Re: [asterisk-users] CallerID

[Rich Adamson]

I have 2 POTs line coming into Asterisk. We have callerid feature from 
Verizon on one of the lines.


I am not able to track any CallerID coming in, in the log. I am pretty 
green with asterisk, and it's not clear if I have to activate for 
CallerID in the dialplan. The voicemail keeps saying " call from an 
unknown caller " etc.
Eventually, i would like to pass on the callerID and name to a pager, if 
the call is not picked up, at the extension, after hanging up the call.


Assuming you are trying to use an X100P or Digium TDM analog card, 
you'll probably want to start with zapata.conf entries something like this:

context=inbound-home
usecallerid=yes
signalling=fxs_ks
faxdetect=no
callerid=asreceived
echocancel=yes
usecallerid=yes
hidecallerid=no
echocancelwhenbridged=yes
rxgain=7
txgain=5
channel => 1

The rxgain and txgain values will need to be tweaked to "your" pstn 
lines, and the values will be highly dependent upon exactly how far you 
are from your Verizon central office. Gains that are to high or to low 
can impact how well callerid info is received.


In your extensions.conf, you might want something like this:
[inbound-home]
exten => s,1,NoOp,${CALLERID(all)}
exten => s,2,Dial(${PHONE3}&${PHONE4})

If you start asterisk from the command like (eg, asterisk -rvv), 
you will see the incoming callerid displayed on the command line because 
of the "NoOP" statement shown above.



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Re: [asterisk-users] CallerID

[Wilson Pickett]

On 7/9/06, Ryder Brook <[EMAIL PROTECTED]> wrote:

I have 2 POTs line coming into Asterisk. We have callerid feature from
Verizon on one of the lines.


What interface are the lines connected to?


I am not able to track any CallerID coming in, in the log. I am pretty green
with asterisk, and it's not clear if I have to activate for CallerID in the
dialplan. The voicemail keeps saying " call from an unknown caller " etc.


For zaptel interface, the configuration is in zapata.conf and
/etc/zaptel.conf. Things like usecallerid=yes are done there. As to
your "greenness", everyone starts there. One suggestion is to read
available books that systematically take you through which config
files do what, such as http://asteriskdocs.org

Another is to look at the info  in /usr/src/asterisk/doc/ and the
sample config files which theoretically each contain all available
statements for that file.

Eveyone complains that the existing docs are not accurate. Asterisk
being a moving target, you can only hope to get a basic understanding
reading most of these, but it's well worth the trouble.
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Re: [asterisk-users] CallerID

[Tom Vile]

and what TDM card are you using and what does your zapata.conf file look like.

On 7/8/06, Ryder Brook <[EMAIL PROTECTED]> wrote:

Hope someone call help me .

I have 2 POTs line coming into Asterisk. We have callerid feature from
Verizon on one of the lines.

I am not able to track any CallerID coming in, in the log. I am pretty green
with asterisk, and it's not clear if I have to activate for CallerID in the
dialplan. The voicemail keeps saying " call from an unknown caller " etc.
Eventually, i would like to pass on the callerID and name to a pager, if the
call is not picked up, at the extension, after hanging up the call.

Thanks,
braman


Ryder Brook Pediatrics
P.O.Box 608
Morrisville, VT 05661


 
Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+
countries) for 2¢/min or less.



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Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [asterisk-users] CallerID in UK on TalkTalk - different to BT?

[Steve Kennedy]

On Sat, Jul 08, 2006 at 01:55:00PM +0100, Thomas Kenyon wrote:

[snip]
> I thought that the line would now go through talktalk (It is an LLU
> service after all).
> FWIW, the same thing happened to me with a line that moved to bulldog.

In the UK BT still own 85% of all copper into premises. Ofcom forced
them to relinquish various controls so others could offer service using
their lines (but BT still remain owners of the copper).

Initially there was CPS (carrier pre select) whereby the line was still
rented from BT, but another operator would get all the call traffic,
this is done by programming the line in the exchange.

There is now also WLR (wholesale line rental) where the billing of the
line is taken over by the other operator, but call traffic really stills
go through the exchange to another operator.

With LLU (local loop unbundling) there are two options, shared metallic
path, whereby BT bill the line and do voice, and the LLU operator takes
the broadband service.

Non-shared, the line is completely jumpered to the LLU operator and they
take voice and data (Bulldog used this).

TalkTalk WILL do full non-shared service, however as they haven't built
out enough most of the customers they're signing up are actually using
BT Wholesale for broadband provision and using WLR for voice. They will
eventually migrate those users to their own kit when they unbundle the
exchange.


Steve

-- 
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Re: [asterisk-users] CallerID in UK on TalkTalk - different to BT?

[Angus Comber]

Its strange, if I reboot my Asterisk you get no callerid.  But then if you
do a reload of the config then callerid comes back.  any ideas why this
could happen?

Angus

- Original Message -
From: "Steve Kennedy" <[EMAIL PROTECTED]>
To: 
Sent: Saturday, July 08, 2006 12:37 PM
Subject: Re: [asterisk-users] CallerID in UK on TalkTalk - different to BT?


> On Sat, Jul 08, 2006 at 10:59:49AM +0100, Angus Comber wrote:
>
> > I had an Asterisk installation working fine for CallerID on BT analog
lines
> > using a Digium analog 4 port card.  However, user switched to TalkTalk
> > without telling me and CallerID no longer works.  However, if you
connect a
> > UK CallerID capable phone into one of these analog lines directly you do
see
> > the CallerID.
> > Does anyone know how to tweak the settings for Talk Talk.  Talk Talk
have
> > basically taken over the line rental - and they supply everything
including
> > the CLIP (CallerID) service now.
> > Just to be clear CallerID was working fine before when line rental
supplied
> > by BT.
>
> Even though the line has been taken over by TalkTalk, it's still a BT
> line off a BT Exchange so the initial leg of the call (or final
> depending on your point of view) is still BT.
>
> Steve
>
> --
> NetTek Ltd  UK mob +44-(0)7775 755503
> UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
> Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
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Re: [asterisk-users] CallerID in UK on TalkTalk - different to BT?

[Thomas Kenyon]

Steve Kennedy wrote:
> On Sat, Jul 08, 2006 at 10:59:49AM +0100, Angus Comber wrote:
>
>   
>> I had an Asterisk installation working fine for CallerID on BT analog lines
>> using a Digium analog 4 port card.  However, user switched to TalkTalk
>> without telling me and CallerID no longer works.  However, if you connect a
>> UK CallerID capable phone into one of these analog lines directly you do see
>> the CallerID.
>> Does anyone know how to tweak the settings for Talk Talk.  Talk Talk have
>> basically taken over the line rental - and they supply everything including
>> the CLIP (CallerID) service now.
>> Just to be clear CallerID was working fine before when line rental supplied
>> by BT.
>> 
>
> Even though the line has been taken over by TalkTalk, it's still a BT
> line off a BT Exchange so the initial leg of the call (or final
> depending on your point of view) is still BT.
>
> Steve
>   
I thought that the line would now go through talktalk (It is an LLU
service after all).

FWIW, the same thing happened to me with a line that moved to bulldog.

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Re: [asterisk-users] CallerID in UK on TalkTalk - different to BT?

[Steve Kennedy]

On Sat, Jul 08, 2006 at 10:59:49AM +0100, Angus Comber wrote:

> I had an Asterisk installation working fine for CallerID on BT analog lines
> using a Digium analog 4 port card.  However, user switched to TalkTalk
> without telling me and CallerID no longer works.  However, if you connect a
> UK CallerID capable phone into one of these analog lines directly you do see
> the CallerID.
> Does anyone know how to tweak the settings for Talk Talk.  Talk Talk have
> basically taken over the line rental - and they supply everything including
> the CLIP (CallerID) service now.
> Just to be clear CallerID was working fine before when line rental supplied
> by BT.

Even though the line has been taken over by TalkTalk, it's still a BT
line off a BT Exchange so the initial leg of the call (or final
depending on your point of view) is still BT.

Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
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Re: [Asterisk-Users] Callerid and trunk

[Julian Lyndon-Smith]

Oh man, I feel such an idiot. I knew that. Just chose to forget it. ;(

Many thanks for reminding me that I am a fool !

Thanks.

Julian.

Hadley Rich wrote:

On Wednesday 31 May 2006 09:08, Julian Lyndon-Smith wrote:

Ok, I must be really stupid here -

I'm playing with ael and svn trunk.

given the following in ael:

context isdn10 {

444601 => {
 Answer();
 NoOp(${CALLERIDNUM});
 Hangup();
 };
};

isdn10 is the incoming isdn context.

why do I get this on the console:

   -- Accepting call from '01702xx' to 'yy' on channel 0/1, span 1
 -- Executing [isdn10:1] Answer("Zap/1-1", "") in new stack
 -- Executing [isdn10:2] NoOp("Zap/1-1", "") in new stack
 -- Executing [isdn10:3] Hangup("Zap/1-1", "") in new stack

callerid must be working: get the from (01702xx) and to yy

but why is ${CALLERIDNUM} blank ?


Because it's deprecated and I assume dropped completely for 1.4. Use 
${CALLERID(num)}


hads



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Re: [Asterisk-Users] Callerid and trunk

[Hadley Rich]

On Wednesday 31 May 2006 09:08, Julian Lyndon-Smith wrote:
> Ok, I must be really stupid here -
>
> I'm playing with ael and svn trunk.
>
> given the following in ael:
>
> context isdn10 {
>
> 444601 => {
>  Answer();
>  NoOp(${CALLERIDNUM});
>  Hangup();
>  };
> };
>
> isdn10 is the incoming isdn context.
>
> why do I get this on the console:
>
>-- Accepting call from '01702xx' to 'yy' on channel 0/1, span 1
>  -- Executing [isdn10:1] Answer("Zap/1-1", "") in new stack
>  -- Executing [isdn10:2] NoOp("Zap/1-1", "") in new stack
>  -- Executing [isdn10:3] Hangup("Zap/1-1", "") in new stack
>
> callerid must be working: get the from (01702xx) and to yy
>
> but why is ${CALLERIDNUM} blank ?

Because it's deprecated and I assume dropped completely for 1.4. Use 
${CALLERID(num)}

hads

-- 
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don't call it destiny; call it injustice, treachery, or simple bad luck.
-- Joseph Heller, "God Knows"
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Re: [Asterisk-Users] CallerID outbound

[C F]

If you are using POTS then the answer is no.
If you are using VoIP then your provider has to support it.
If you are using a PRI then your provider has to support it.
I believe for anything else (RBS, etc.) the answer is no.

