2.8 and now incoming calls are showing that they
are coming from internal extensions in the CallerID field... A call comes in
on our T1, get's picked up by the autoattendant, caller selects extension,
and both the CLI the receiving phone display an internal extension as
Re: my previous post about callid.. Seems to be working now after a complete
restart... Still unsure why it was borking though...
Thx all for your patience.
Dan
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How do you do it on a PRI and what do I ask my provider (Bellsouth) for
if they permit it?
Eric ManxPower Wieling wrote:
Tim Litwiller wrote:
Not, on your question - but you brought up something I would really
like to do and I was told it wasn't possible.
how do you do the transfer to
to do it
correctly. But I saw a post from -dev that someone has a patch for 1.2.5 to
improve this.
hth
-Original Message-
From: Tim Litwiller [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 09, 2006 8:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
It happens by default. I don't know what you would ask BellSouth.
Lenwood S. Sawyer III wrote:
How do you do it on a PRI and what do I ask my provider (Bellsouth) for
if they permit it?
Eric ManxPower Wieling wrote:
Tim Litwiller wrote:
Not, on your question - but you brought up something
I doubt it's possible but I'll ask just in case there's a legal way
to do that.
I have an asterisk server setup at work. When someone call from a
PSTN line and enter an extension it rings for a few seconds on the
SIP phones of that extension and then if there's no answer it
transfer the
Not with an analog POTS line. You need a PRI, or a SIP provider to do that.
hth
-Original Message-
From: Martin Roy [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 09, 2006 3:58 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Caller ID forwarding
I doubt it's possible
-
From: Martin Roy [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 09, 2006 3:58 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Caller ID forwarding
I doubt it's possible but I'll ask just in case there's a legal way
to do that.
I have an asterisk server setup at work. When
Not, on your question - but you brought up something I would really like
to do and I was told it wasn't possible.
how do you do the transfer to cell phone with the hook flash.
Martin Roy wrote:
I doubt it's possible but I'll ask just in case there's a legal way
to do that.
I have an
Tim Litwiller wrote:
Not, on your question - but you brought up something I would really like
to do and I was told it wasn't possible.
how do you do the transfer to cell phone with the hook flash.
Martin Roy wrote:
I doubt it's possible but I'll ask just in case there's a legal way
to do
Hello i am a asterik user from Indiai can't receive the caller id information in my WildCard X100P.in India we are using CLIP for Caller ID Can any body help me on this matter..Thanks in advance
Raju
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What is maximum length of name in caller ID? How much charters can I put and be
sure it will work fine?
--
Tomislav Parcina
tparcina#lama.hr
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For the US PSTN network the limit seems to be 15 characters. For
Asterisk you can safely use 20 characters with most VOIP phones.
MATT---
On 3/27/06, Tomislav Parčina [EMAIL PROTECTED] wrote:
What is maximum length of name in caller ID? How much charters can I put and
be sure it will work
Please be aware, when connecting some versions of Avaya Legen/magix
system to Asterisk using a PRI, and the CIDname is longer than 15
characters, then the Avaya will reset the dchannel and all calls in
progress will be dropped, I learned this the hard way.
There are some other things that will
Hi,
I'm using the Pickup() application for direct call pickup having the following
line in the dialplan:
exten = _*88XX,1,Pickup(${EXTEN:2})
It works OK, though I would like to have to get the original caller ID number
forwarded to the phone where I do the pickup and have it displayed during
Check this out:
http://lists.digium.com/pipermail/asterisk-users/2006-March/143394.html
When that one will work, then yours will.
On 3/21/06, Tamás Bondár [EMAIL PROTECTED] wrote:
Hi,
I'm using the Pickup() application for direct call pickup having the following
line in the dialplan:
exten
Marc,
The links to the patches on the site seem to be broken... can you supply
correct links?
Adam Hatia
-Original Message-
From: Marc McLaughlin (LUSYN) [mailto:[EMAIL PROTECTED]
Sent: 01 February 2006 18:58
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Caller ID
Hello all,
The Caller ID patches have been updated to work with X100P and TDM400P
cards. There is also a patch that should fix distinctive ring on TDM400P
cards when using polarity reversal check for Caller ID. It may be required
for the history buffer method too.
I have a quick Caller*ID question.
I have an inbound call to my PBX which I am attempting to bridge with
a PSTN number (specifically my cell phone, so when someone dials my
extension the cell phone rings).
In my extentions.conf I have:
; Daniel -- 1102
exten = 1102,1,Answer()
exten =
I had this similar problem
I have an extension (Sipura 1001) ext 2000
I call the phone, and i have an AGI lookup the incoming CID in a database,
and reset the CIDname to be that from the database
However, In my Mysql table i had callerid 2000 as the value.
so.. if it was my cell calling it
Could anyone please help me with that:
I have an analog telephone connected to Asterisk using a Sipura 2002 ATA. When
calling the extension, the caller ID presented is always the number of that
extension rather than the number of the calling one.
While I learned that this is the new standard
On Saturday 21 January 2006 20:30, Conrad Beckert wrote:
Could anyone please help me with that:
I have an analog telephone connected to Asterisk using a Sipura 2002 ATA.
When calling the extension, the caller ID presented is always the number of
that extension rather than the number of the
Hello
I am using X100P card and set up asterisk box in
india. can any one tell me the procedure for caller ID detection.
here is zapata.conf file
usecallerid=yes
cidsignalling=dtmf
cidstart = polarity
callerid=asreceived
immediate=yes
The CLI output I get is
Starting simple switch on
Hello all,
All my sip users are identified by their
name.lastname (mine would be dov.bigio).
But I have to associate them to extension numbers
too, so I did the following on my extensions.conf.
The problem is that when a call is logged on CDR
and also the caller ids that appear for end
Hi to all,
I am having problems with the caller id using IAX. The
caller id feature does not function for an incoming
IAX2 call when the incoming caller hides the caller
id.
The caller id is presented as blank on my phone
instead of the number i set it to be. It works fine
otherwise and also
Not sure if this was just an error in your email, but your priorities go
1, 1, 2. Make 1, 2, 3?
Also, What version asterisk? I think it's become
Set(CALLERID(name)=blah) and
Set(CALLERID(num[ber?])=12345) lately. Maybe see if these work
differenly or better ;)
Moj
ahmed kassim wrote:
Client wants to use a *67 feature to block caller id on next call. In the
Wiki I have seen references to this being available but I haven't see any
code to actually make it work. Does anyone have a quick solution for
implementing this type of function?
-Kerry
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