On 5/30/06, George A. Roberts IV <[EMAIL PROTECTED]> wrote:



We're using [EMAIL PROTECTED] 2.8 (same thing occurs with earlier versions).  We
have it set up so that if we don't answer our internal SIP phones it does
"follow me" to our cell phones.  When Asterisk forwards the calls to our
cell phones, the Caller ID shows our outbound number, not the caller's
number.

We have 3 queues set up (1,2, and 3) and our extensions (801 and 802) are
listed as permanently logged in to those queues.

I'm just wondering if there's any way to either 1) have the original
caller's CallerID show up when calls get forwarded to our cell phones; or
ideally 2) be able to override the caller ID when calls coming into the
queues are forwarded our cell phones so we can tell which queue it is.

Thoughts?

George
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Re: [Asterisk-Users] CallerID

[Greg Oliver]

Yeah - I have tried everything - even turning it off on the other PBX
for the entire system - then XO kindly just put in the last 4 and passes
it on - which would normally be OK, but the other PBX I am calling
accepts that as valid and therefore I still get data..

I am going to have to get XO to turn it off momentarily or ask a bill
collector to call the number for me :)

Thanks,

Greg

On Tue, 2006-05-23 at 12:55 -0400, C F wrote:
> It appears that the PBX sitting between Asterisk and your provider is
> not passing on the calling pres flags.
> 
> On 5/23/06, Lee Archer <[EMAIL PROTECTED]> wrote:
> > I have a problem with BT in the UK.  Using setcallerpres I can change
> > the number shown on the recipents phones to Private or unknown but no
> > matter what I set my asterisk cid and callerpres to it still displays
> > the base number of my PRI ddi range.
> >
> > Regards
> >
> > Lee
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of C F
> > Sent: 23 May 2006 15:05
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] CallerID
> >
> > You should set the presentation flags to private.
> > http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres
> >
> > On 5/23/06, Greg Oliver <[EMAIL PROTECTED]> wrote:
> > > I am trying to set CIDNum to nothing, but my outgoing PRI controlled
> > > by another PBX seems to fill in something when asterisk does not..  If
> >
> > > I set a number either in the sip channel for the phone, or from
> > > extensions.con, it is realized..  If I try to leave them blank, or
> > > even Not Defined, the main number of the pri gets sent out..
> > >
> > > I am trying to debug a glitvh in or software and I need to be able to
> > > make a test call with unknown (blank callerid)..  I can successfully
> > > set it to private, but that is not the same..
> > >
> > > Any ideas?
> > >
> > > TIA
> > >
> > > -Greg
> > >
> > > ___
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Re: [Asterisk-Users] CallerID

[C F]

Sorry I thought you were the OP. I guess in the UK (or maybe on E1s)
the flags are different. I can't help you with (voice) E1s as I have
never dealt with one.

On 5/24/06, Lee Archer <[EMAIL PROTECTED]> wrote:

I don't have a PBX sitting between Asterisk and the telco.  Asterisk is
the PBX.  I'm using a TE110P card.

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: 23 May 2006 17:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID

It appears that the PBX sitting between Asterisk and your provider is
not passing on the calling pres flags.

On 5/23/06, Lee Archer <[EMAIL PROTECTED]> wrote:
> I have a problem with BT in the UK.  Using setcallerpres I can change
> the number shown on the recipents phones to Private or unknown but no
> matter what I set my asterisk cid and callerpres to it still displays
> the base number of my PRI ddi range.
>
> Regards
>
> Lee
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of C F
> Sent: 23 May 2006 15:05
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] CallerID
>
> You should set the presentation flags to private.
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres
>
> On 5/23/06, Greg Oliver <[EMAIL PROTECTED]> wrote:
> > I am trying to set CIDNum to nothing, but my outgoing PRI controlled

> > by another PBX seems to fill in something when asterisk does not..
> > If
>
> > I set a number either in the sip channel for the phone, or from
> > extensions.con, it is realized..  If I try to leave them blank, or
> > even Not Defined, the main number of the pri gets sent out..
> >
> > I am trying to debug a glitvh in or software and I need to be able
> > to make a test call with unknown (blank callerid)..  I can
> > successfully set it to private, but that is not the same..
> >
> > Any ideas?
> >
> > TIA
> >
> > -Greg
> >
> > ___
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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RE: [Asterisk-Users] CallerID

[Lee Archer]

I don't have a PBX sitting between Asterisk and the telco.  Asterisk is
the PBX.  I'm using a TE110P card.

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: 23 May 2006 17:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID

It appears that the PBX sitting between Asterisk and your provider is
not passing on the calling pres flags.

On 5/23/06, Lee Archer <[EMAIL PROTECTED]> wrote:
> I have a problem with BT in the UK.  Using setcallerpres I can change 
> the number shown on the recipents phones to Private or unknown but no 
> matter what I set my asterisk cid and callerpres to it still displays 
> the base number of my PRI ddi range.
>
> Regards
>
> Lee
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of C F
> Sent: 23 May 2006 15:05
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] CallerID
>
> You should set the presentation flags to private.
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres
>
> On 5/23/06, Greg Oliver <[EMAIL PROTECTED]> wrote:
> > I am trying to set CIDNum to nothing, but my outgoing PRI controlled

> > by another PBX seems to fill in something when asterisk does not..  
> > If
>
> > I set a number either in the sip channel for the phone, or from 
> > extensions.con, it is realized..  If I try to leave them blank, or 
> > even Not Defined, the main number of the pri gets sent out..
> >
> > I am trying to debug a glitvh in or software and I need to be able 
> > to make a test call with unknown (blank callerid)..  I can 
> > successfully set it to private, but that is not the same..
> >
> > Any ideas?
> >
> > TIA
> >
> > -Greg
> >
> > ___
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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Re: [Asterisk-Users] CallerID

[C F]

It appears that the PBX sitting between Asterisk and your provider is
not passing on the calling pres flags.

On 5/23/06, Lee Archer <[EMAIL PROTECTED]> wrote:

I have a problem with BT in the UK.  Using setcallerpres I can change
the number shown on the recipents phones to Private or unknown but no
matter what I set my asterisk cid and callerpres to it still displays
the base number of my PRI ddi range.

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: 23 May 2006 15:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID

You should set the presentation flags to private.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres

On 5/23/06, Greg Oliver <[EMAIL PROTECTED]> wrote:
> I am trying to set CIDNum to nothing, but my outgoing PRI controlled
> by another PBX seems to fill in something when asterisk does not..  If

> I set a number either in the sip channel for the phone, or from
> extensions.con, it is realized..  If I try to leave them blank, or
> even Not Defined, the main number of the pri gets sent out..
>
> I am trying to debug a glitvh in or software and I need to be able to
> make a test call with unknown (blank callerid)..  I can successfully
> set it to private, but that is not the same..
>
> Any ideas?
>
> TIA
>
> -Greg
>
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RE: [Asterisk-Users] CallerID

[Lee Archer]

I have a problem with BT in the UK.  Using setcallerpres I can change
the number shown on the recipents phones to Private or unknown but no
matter what I set my asterisk cid and callerpres to it still displays
the base number of my PRI ddi range. 

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: 23 May 2006 15:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID

You should set the presentation flags to private.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres

On 5/23/06, Greg Oliver <[EMAIL PROTECTED]> wrote:
> I am trying to set CIDNum to nothing, but my outgoing PRI controlled 
> by another PBX seems to fill in something when asterisk does not..  If

> I set a number either in the sip channel for the phone, or from 
> extensions.con, it is realized..  If I try to leave them blank, or 
> even Not Defined, the main number of the pri gets sent out..
>
> I am trying to debug a glitvh in or software and I need to be able to 
> make a test call with unknown (blank callerid)..  I can successfully 
> set it to private, but that is not the same..
>
> Any ideas?
>
> TIA
>
> -Greg
>
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> To UNSUBSCRIBE or update options visit:
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Re: [Asterisk-Users] CallerID

[C F]

You should set the presentation flags to private.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres

On 5/23/06, Greg Oliver <[EMAIL PROTECTED]> wrote:

I am trying to set CIDNum to nothing, but my outgoing PRI controlled by
another PBX seems to fill in something when asterisk does not..  If I
set a number either in the sip channel for the phone, or from
extensions.con, it is realized..  If I try to leave them blank, or even
Not Defined, the main number of the pri gets sent out..

I am trying to debug a glitvh in or software and I need to be able to
make a test call with unknown (blank callerid)..  I can successfully set
it to private, but that is not the same..

Any ideas?

TIA

-Greg

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Re: [Asterisk-Users] CallerID

[Greg Oliver]

On Tue, 2006-05-23 at 06:27 -0400, Steve Totaro wrote:
> Greg Oliver wrote:
> > I am trying to set CIDNum to nothing, but my outgoing PRI controlled by
> > another PBX seems to fill in something when asterisk does not..  If I
> > set a number either in the sip channel for the phone, or from
> > extensions.con, it is realized..  If I try to leave them blank, or even
> > Not Defined, the main number of the pri gets sent out..
> >
> > I am trying to debug a glitvh in or software and I need to be able to
> > make a test call with unknown (blank callerid)..  I can successfully set
> > it to private, but that is not the same..
> >
> > Any ideas?
> >
> > TIA 
> >
> > -Greg
> >
> >   
> 
> On one of my T1 circuits, ten digits always appear on the other side.  I 
> can set all ten digits to zero.  If I set less than ten digits then the 
> last digits of the default ten digit string (which is our billing phone 
> number) are overwritten with what is set.  On our T3 (different 
> provider), we can set any length of digits but I have never tried to 
> send blank or null values. 
> 
> Does your other PBX send blank callerID?  Is the PRI from the same 
> provider?  When you have CIDNum= do you see errors in the log that the 
> value must not be null?

Unfortunately not - if I fill in anywhere up to the 40 digit max in *,
then the other PBX allows it, but anything that is not valid, it rejects
and puts the main hunt number in..  

I think I am kind of screwed thanks to the 800lb gorilla Cisco
Gotta love 'em.

-Greg

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RE: [Asterisk-Users] CallerID

[Greg Oliver]

On Tue, 2006-05-23 at 09:32 +0100, Mark Ackroyd wrote:
> Here in the UK on pri, setting the callerid to 0, withholds it.
> 
> > I am trying to set CIDNum to nothing, but my outgoing PRI controlled by
> > another PBX seems to fill in something when asterisk does not..  If I
> > set a number either in the sip channel for the phone, or from
> > extensions.con, it is realized..  If I try to leave them blank, or even
> > Not Defined, the main number of the pri gets sent out..
> > 
> > I am trying to debug a glitvh in or software and I need to be able to
> > make a test call with unknown (blank callerid)..  I can successfully set
> > it to private, but that is not the same..

Tried that already - the PBX the PRI is connected to fills it in when it
is invalid..

-Greg

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Re: [Asterisk-Users] CallerID

[Steve Totaro]

Greg Oliver wrote:

I am trying to set CIDNum to nothing, but my outgoing PRI controlled by
another PBX seems to fill in something when asterisk does not..  If I
set a number either in the sip channel for the phone, or from
extensions.con, it is realized..  If I try to leave them blank, or even
Not Defined, the main number of the pri gets sent out..

I am trying to debug a glitvh in or software and I need to be able to
make a test call with unknown (blank callerid)..  I can successfully set
it to private, but that is not the same..

Any ideas?

TIA 


-Greg

  


On one of my T1 circuits, ten digits always appear on the other side.  I 
can set all ten digits to zero.  If I set less than ten digits then the 
last digits of the default ten digit string (which is our billing phone 
number) are overwritten with what is set.  On our T3 (different 
provider), we can set any length of digits but I have never tried to 
send blank or null values. 

Does your other PBX send blank callerID?  Is the PRI from the same 
provider?  When you have CIDNum= do you see errors in the log that the 
value must not be null?



Thanks,
Steve
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RE: [Asterisk-Users] CallerID

[Mark Ackroyd]

Here in the UK on pri, setting the callerid to 0, withholds it.

> I am trying to set CIDNum to nothing, but my outgoing PRI controlled by
> another PBX seems to fill in something when asterisk does not..  If I
> set a number either in the sip channel for the phone, or from
> extensions.con, it is realized..  If I try to leave them blank, or even
> Not Defined, the main number of the pri gets sent out..
> 
> I am trying to debug a glitvh in or software and I need to be able to
> make a test call with unknown (blank callerid)..  I can successfully set
> it to private, but that is not the same..

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Re: [Asterisk-Users] CallerID retain on internal transfer

[Kevin P. Fleming]

Steve Davies wrote:

> In the cases previously mentioned, the user is doing an attended
> transfer using the handset features, and not Asterisk. I do not know
> whether SIP even allows the Caller ID to be changed at the point when
> two separate calls are bridged to one...

It does, but Asterisk does not currently support that behavior (even in
the development branch). I believe Olle's SIP transfer re-write may
provide this functionality when Asterisk 1.4 is released, but I am not
positive.
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Re: [Asterisk-Users] CallerID retain on internal transfer

[Steve Davies]

On 5/16/06, Avi Miller <[EMAIL PROTECTED]> wrote:

Michael J. Tubby B.Sc (Hons) G8TIC wrote:
> call then transfers it on to another extension transferee (recipeient)
> sees the Caller*ID

This behaviour changed in Asterisk 1.2 -- add "o" to your Dial options
and Asterisk will retain the original Caller ID on transfer.



This does not sound like quite the same thing "o" reverts to 1.0.x
behaviour, which is still not presenting the original number after an
attended transfer that is managed from the phone handset (perhaps it
does work if the *2 feature is used?)

In the cases previously mentioned, the user is doing an attended
transfer using the handset features, and not Asterisk. I do not know
whether SIP even allows the Caller ID to be changed at the point when
two separate calls are bridged to one...

i.e. The current behaviour (on our system) is:

 Caller -> Phone A (Caller's ID)

 Caller -> On Hold
  Phone A -> Phone B (A's ID)

 Phone A disconnected
 Caller -> Phone B (A's ID)

But the desired behaviour is:

 Caller -> Phone A (Caller's ID)

 Caller -> On Hold
  Phone A -> Phone B (A's ID)

 Phone A disconnected
 Caller -> Phone B (Caller's ID)

The "o" option looks as if it changes the number that is initially
presented on Phone B, and is only under Asterisk's control if you use
Asterisk's built-in attended transfer facility, otherwise the 'Phone
A' handset is responsible for the change. I suspect that the desired
behaviour is also only possible if 'Phone A' does the "Right Thing
(tm)"

*THINKS* In fact, when the phone is doing the attended transfer, the
caller-ID that should be presented AFTER a transfer will depend
entirely upon the final destination of the call. It may not even be
possible to change the CID upon transfer if the attended call went
(for example) out of a ZAP channel.

Looks like this is not an easy one to solve, but I am not 100% sure of
which party is responsible for what during this type of transfer so I
may be wrong...

Cheers,
Steve
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Re: [Asterisk-Users] CallerID retain on internal transfer

[Avi Miller]

Michael J. Tubby B.Sc (Hons) G8TIC wrote:
call then transfers it on to another extension transferee (recipeient) 
sees the Caller*ID


This behaviour changed in Asterisk 1.2 -- add "o" to your Dial options 
and Asterisk will retain the original Caller ID on transfer.


--
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Re: [Asterisk-Users] CallerID retain on internal transfer

[C F]

You sure you doing a blind xfer?

On 5/15/06, Michael J. Tubby B.Sc (Hons) G8TIC <[EMAIL PROTECTED]> wrote:

I'm seeing a similar thing...

We have Asterisk 1.2.7.1, Chan CAPI and a home-brewed IVR/Auto Attendant
that
places in-bound calls to role-based riinging groups like "sales", "support",
"admin" etc.
which works well, but from a 7960G phone (SIP 7.5) if the person that
answers a
call then transfers it on to another extension transferee (recipeient) sees
the Caller*ID
of the transferor (internal role-based extension number) and not the
Caller*ID of the
original calling party.

I took this to be a limitation of the way in which Cisco implement
Attended/Blind
transfers in the SIP firmware of the 7960G phone...?

Mike


- Original Message -
From: "Steve Davies" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Wednesday, May 10, 2006 10:52 AM
Subject: Re: [Asterisk-Users] CallerID retain on internal transfer


On 5/10/06, Joseph Rothstein <[EMAIL PROTECTED]> wrote:
> From what I have tested, using cisco phones and 1.2.5, the original
> callerID
> is not kept when making a transfer.
>
> Any other ideas?
>
We use SPA, snom and aastra phones, and I had assumed that this was a
limitation of the SIP protocol. I would be pleased to be told I am
wrong on this one :)

Regards,
Steve
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Re: [Asterisk-Users] CallerID retain on internal transfer

[Michael J. Tubby B.Sc \(Hons\) G8TIC]

I'm seeing a similar thing...

We have Asterisk 1.2.7.1, Chan CAPI and a home-brewed IVR/Auto Attendant 
that
places in-bound calls to role-based riinging groups like "sales", "support", 
"admin" etc.
which works well, but from a 7960G phone (SIP 7.5) if the person that 
answers a
call then transfers it on to another extension transferee (recipeient) sees 
the Caller*ID
of the transferor (internal role-based extension number) and not the 
Caller*ID of the

original calling party.

I took this to be a limitation of the way in which Cisco implement 
Attended/Blind

transfers in the SIP firmware of the 7960G phone...?

Mike


- Original Message - 
From: "Steve Davies" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Wednesday, May 10, 2006 10:52 AM
Subject: Re: [Asterisk-Users] CallerID retain on internal transfer


On 5/10/06, Joseph Rothstein <[EMAIL PROTECTED]> wrote:
From what I have tested, using cisco phones and 1.2.5, the original 
callerID

is not kept when making a transfer.

Any other ideas?


We use SPA, snom and aastra phones, and I had assumed that this was a
limitation of the SIP protocol. I would be pleased to be told I am
wrong on this one :)

Regards,
Steve
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Re: [Asterisk-Users] CallerID retain on internal transfer

[Steve Davies]

On 5/10/06, Joseph Rothstein <[EMAIL PROTECTED]> wrote:

From what I have tested, using cisco phones and 1.2.5, the original callerID
is not kept when making a transfer.

Any other ideas?


We use SPA, snom and aastra phones, and I had assumed that this was a
limitation of the SIP protocol. I would be pleased to be told I am
wrong on this one :)

Regards,
Steve
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Re: [Asterisk-Users] CallerID retain on internal transfer

[Eric \"ManxPower\" Wieling]

Joe wrote:

I was just looking through the Wiki for some info on how to retain the
original caller's callerid when make transfers to internal extensions, and I
came across the parameter below:

useincomingcalleridonzaptransfer=yes

There is nothing in the zapata.conf file from vers. 1.2.7 so I am wondering
if this is still a valid parameter. If not, does anyone know how I can do
this?



1.2 will keep the original Caller*ID when doing a transfer.  1.0.x did 
not do this.





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RE: [Asterisk-Users] CallerID retain on internal transfer

[Bevan Blackie]

I've been wondering about how to do what you're describing as well! Anyone
know how it can be done?

Regards,
Bevan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Sent: Monday, 8 May 2006 4:03 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] CallerID retain on internal transfer

I was just looking through the Wiki for some info on how to retain the
original caller's callerid when make transfers to internal extensions, and I
came across the parameter below:

useincomingcalleridonzaptransfer=yes

There is nothing in the zapata.conf file from vers. 1.2.7 so I am wondering
if this is still a valid parameter. If not, does anyone know how I can do
this?

Thanks,
Joe



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RE: [Asterisk-Users] CallerID Name problem

[Damon Estep]

Cross-posted to Dev on purpose.

I missed Astricon (not willing to give up a planned vacation!).

I looks like there is already code that detects the facility IE, as asterisk 
whines about "do not know what to do with a second ROSE component" (not 
verified in current version). That second ROSE component is the message that 
says the CNAM is coming in a facility IE according to the PM sessions I have 
been able to look at.

Why not put a flag in the config? WaitforfacilityIE=yes|no <- this would allow 
the user to decide on a per span basis if it is needed.

The Bellcore spec on CNAM states that the SETUP must contain an "information 
following" message element to indicate that the CNAM will arrive in the 
Facility IE.

Libpri could be written to look for the "information following" element in 
SETUP and wait for the FACIILITY IE if it is detected AND the user flag is set 
to wait for Facility IE.

Of course, when I say "could be written", I mean by someone like Matthew 
Frederickson, not me :). I do not have the required knowledge of the code to 
even try and attempt this.

I spent way too much time on this last year trying to use a TNT as a media 
gateway, now it (the TNT) is an outbound gateway for our 911 trunks and nothing 
else.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Monday, May 01, 2006 6:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] CallerID Name problem

You explained this very well thank you!!, We discussed (Astricon 2005 Anaheim) 
having LibPri either wait 1 second before passing the call on to asterisk, or 
waiting until CNAME was received, both ideas were not good as it will introduce 
delays for all instead of just those that needed it.

For the time being we will put the Wait(1) in.
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Damon Estep
> Sent: Monday, May 01, 2006 8:24 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] CallerID Name problem
> 
> The CDR is written at the end of the call, the SIP invite is generated
> when you tell it to.
> 
> You must wait for the CID name before you "dial" a SIP channel, as this is
> when the SIP invite is created.
> 
> Wait 1 seems to be long enough.
> 
> For analog TDM this is not an issue as the CNAM is sent between the first
> and second ring.
> 
> In nearly every case I have seen, the facility IE that contains the CNAM
> on most NI-2 ISDN PRIs, the Facility IE is not sent until after you send
> progress back to the telco.
> 
> Like this;
> 
> Telco sends ISDN SETUP message
> You send back ISDN PROGRESS (on a normal call)
> The Telco sends ISDN FACILITY INFORMATION ELEMENT (IE) containing CNAM
> 
> I have looked at several protocol monitoring session form both Nortel and
> lucent class 5 switches, and every one seems to send the facility IE in
> less than 500 milliseconds following SETUP assuming you (Zaptel) reply
> PROGRESS in a timely fashion, which does not appear to ever be an issue
> with Asterisk.
> 
> I have been lead to believe there are two reasons for sending the CNAM in
> the facility IE.
> 
> One explanation is that the SETUP message has a restriction on length, and
> in order to provide other features, the CNAM had to be moved.
> 
> The other explanation is that the Telco is saving costs by not dipping
> into LIDB (line information databases) for all calls where PROGRESS is not
> received from the endpoint (a high percentage of all calls). There is a
> cost associated with LIDB dips.
> 
> Both are believable explanations, but the result is the same, you have to
> wait 1 second!
> 
> Sure would be nice if asterisk could wait 500 milliseconds instead, since
> there is a need to put a "play 1 second of silence" in the first step of
> the IVR (if used) to avoid clipping the first phrase also. The combined 2
> seconds of "wait" gets noticed by some users/callers.
> 
> For the archives - The Lucent TNT (as of 11.0 firmware) is not capable of
> "waiting" for CNAM before generating the SIP invite, so it is not
> compatible with PRIs where the CNAM is sent in the facility IE instead of
> the SETUP message, which is the case with most telco switches with current
> software on them.
> 
> Damon
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Josh McAllister
> > Sent: Monday, May 01, 2006 4:25 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [A

RE: [Asterisk-Users] CallerID Name problem

[Alexander Lopez]

You explained this very well thank you!!, We discussed (Astricon 2005 Anaheim) 
having LibPri either wait 1 second before passing the call on to asterisk, or 
waiting until CNAME was received, both ideas were not good as it will introduce 
delays for all instead of just those that needed it.

For the time being we will put the Wait(1) in.
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Damon Estep
> Sent: Monday, May 01, 2006 8:24 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] CallerID Name problem
> 
> The CDR is written at the end of the call, the SIP invite is generated
> when you tell it to.
> 
> You must wait for the CID name before you "dial" a SIP channel, as this is
> when the SIP invite is created.
> 
> Wait 1 seems to be long enough.
> 
> For analog TDM this is not an issue as the CNAM is sent between the first
> and second ring.
> 
> In nearly every case I have seen, the facility IE that contains the CNAM
> on most NI-2 ISDN PRIs, the Facility IE is not sent until after you send
> progress back to the telco.
> 
> Like this;
> 
> Telco sends ISDN SETUP message
> You send back ISDN PROGRESS (on a normal call)
> The Telco sends ISDN FACILITY INFORMATION ELEMENT (IE) containing CNAM
> 
> I have looked at several protocol monitoring session form both Nortel and
> lucent class 5 switches, and every one seems to send the facility IE in
> less than 500 milliseconds following SETUP assuming you (Zaptel) reply
> PROGRESS in a timely fashion, which does not appear to ever be an issue
> with Asterisk.
> 
> I have been lead to believe there are two reasons for sending the CNAM in
> the facility IE.
> 
> One explanation is that the SETUP message has a restriction on length, and
> in order to provide other features, the CNAM had to be moved.
> 
> The other explanation is that the Telco is saving costs by not dipping
> into LIDB (line information databases) for all calls where PROGRESS is not
> received from the endpoint (a high percentage of all calls). There is a
> cost associated with LIDB dips.
> 
> Both are believable explanations, but the result is the same, you have to
> wait 1 second!
> 
> Sure would be nice if asterisk could wait 500 milliseconds instead, since
> there is a need to put a "play 1 second of silence" in the first step of
> the IVR (if used) to avoid clipping the first phrase also. The combined 2
> seconds of "wait" gets noticed by some users/callers.
> 
> For the archives - The Lucent TNT (as of 11.0 firmware) is not capable of
> "waiting" for CNAM before generating the SIP invite, so it is not
> compatible with PRIs where the CNAM is sent in the facility IE instead of
> the SETUP message, which is the case with most telco switches with current
> software on them.
> 
> Damon
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> > -----Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Josh McAllister
> > Sent: Monday, May 01, 2006 4:25 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [Asterisk-Users] CallerID Name problem
> >
> >
> > >From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora
> > >Sent: Monday, May 01, 2006 3:06 PM
> > >To: Asterisk Users Mailing List - Non-Commercial Discussion
> > >Subject: Re: [Asterisk-Users] CallerID Name problem
> > >
> > >Do you get caller ID number?  If so, WAITing is not going to help,
> since
> > you already get the info.  If you >get caller ID number, then your telco
> > is not >sending the name.
> >
> > This is not necessarily true. I've always gotten cID number, but only
> > recently when I added a wait(1) did I start getting channel vars
> populated
> > with cID Name. Same as Eric, I was getting cID Name in the CDR records
> all
> > along as well.
> >
> > Eric -- Go ahead and give it a shot... even if you are getting the cID
> > number. This will likely fix your problem.
> >
> > Josh McAllister
> >
> >
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] CallerID Name problem

[Alexander Lopez]

Title: RE: [Asterisk-Users] CallerID Name problem








Glad to help. It is NOT a BUG, but it
works that way due to the way CallerID name is transmited to the customer with
some Telcos

 

 











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Monday, May 01, 2006 6:29 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
CallerID Name problem



 

That worked GREAT 

Thank you so so MUCH for your help!!

 







From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Monday, May 01, 2006 5:06 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
CallerID Name problem

You don't need the answer, But you need
the wait. CallerID Name comes over the FACILITY messge many times and it takes
a slpit second for it to come in.

 



 







From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Monday, May 01, 2006 4:34 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
CallerID Name problem

Do
you wait before or after the answer? Do you even need the answer?



 -Original Message-
From:   Alexander Lopez [mailto:[EMAIL PROTECTED]]
Sent:   Mon May 01 14:26:49 2006
To: Asterisk Users Mailing
 List - Non-Commercial Discussion
Subject:        RE: [Asterisk-Users]
CallerID Name problem

How are the calls coming into the PBX. PRI? If so add a Wait(1) before
your try ringing the SIP channel.


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Hall, Eric M.
> Sent: Monday, May 01, 2006 12:37 PM
> To: Asterisk Users Mailing List - Non-Commercial
 Discussion
> Subject: [Asterisk-Users] CallerID Name problem
>
>
> I'm having trouble getting callerid name to show up on my phones
(Cisco
> 7960 and a few softphones)
> When I look in the CDR database I see the name but not on any phone
when
> being called.
>
> I'm running
> Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC
>
>
> Any help would be great !
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] CallerID Name problem

[Damon Estep]

The CDR is written at the end of the call, the SIP invite is generated when you 
tell it to.

You must wait for the CID name before you "dial" a SIP channel, as this is when 
the SIP invite is created.

Wait 1 seems to be long enough.

For analog TDM this is not an issue as the CNAM is sent between the first and 
second ring.

In nearly every case I have seen, the facility IE that contains the CNAM on 
most NI-2 ISDN PRIs, the Facility IE is not sent until after you send progress 
back to the telco.

Like this;

Telco sends ISDN SETUP message
You send back ISDN PROGRESS (on a normal call)
The Telco sends ISDN FACILITY INFORMATION ELEMENT (IE) containing CNAM

I have looked at several protocol monitoring session form both Nortel and 
lucent class 5 switches, and every one seems to send the facility IE in less 
than 500 milliseconds following SETUP assuming you (Zaptel) reply PROGRESS in a 
timely fashion, which does not appear to ever be an issue with Asterisk.

I have been lead to believe there are two reasons for sending the CNAM in the 
facility IE.

One explanation is that the SETUP message has a restriction on length, and in 
order to provide other features, the CNAM had to be moved.

The other explanation is that the Telco is saving costs by not dipping into 
LIDB (line information databases) for all calls where PROGRESS is not received 
from the endpoint (a high percentage of all calls). There is a cost associated 
with LIDB dips.

Both are believable explanations, but the result is the same, you have to wait 
1 second!

Sure would be nice if asterisk could wait 500 milliseconds instead, since there 
is a need to put a "play 1 second of silence" in the first step of the IVR (if 
used) to avoid clipping the first phrase also. The combined 2 seconds of "wait" 
gets noticed by some users/callers.

For the archives - The Lucent TNT (as of 11.0 firmware) is not capable of 
"waiting" for CNAM before generating the SIP invite, so it is not compatible 
with PRIs where the CNAM is sent in the facility IE instead of the SETUP 
message, which is the case with most telco switches with current software on 
them.

Damon












> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Josh McAllister
> Sent: Monday, May 01, 2006 4:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] CallerID Name problem
> 
> 
> >From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora
> >Sent: Monday, May 01, 2006 3:06 PM
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
> >Subject: Re: [Asterisk-Users] CallerID Name problem
> >
> >Do you get caller ID number?  If so, WAITing is not going to help, since
> you already get the info.  If you >get caller ID number, then your telco
> is not >sending the name.
> 
> This is not necessarily true. I've always gotten cID number, but only
> recently when I added a wait(1) did I start getting channel vars populated
> with cID Name. Same as Eric, I was getting cID Name in the CDR records all
> along as well.
> 
> Eric -- Go ahead and give it a shot... even if you are getting the cID
> number. This will likely fix your problem.
> 
> Josh McAllister
> 
> 
> ___
> --Bandwidth and Colocation provided by Easynews.com --
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> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] CallerID Name problem

[Lacy Moore - Aspendora]

Good to know.  I always thought that it was all sent along the D channel in the call setup.  But, until Wednesday, I'm just talking.  After Wednesday, I'll have my PRI.  At least now I know to include the wait.
 
Thanks! 
On 5/1/06, Josh McAllister <[EMAIL PROTECTED]> wrote:
>From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED]] On Behalf Of Lacy Moore - Aspendora>Sent: Monday, May 01, 2006 3:06 PM>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] CallerID Name problem>>Do you get caller ID number? If so, WAITing is not going to help, since you already get the info. If you >get caller ID number, then your telco is not >sending the name.
This is not necessarily true. I've always gotten cID number, but only recently when I added a wait(1) did I start getting channel vars populated with cID Name. Same as Eric, I was getting cID Name in the CDR records all along as well.
Eric -- Go ahead and give it a shot... even if you are getting the cID number. This will likely fix your problem.Josh McAllister___--Bandwidth and Colocation provided by 
Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users
-- Lacy MooreAspendora, Inc. 
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RE: [Asterisk-Users] CallerID Name problem

[Hall, Eric M.]

Title: RE: [Asterisk-Users] CallerID Name problem



That worked GREAT 
Thank you so so MUCH for your 
help!!


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander 
LopezSent: Monday, May 01, 2006 5:06 PMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] 
CallerID Name problem

You don't need the answer, But you need the wait. 
CallerID Name comes over the FACILITY messge many times and it takes a slpit 
second for it to come in.
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Hall, 
  Eric M.Sent: Monday, May 01, 2006 4:34 PMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] CallerID Name problem
  
  Do you wait before or after the answer? Do you even need the 
  answer? -Original Message-From:   
  Alexander Lopez [mailto:[EMAIL PROTECTED]]Sent:   
  Mon May 01 14:26:49 2006To: Asterisk Users Mailing 
  List - Non-Commercial 
  DiscussionSubject:        RE: 
  [Asterisk-Users] CallerID Name problemHow are the calls coming into 
  the PBX. PRI? If so add a Wait(1) beforeyour try ringing the SIP 
  channel.> -Original Message-> From: 
  [EMAIL PROTECTED] [mailto:asterisk-users-> 
  [EMAIL PROTECTED] On Behalf Of Hall, Eric M.> Sent: Monday, May 
  01, 2006 12:37 PM> To: Asterisk Users Mailing List - Non-Commercial 
  Discussion> Subject: [Asterisk-Users] CallerID Name 
  problem>>> I'm having trouble getting callerid name to 
  show up on my phones(Cisco> 7960 and a few softphones)> When 
  I look in the CDR database I see the name but not on any phonewhen> 
  being called.>> I'm running> Asterisk SVN-trunk-r7498 
  built on 2006-04-30 15:11:39 UTC>>> Any help would be 
  great !> ___> 
  --Bandwidth and Colocation provided by Easynews.com -->> 
  Asterisk-Users mailing list> To UNSUBSCRIBE or update options 
  visit:>    http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth 
  and Colocation provided by Easynews.com --Asterisk-Users mailing 
  listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] CallerID Name problem

[Josh McAllister]

>From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - 
>Aspendora
>Sent: Monday, May 01, 2006 3:06 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] CallerID Name problem
>
>Do you get caller ID number?  If so, WAITing is not going to help, since you 
>already get the info.  If you >get caller ID number, then your telco is not 
>>sending the name.

This is not necessarily true. I've always gotten cID number, but only recently 
when I added a wait(1) did I start getting channel vars populated with cID 
Name. Same as Eric, I was getting cID Name in the CDR records all along as well.

Eric -- Go ahead and give it a shot... even if you are getting the cID number. 
This will likely fix your problem.

Josh McAllister
 

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RE: Spam? Re: [Asterisk-Users] CallerID Name problem

[Hall, Eric M.]

Title: RE: Spam? Re: [Asterisk-Users] CallerID Name problem






I'm getting Number but when I look at the CDR database. I do see the name



 -Original Message-
From:   Lacy Moore - Aspendora [mailto:[EMAIL PROTECTED]]
Sent:   Mon May 01 17:10:26 2006
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:    Spam? Re: [Asterisk-Users] CallerID Name problem

Do you get caller ID number?  If so, WAITing is not going to help, since you
already get the info.  If you get caller ID number, then your telco is not
sending the name.

On 5/1/06, Hall, Eric M. <[EMAIL PROTECTED]> wrote:
>
>  Do you wait before or after the answer? Do you even need the answer?
>
>
>
>
>  -Original Message-
> From:   Alexander Lopez [mailto:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
> ]
> Sent:   Mon May 01 14:26:49 2006
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject:    RE: [Asterisk-Users] CallerID Name problem
>
> How are the calls coming into the PBX. PRI? If so add a Wait(1) before
> your try ringing the SIP channel.
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Hall, Eric M.
> > Sent: Monday, May 01, 2006 12:37 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] CallerID Name problem
> >
> >
> > I'm having trouble getting callerid name to show up on my phones
> (Cisco
> > 7960 and a few softphones)
> > When I look in the CDR database I see the name but not on any phone
> when
> > being called.
> >
> > I'm running
> > Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC
> >
> >
> > Any help would be great !
> > ___
> > --Bandwidth and Colocation provided by Easynews.com<http://easynews.com/>--
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> ___
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>
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> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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> To UNSUBSCRIBE or update options visit:
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>
>
>


--
Lacy Moore
Aspendora, Inc.




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RE: [Asterisk-Users] CallerID Name problem

[Hall, Eric M.]

Title: RE: [Asterisk-Users] CallerID Name problem






Thanks will try that tonight.

Thanks again



 -Original Message-
From:   Alexander Lopez [mailto:[EMAIL PROTECTED]]
Sent:   Mon May 01 17:07:43 2006
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:    RE: [Asterisk-Users] CallerID Name problem

You don't need the answer, But you need the wait. CallerID Name comes
over the FACILITY messge many times and it takes a slpit second for it
to come in.





    From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Hall, Eric
M.
    Sent: Monday, May 01, 2006 4:34 PM
    To: Asterisk Users Mailing List - Non-Commercial Discussion
    Subject: RE: [Asterisk-Users] CallerID Name problem
   
   

    Do you wait before or after the answer? Do you even need the
answer?
   
   
   
 -Original Message-
    From:   Alexander Lopez [mailto:[EMAIL PROTECTED]]
    Sent:   Mon May 01 14:26:49 2006
    To: Asterisk Users Mailing List - Non-Commercial Discussion
    Subject:        RE: [Asterisk-Users] CallerID Name problem
   
    How are the calls coming into the PBX. PRI? If so add a Wait(1)
before
    your try ringing the SIP channel.
   
   
    > -Original Message-
    > From: [EMAIL PROTECTED]
[mailto:asterisk-users-
    > [EMAIL PROTECTED] On Behalf Of Hall, Eric M.
    > Sent: Monday, May 01, 2006 12:37 PM
    > To: Asterisk Users Mailing List - Non-Commercial Discussion
    > Subject: [Asterisk-Users] CallerID Name problem
    >
    >
    > I'm having trouble getting callerid name to show up on my
phones
    (Cisco
    > 7960 and a few softphones)
    > When I look in the CDR database I see the name but not on any
phone
    when
    > being called.
    >
    > I'm running
    > Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC
    >
    >
    > Any help would be great !
    > ___
    > --Bandwidth and Colocation provided by Easynews.com --
    >
    > Asterisk-Users mailing list
    > To UNSUBSCRIBE or update options visit:
    >    http://lists.digium.com/mailman/listinfo/asterisk-users
    ___
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    To UNSUBSCRIBE or update options visit:
       http://lists.digium.com/mailman/listinfo/asterisk-users
   





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Re: [Asterisk-Users] CallerID Name problem

[Lacy Moore - Aspendora]

Do you get caller ID number?  If so, WAITing is not going to help, since you already get the info.  If you get caller ID number, then your telco is not sending the name.
On 5/1/06, Hall, Eric M. <[EMAIL PROTECTED]> wrote:


Do you wait before or after the answer? Do you even need the answer?
 -Original Message-From:   Alexander Lopez [
mailto:[EMAIL PROTECTED]]Sent:   Mon May 01 14:26:49 2006To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:        RE: [Asterisk-Users] CallerID Name problemHow are the calls coming into the PBX. PRI? If so add a Wait(1) beforeyour try ringing the SIP channel.
> -Original Message-> From: [EMAIL PROTECTED]
 [mailto:asterisk-users-> 
[EMAIL PROTECTED]] On Behalf Of Hall, Eric M.> Sent: Monday, May 01, 2006 12:37 PM> To: Asterisk Users Mailing List - Non-Commercial Discussion> Subject: [Asterisk-Users] CallerID Name problem
>>> I'm having trouble getting callerid name to show up on my phones(Cisco> 7960 and a few softphones)> When I look in the CDR database I see the name but not on any phonewhen
> being called.>> I'm running> Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC>>> Any help would be great !> ___
> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users___
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RE: [Asterisk-Users] CallerID Name problem

[Alexander Lopez]

Title: RE: [Asterisk-Users] CallerID Name problem



You don't need the answer, But you need the wait. 
CallerID Name comes over the FACILITY messge many times and it takes a slpit 
second for it to come in.
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Hall, 
  Eric M.Sent: Monday, May 01, 2006 4:34 PMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] CallerID Name problem
  
  Do you wait before or after the answer? Do you even need the 
  answer? -Original Message-From:   
  Alexander Lopez [mailto:[EMAIL PROTECTED]]Sent:   
  Mon May 01 14:26:49 2006To: Asterisk Users Mailing 
  List - Non-Commercial 
  DiscussionSubject:        RE: 
  [Asterisk-Users] CallerID Name problemHow are the calls coming into 
  the PBX. PRI? If so add a Wait(1) beforeyour try ringing the SIP 
  channel.> -Original Message-> From: 
  [EMAIL PROTECTED] [mailto:asterisk-users-> 
  [EMAIL PROTECTED] On Behalf Of Hall, Eric M.> Sent: Monday, May 
  01, 2006 12:37 PM> To: Asterisk Users Mailing List - Non-Commercial 
  Discussion> Subject: [Asterisk-Users] CallerID Name 
  problem>>> I'm having trouble getting callerid name to 
  show up on my phones(Cisco> 7960 and a few softphones)> When 
  I look in the CDR database I see the name but not on any phonewhen> 
  being called.>> I'm running> Asterisk SVN-trunk-r7498 
  built on 2006-04-30 15:11:39 UTC>>> Any help would be 
  great !> ___> 
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RE: [Asterisk-Users] CallerID Name problem

[Hall, Eric M.]

Title: RE: [Asterisk-Users] CallerID Name problem






Do you wait before or after the answer? Do you even need the answer?



 -Original Message-
From:   Alexander Lopez [mailto:[EMAIL PROTECTED]]
Sent:   Mon May 01 14:26:49 2006
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:    RE: [Asterisk-Users] CallerID Name problem

How are the calls coming into the PBX. PRI? If so add a Wait(1) before
your try ringing the SIP channel.


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Hall, Eric M.
> Sent: Monday, May 01, 2006 12:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] CallerID Name problem
>
>
> I'm having trouble getting callerid name to show up on my phones
(Cisco
> 7960 and a few softphones)
> When I look in the CDR database I see the name but not on any phone
when
> being called.
>
> I'm running
> Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC
>
>
> Any help would be great !
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RE: [Asterisk-Users] CallerID Name problem

[Alexander Lopez]

How are the calls coming into the PBX. PRI? If so add a Wait(1) before
your try ringing the SIP channel.


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Hall, Eric M.
> Sent: Monday, May 01, 2006 12:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] CallerID Name problem
> 
> 
> I'm having trouble getting callerid name to show up on my phones
(Cisco
> 7960 and a few softphones)
> When I look in the CDR database I see the name but not on any phone
when
> being called.
> 
> I'm running
> Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC
> 
> 
> Any help would be great !
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RE: [Asterisk-Users] CallerID Name problem

[Hall, Eric M.]

Using SIP and SCCP. The softphone uses SIP.

Doing a debug  I see no name being sent. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kevin ling
Sent: Monday, May 01, 2006 2:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] CallerID Name problem

Hi,

What protocol for your 7960 phone? SCCP or SIP? You can turn on the SIP
debug on CLI to make sure the callerid and name pass to your phone. 

Kevin 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Tuesday, May 02, 2006 12:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] CallerID Name problem

 
I'm having trouble getting callerid name to show up on my phones (Cisco
7960 and a few softphones) When I look in the CDR database I see the
name but not on any phone when being called.

I'm running
Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC 


Any help would be great !



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RE: [Asterisk-Users] CallerID Name problem

[kevin ling]

Hi,

What protocol for your 7960 phone? SCCP or SIP? You can turn on the SIP
debug on CLI to make sure the callerid and name pass to your phone. 

Kevin 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Tuesday, May 02, 2006 12:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] CallerID Name problem

 
I'm having trouble getting callerid name to show up on my phones (Cisco 7960
and a few softphones) When I look in the CDR database I see the name but not
on any phone when being called.

I'm running
Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC 


Any help would be great !



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Re: [Asterisk-Users] CallerID/variable setting.

[Anthony Rodgers]

Hi Ken,

Here is what we do, if it helps:

For incoming calls (we use some Centrex lines, but want to make them  
look like 4-digit locals):


; If it looks like one of ours, only show the last 4 digits
exten => s,40,GotoIf($["${CALLERIDNUM:0:8}" = "60498131"]?50:)
exten => s,50,SetCallerID("${CALLERIDNAME}" <${CALLERIDNUM:-4}>)

For outgoing calls:

; If it looks like the local has already been expanded to NANPA, skip  
to dialing

exten => s,3,GotoIf($["${ARG1:0:3}" = "604"]?20:)
; AllStream DIDs get a 604998 prefix, the rest get 604990
exten => s,4,GotoIf($["${ARG1:0:3}" = "303"]?:10)
exten => s,5,SetCallerID("${CALLERIDNAME}" <604998${CALLERIDNUM}>)
exten => s,6,Goto(20)
exten => s,10,SetCallerID("${CALLERIDNAME}" <604990${CALLERIDNUM}>)
exten => s,11,Goto(20)

Hope this helps - let me know if you need more details.

Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp



On 24-Apr-06, at 9:57 AM, Ken D'Ambrosio wrote:


Hey, all.  I'm trying to set my CID such that, internally, I see a
four-digit extension (which is also handy when checking VM), but
externally, I see the full 10-digit number.  So I plugged these  
lines into

my extensions.conf:

exten => _XXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2)
exten => _XXX,2,Set(CALLERIDNUM=6031234${CALLERIDNUM:1})
exten => _XXX,3,NoOp(${CALLERIDNUM})
exten => _XXX,4,Dial(${OUTBOUNDTRUNK}/${EXTEN})

(I wanted to test against my own extension, "1625"; if that worked, I
wanted to strip off the "1", and then prepend the 603-123-4 to my
remaining three digits.)

Which is all well and good -- until I actually try to use it.   
Then, I get:


-- Executing GotoIf("SIP/1625-f89a", "0?4:2") in new stack
-- Goto (internal,7654321,2)
-- Executing Set("SIP/1625-f89a", "CALLERIDNUM=6031234625") in  
new stack

-- Executing NoOp("SIP/1625-f89a", "1625") in new stack
-- Executing Dial("SIP/1625-f89a", "Zap/g1/7654321") in new stack

Why does my "NoOp" line show my 1625 extension, when CALLERIDNUM is  
-- as
far as I can tell -- being set to 6031234625?  (I looked against  
the "Set"

command page on the Wiki, and I think I'm doing it right.)

Asterisk 1.2.3, if that matters.

Thanks,

-Ken

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Re: [Asterisk-Users] Callerid matching in extensions.conf

[Michiel van Baak]

On 09:34, Wed 19 Apr 06, Douglas Garstang wrote:
> I'm using Asterisk 1.2.7.1. Has the callerid number matching functionality in 
> extensions.conf changed recently?
> 
> exten => ,1,NoOp(${CALLERID})
> 
> hestia*CLI> 
> -- Executing NoOp("SIP/2944093-d24d", ""Cletus the Slaw Jawed Yokel" 
> <2944093>") in new stack
>   == Auto fallthrough, channel 'SIP/2944093-d24d' status is 'UNKNOWN'
> 
> This does not match...
> exten => 2944093/,1,NoOp(${CALLERID})

try this: exten => /2944093,1,NoOp(${CALLERID})

> Can't figure out why It used to work.

Impossible.

-- 
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"

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Re: [Asterisk-Users] callerid name inboune from PRI

[Matthew Fredrickson]

It is in fact required for some implementations of callerid name.  It 
comes on a facility message that arrives after the call is setup.  It 
also can come in a display IE in the call setup.  It really depends on 
which way they are sending it.


Matthew Fredrickson

On Apr 11, 2006, at 12:49 PM, C F wrote:


No, I'm taking receiving CallerID name and *not* sending. and no on a
PRI wait should not be required for callerID to come in.

On 4/11/06, Jerry Jones <[EMAIL PROTECTED]> wrote:

I CAN VERIFY via aa dozen PRI from XO that yes indeed provide
incoming callerID on PRI. It arrives shortly after the setup message.
Hence the wait(1) required to display it.
Now if you are referring to sending caller name across PSTN - that
does NOT work since the terminating switch will do a CNAM lookup.

On Apr 10, 2006, at 10:55 PM, C F wrote:


On 4/10/06, Andres <[EMAIL PROTECTED]> wrote:

Steven wrote:


I switched PRI vendors recently, and one of my questions was "do
you provide caller ID name in addition to number?"
AT&T Local did not, But XO communications said they did.



You heard wrong.  We have multiple PRIs from XO and they DO NOT send
caller name.  We have discussed the issue with them on several
ocassions.  The sales people will say whatever they want, but the
tech
people who actually work in the switches know that caller name is 
not

supported.


I believe he didn't hear wrong, a lot of providers are now providing
CallerID with name over PRI. The tech people know it *is* supported
(all it is is a stupid simple lookup on the SS7 side of the 
equipment)
but wasn't done until now because the competition didn't offer it, 
but

now that the competition is offering it, everyone else is as well.




Before I call to complain, is there an setting to turn this on in
asterisk?
I want to make sure that I have my side covered before I call XO.

My current zaptel.conf is:

context=from-pstn
switchtype=national
pridialplan=unknown
prilocaldialplan=unknown
priindication = outofband
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
usecallingpres=yes
echocancel=yes
echocancelwhenbridged=no
group=0
callgroup=1
pickupgroup=1
accountcode=I
musiconhold=default
channel => 1-23









--
Andres
Technical Support
http://www.telesip.net

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Re: [Asterisk-Users] callerid name inboune from PRI

[Greg Oliver]

On Mon, 2006-04-10 at 22:42 -0400, Andres wrote:
> Steven wrote:
> You heard wrong.  We have multiple PRIs from XO and they DO NOT send 
> caller name.  We have discussed the issue with them on several 
> ocassions.  The sales people will say whatever they want, but the tech 
> people who actually work in the switches know that caller name is not 
> supported.
> 

I would say it depends on the market and location as well, since XO has
several flavors of switches and provisioning systems and has quite the
disparate network.  The ALGX infrastructure they purchased is very much
still in place, and did provide CNAME on PRIs for extra cost.

-Greg 

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Re: [Asterisk-Users] callerid name inboune from PRI

[C F]

No, I'm taking receiving CallerID name and *not* sending. and no on a
PRI wait should not be required for callerID to come in.

On 4/11/06, Jerry Jones <[EMAIL PROTECTED]> wrote:
> I CAN VERIFY via aa dozen PRI from XO that yes indeed provide
> incoming callerID on PRI. It arrives shortly after the setup message.
> Hence the wait(1) required to display it.
> Now if you are referring to sending caller name across PSTN - that
> does NOT work since the terminating switch will do a CNAM lookup.
>
> On Apr 10, 2006, at 10:55 PM, C F wrote:
>
> > On 4/10/06, Andres <[EMAIL PROTECTED]> wrote:
> >> Steven wrote:
> >>
> >>> I switched PRI vendors recently, and one of my questions was "do
> >>> you provide caller ID name in addition to number?"
> >>> AT&T Local did not, But XO communications said they did.
> >>>
> >>>
> >> You heard wrong.  We have multiple PRIs from XO and they DO NOT send
> >> caller name.  We have discussed the issue with them on several
> >> ocassions.  The sales people will say whatever they want, but the
> >> tech
> >> people who actually work in the switches know that caller name is not
> >> supported.
> >
> > I believe he didn't hear wrong, a lot of providers are now providing
> > CallerID with name over PRI. The tech people know it *is* supported
> > (all it is is a stupid simple lookup on the SS7 side of the equipment)
> > but wasn't done until now because the competition didn't offer it, but
> > now that the competition is offering it, everyone else is as well.
> >
> >>
> >>> Before I call to complain, is there an setting to turn this on in
> >>> asterisk?
> >>> I want to make sure that I have my side covered before I call XO.
> >>>
> >>> My current zaptel.conf is:
> >>>
> >>> context=from-pstn
> >>> switchtype=national
> >>> pridialplan=unknown
> >>> prilocaldialplan=unknown
> >>> priindication = outofband
> >>> signalling=pri_cpe
> >>> usecallerid=yes
> >>> hidecallerid=no
> >>> usecallingpres=yes
> >>> echocancel=yes
> >>> echocancelwhenbridged=no
> >>> group=0
> >>> callgroup=1
> >>> pickupgroup=1
> >>> accountcode=I
> >>> musiconhold=default
> >>> channel => 1-23
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>
> >>
> >> --
> >> Andres
> >> Technical Support
> >> http://www.telesip.net
> >>
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> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] callerid name inboune from PRI

[Jerry Jones]

I CAN VERIFY via aa dozen PRI from XO that yes indeed provide  
incoming callerID on PRI. It arrives shortly after the setup message.  
Hence the wait(1) required to display it.
Now if you are referring to sending caller name across PSTN - that  
does NOT work since the terminating switch will do a CNAM lookup.


On Apr 10, 2006, at 10:55 PM, C F wrote:


On 4/10/06, Andres <[EMAIL PROTECTED]> wrote:

Steven wrote:

I switched PRI vendors recently, and one of my questions was "do  
you provide caller ID name in addition to number?"

AT&T Local did not, But XO communications said they did.



You heard wrong.  We have multiple PRIs from XO and they DO NOT send
caller name.  We have discussed the issue with them on several
ocassions.  The sales people will say whatever they want, but the  
tech

people who actually work in the switches know that caller name is not
supported.


I believe he didn't hear wrong, a lot of providers are now providing
CallerID with name over PRI. The tech people know it *is* supported
(all it is is a stupid simple lookup on the SS7 side of the equipment)
but wasn't done until now because the competition didn't offer it, but
now that the competition is offering it, everyone else is as well.



Before I call to complain, is there an setting to turn this on in  
asterisk?

I want to make sure that I have my side covered before I call XO.

My current zaptel.conf is:

context=from-pstn
switchtype=national
pridialplan=unknown
prilocaldialplan=unknown
priindication = outofband
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
usecallingpres=yes
echocancel=yes
echocancelwhenbridged=no
group=0
callgroup=1
pickupgroup=1
accountcode=I
musiconhold=default
channel => 1-23









--
Andres
Technical Support
http://www.telesip.net

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Re: [Asterisk-Users] callerid name inboune from PRI

[C F]

On 4/10/06, Andres <[EMAIL PROTECTED]> wrote:
> Steven wrote:
>
> >I switched PRI vendors recently, and one of my questions was "do you provide 
> >caller ID name in addition to number?"
> >AT&T Local did not, But XO communications said they did.
> >
> >
> You heard wrong.  We have multiple PRIs from XO and they DO NOT send
> caller name.  We have discussed the issue with them on several
> ocassions.  The sales people will say whatever they want, but the tech
> people who actually work in the switches know that caller name is not
> supported.

I believe he didn't hear wrong, a lot of providers are now providing
CallerID with name over PRI. The tech people know it *is* supported
(all it is is a stupid simple lookup on the SS7 side of the equipment)
but wasn't done until now because the competition didn't offer it, but
now that the competition is offering it, everyone else is as well.

>
> >Before I call to complain, is there an setting to turn this on in asterisk?
> >I want to make sure that I have my side covered before I call XO.
> >
> >My current zaptel.conf is:
> >
> >context=from-pstn
> >switchtype=national
> >pridialplan=unknown
> >prilocaldialplan=unknown
> >priindication = outofband
> >signalling=pri_cpe
> >usecallerid=yes
> >hidecallerid=no
> >usecallingpres=yes
> >echocancel=yes
> >echocancelwhenbridged=no
> >group=0
> >callgroup=1
> >pickupgroup=1
> >accountcode=I
> >musiconhold=default
> >channel => 1-23
> >
> >
> >
> >
> >
> >
>
>
> --
> Andres
> Technical Support
> http://www.telesip.net
>
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Re: [Asterisk-Users] callerid name inboune from PRI

[Andres]

Steven wrote:


I switched PRI vendors recently, and one of my questions was "do you provide caller 
ID name in addition to number?"
AT&T Local did not, But XO communications said they did.
 

You heard wrong.  We have multiple PRIs from XO and they DO NOT send 
caller name.  We have discussed the issue with them on several 
ocassions.  The sales people will say whatever they want, but the tech 
people who actually work in the switches know that caller name is not 
supported.



Before I call to complain, is there an setting to turn this on in asterisk?
I want to make sure that I have my side covered before I call XO.

My current zaptel.conf is:

context=from-pstn
switchtype=national
pridialplan=unknown
prilocaldialplan=unknown
priindication = outofband
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
usecallingpres=yes
echocancel=yes
echocancelwhenbridged=no
group=0
callgroup=1
pickupgroup=1
accountcode=I
musiconhold=default
channel => 1-23




 




--
Andres
Technical Support
http://www.telesip.net

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RE: [Asterisk-Users] callerid name inboune from PRI

[Alexander Lopez]

There is nothing you really need 'to do' if your PRI is working already,
If you are able to receive and make calls your D-Channel is functioning
properly.  In the case of CallerID, some telcos provide this extra
function via the FACILITY messages instead of the SETUP messages, If
that is the case, you will get no Name but you will get a number. IT
simply means that Asterisk answered the call with the SETUP message but
was unable to read in the CALLERID Name to pass on to your devices
because it comes later on in the call via the FACILITY.

Add a Wait(2) before you answer the call for your PRI, see if that
helps.

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Steven
> Sent: Monday, April 10, 2006 8:57 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] callerid name inboune from PRI
> 
> I switched PRI vendors recently, and one of my questions was 
> "do you provide caller ID name in addition to number?"
> AT&T Local did not, But XO communications said they did.
> 
> Before I call to complain, is there an setting to turn this 
> on in asterisk?
> I want to make sure that I have my side covered before I call XO.
> 
> My current zaptel.conf is:
> 
> context=from-pstn
> switchtype=national
> pridialplan=unknown
> prilocaldialplan=unknown
> priindication = outofband
> signalling=pri_cpe
> usecallerid=yes
> hidecallerid=no
> usecallingpres=yes
> echocancel=yes
> echocancelwhenbridged=no
> group=0
> callgroup=1
> pickupgroup=1
> accountcode=I
> musiconhold=default
> channel => 1-23
> 
> 
> 
> 
> --
> --
> Steven
> 
> http://www.glimasoutheast.org
> 
> 
> 
> 
> 
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Re: [Asterisk-Users] CallerID

[Jay Milk]

Steve Totaro wrote:
I have no idea what the issues here are, nor do I care but I do have a 
question about this statement "Since you are selling support for this 
script, that qualifies as commercial

use and is expressly prohibited by the micro-license included in the
original script."  Is this an accurate statement?  Don't alot of firms 
provide support for opensource software?  I thought it was only 
considered commercial use if it is sold as a product and source was 
not supplied.  I have not read the microlicense but couldnt Microsoft 
include such a microlicense and prevent any other firms from 
supporting MS products?


Thanks,
Steve
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Under "normal" circumstances, you would be perfectly right.  The concern 
here is that this is open-source that was intentionally or ignorantly 
dis-credited and modified, possibly broken.  For this modified/broken 
script, this dishonest company now offers paid support.  This would 
classify as commercial (ab-) use -- it's akin to distributing asterisk 
as your own, without giving credit to Digium or Mark Spencer, and thus 
intentionally closing other, unpaid,  support venues. 

The possibility that OCG/Generation D may have broken the script in an 
attempt to "fix" an error that wasn't there, calls into question the 
good intentions they claim.  The further fact that the "many times" I 
was allegedly contacted turned out to be a single 
"how-about-this-feature?" email, solidifies my conviction that 
profiteering was their first and only intention.  This was -- and, hey, 
OCG listen! -- *IS* specifically forbidden regarding this script.


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RE: [Asterisk-Users] CallerID

[Technical Support]

Miles,

I think this is a limitation of the AGI - I don't believe that asterisk can
fork a new process.  If so, that would be interesting!

The script uses Wget - I believe we can set a timeout so that your system
doesn't hang waiting for the HTTP response.  Let me know if that would solve
your problem.  (You can also set the WGET timeout in your system's config I
believe)

MD

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Miles Scruggs
Sent: Sunday, April 09, 2006 2:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID

I just installed the script, it seems to hang while going out to the web.
Is there someway to have it run in the background while a
background() is playing or something like that?

Thanks

Miles


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Re: [Asterisk-Users] CallerID

[Steve Totaro]

Jay Milk wrote:

Michelle,

you sent a single message containing suggestions to me on 11/02/2005.  
Your claim to have contacted me "many times" is clearly false.  Due to 
demands outside the asterisk world, I have not been monitoring the 
list, but I doubt that should have been necessary, considering that 
contact information and even a mailing list are available for 
cid-rewrite.
Nobody at all contacted me about reverse lookup not working, and since 
the script has published was "in production" here for over a month as 
well as on many other servers, I have to question the validity of that 
claim.


My comments about spelling and commercial use are very productive.   
Much unlike you seem to, I take pride in the work I do, and being 
associated with something so poorly written as your changes to the 
readme is an embarrassment to both of parties.  Additionally, 
programming is a very exact process, and the quality of your 
documentation betrays your ability.  I do maintain that you are in 
fact misleading potential downloaders on the origin of the script.  
You have removed contact information and effectively taken credit for 
this work.
You furthermore are offering paid support, which qualifies as 
"commercial use" and you have neither asked for nor been granted 
permission for commercial use of my intellectual property.


Expect no help or cooperation from me in integrating your changes -- 
your changes are hacks at best, and a far cry from the properly 
architected changes I have planned and partially integrated in my 
production script.


In the meantime, either remove the download of this bastardized script 
from your site, or add full contact information back into the readme 
file and offer FREE support for it.  Please comply within 72 hours of 
receipt of this message.


Regards,
-- Jay Milk

Technical Support wrote:

Jay,

I contacted you many times regarding the script, whether you planned to
update it, suggestions for features, etc.  You did not respond to any 
of my

later emails.  Similarly, there was discussion between list members
regarding whether this script was orphaned after changes to 411.com 
made the
reverse lookup non-functional - for a long time.  I assumed 
responsibility

for updating the script as a courtesy to Asterisk users.

Your comments about spelling, resale, etc. are abrasive, 
unproductive, and

misleading.  Not only is the script available without charge on the web
site, credit to you remains with the script - in fact even the 
download link

of the web site gives you credit!  And of course, why would I update the
script and then encourage users to download an older version from 
another

site?

If you have time to dedicate to the cid_rewrite project terrific - I 
would
rather see one stream benefit all users.  Let's work to integrate 
changes
going forward.  If you would prefer not to, I would be pleased to 
rename the

script so that there is no confusion.

Regards,
Michelle

-Original Message-
From: Jay Milk [mailto:[EMAIL PROTECTED] Sent: Saturday, April 
08, 2006 1:05 AM

To: Technical Support; Asterisk Users Mailing List - Non-Commercial
Discussion; Michael Stahl
Subject: Re: [Asterisk-Users] CallerID

Michelle,

1. Courtesy would suggest that you would have contacted the author of 
the
script (me) to ask permission to modify this and host it elsewhere. 
2. What possessed you to remove ALL credits and original download 
location

from the readme file?  Are you trying to pawn other people's work off as
yours?
3. It's not exactly smart to continue someone else's versioning 
scheme if

you're intending to make a "fork". 4. Your spelling is atrocious.
5. The script is not orphaned, even though you seem to imply this in the
readme file.

Since you are selling support for this script, that qualifies as 
commercial

use and is expressly prohibited by the micro-license included in the
original script.  Please remove it from your download page until you 
have
made arrangements for further distribution with me.  I'm utterly 
amazed at

the bad form I see here.

Downloads of the original script are available here:
http://www.muware.com/asterisk/

The script is alive and working well, and I've made various 
enhancements to

user-requests in the recent past.

-- JM


Technical Support wrote:
 

Miles,

You can also download cid_rewrite from www.generationd.com  This PHP 
script looks up the phone numbers in a local MySQL table, and/or 
uses reverse 411 on the web to lookup the name, and/or more options.


Michelle

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of 
Alejandro Vargas

Sent: Friday, April 07, 2006 4:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID

2006/4/7, Miles Scruggs <[EMAIL PROTECTED]>:
 
Could you give me an example code of how this would work, and how 
to setup the 

Re: [Asterisk-Users] CallerID

[Miles Scruggs]

I just installed the script, it seems to hang while going out to the 
web.  Is there someway to have it run in the background while a 
background() is playing or something like that?


Thanks

Miles

Jay Milk wrote:

Michelle,

1. Courtesy would suggest that you would have contacted the author of 
the script (me) to ask permission to modify this and host it 
elsewhere. 2. What possessed you to remove ALL credits and original 
download location from the readme file?  Are you trying to pawn other 
people's work off as yours?
3. It's not exactly smart to continue someone else's versioning scheme 
if you're intending to make a "fork". 4. Your spelling is atrocious.
5. The script is not orphaned, even though you seem to imply this in 
the readme file.


Since you are selling support for this script, that qualifies as 
commercial use and is expressly prohibited by the micro-license 
included in the original script.  Please remove it from your download 
page until you have made arrangements for further distribution with 
me.  I'm utterly amazed at the bad form I see here.


Downloads of the original script are available here:
http://www.muware.com/asterisk/

The script is alive and working well, and I've made various 
enhancements to user-requests in the recent past.



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Re: [Asterisk-Users] CallerID

[Jay Milk]

Michelle,

you sent a single message containing suggestions to me on 11/02/2005.  
Your claim to have contacted me "many times" is clearly false.  Due to 
demands outside the asterisk world, I have not been monitoring the list, 
but I doubt that should have been necessary, considering that contact 
information and even a mailing list are available for cid-rewrite. 

Nobody at all contacted me about reverse lookup not working, and since 
the script has published was "in production" here for over a month as 
well as on many other servers, I have to question the validity of that 
claim.


My comments about spelling and commercial use are very productive.   
Much unlike you seem to, I take pride in the work I do, and being 
associated with something so poorly written as your changes to the 
readme is an embarrassment to both of parties.  Additionally, 
programming is a very exact process, and the quality of your 
documentation betrays your ability.  I do maintain that you are in fact 
misleading potential downloaders on the origin of the script.  You have 
removed contact information and effectively taken credit for this work. 

You furthermore are offering paid support, which qualifies as 
"commercial use" and you have neither asked for nor been granted 
permission for commercial use of my intellectual property.


Expect no help or cooperation from me in integrating your changes -- 
your changes are hacks at best, and a far cry from the properly 
architected changes I have planned and partially integrated in my 
production script.


In the meantime, either remove the download of this bastardized script 
from your site, or add full contact information back into the readme 
file and offer FREE support for it.  Please comply within 72 hours of 
receipt of this message.


Regards,
-- Jay Milk

Technical Support wrote:

Jay,

I contacted you many times regarding the script, whether you planned to
update it, suggestions for features, etc.  You did not respond to any of my
later emails.  Similarly, there was discussion between list members
regarding whether this script was orphaned after changes to 411.com made the
reverse lookup non-functional - for a long time.  I assumed responsibility
for updating the script as a courtesy to Asterisk users.

Your comments about spelling, resale, etc. are abrasive, unproductive, and
misleading.  Not only is the script available without charge on the web
site, credit to you remains with the script - in fact even the download link
of the web site gives you credit!  And of course, why would I update the
script and then encourage users to download an older version from another
site?

If you have time to dedicate to the cid_rewrite project terrific - I would
rather see one stream benefit all users.  Let's work to integrate changes
going forward.  If you would prefer not to, I would be pleased to rename the
script so that there is no confusion.

Regards,
Michelle

-Original Message-
From: Jay Milk [mailto:[EMAIL PROTECTED] 
Sent: Saturday, April 08, 2006 1:05 AM

To: Technical Support; Asterisk Users Mailing List - Non-Commercial
Discussion; Michael Stahl
Subject: Re: [Asterisk-Users] CallerID

Michelle,

1. Courtesy would suggest that you would have contacted the author of the
script (me) to ask permission to modify this and host it elsewhere. 
2. What possessed you to remove ALL credits and original download location

from the readme file?  Are you trying to pawn other people's work off as
yours?
3. It's not exactly smart to continue someone else's versioning scheme if
you're intending to make a "fork". 
4. Your spelling is atrocious.

5. The script is not orphaned, even though you seem to imply this in the
readme file.

Since you are selling support for this script, that qualifies as commercial
use and is expressly prohibited by the micro-license included in the
original script.  Please remove it from your download page until you have
made arrangements for further distribution with me.  I'm utterly amazed at
the bad form I see here.

Downloads of the original script are available here:
http://www.muware.com/asterisk/

The script is alive and working well, and I've made various enhancements to
user-requests in the recent past.

-- JM


Technical Support wrote:
  

Miles,

You can also download cid_rewrite from www.generationd.com  This PHP 
script looks up the phone numbers in a local MySQL table, and/or uses 
reverse 411 on the web to lookup the name, and/or more options.


Michelle

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of 
Alejandro Vargas

Sent: Friday, April 07, 2006 4:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID

2006/4/7, Miles Scruggs <[EMAIL PROTECTED]>:
  

Could you give me an example code of how this would work, and how to 
setup the database, I'm pretty new and wh

Re: [Asterisk-Users] CallerID

[Jay Milk]

Michelle,

1. Courtesy would suggest that you would have contacted the author of 
the script (me) to ask permission to modify this and host it elsewhere. 
2. What possessed you to remove ALL credits and original download 
location from the readme file?  Are you trying to pawn other people's 
work off as yours?
3. It's not exactly smart to continue someone else's versioning scheme 
if you're intending to make a "fork". 
4. Your spelling is atrocious.
5. The script is not orphaned, even though you seem to imply this in the 
readme file.


Since you are selling support for this script, that qualifies as 
commercial use and is expressly prohibited by the micro-license included 
in the original script.  Please remove it from your download page until 
you have made arrangements for further distribution with me.  I'm 
utterly amazed at the bad form I see here.


Downloads of the original script are available here:
http://www.muware.com/asterisk/

The script is alive and working well, and I've made various enhancements 
to user-requests in the recent past.


-- JM


Technical Support wrote:

Miles,

You can also download cid_rewrite from www.generationd.com  This PHP script
looks up the phone numbers in a local MySQL table, and/or uses reverse 411
on the web to lookup the name, and/or more options.

Michelle 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Vargas
Sent: Friday, April 07, 2006 4:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID

2006/4/7, Miles Scruggs <[EMAIL PROTECTED]>:
  
Could you give me an example code of how this would work, and how to 
setup the database, I'm pretty new and while what you have written 
makes sense, and sounds like a good plan I'm not sure I can implement it.



I'm using my own agi-bin for "patching" callerid and adding the name if the
number is found in a table (a csv that is mantained with a spreadsheet), it
adds the name taken from this table. Then you can see the name in the
display of the phones.

--
Alejandro Vargas
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RE: [Asterisk-Users] CallerID

[Technical Support]

Miles,

You can also download cid_rewrite from www.generationd.com  This PHP script
looks up the phone numbers in a local MySQL table, and/or uses reverse 411
on the web to lookup the name, and/or more options.

Michelle 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Vargas
Sent: Friday, April 07, 2006 4:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID

2006/4/7, Miles Scruggs <[EMAIL PROTECTED]>:
> Could you give me an example code of how this would work, and how to 
> setup the database, I'm pretty new and while what you have written 
> makes sense, and sounds like a good plan I'm not sure I can implement it.

I'm using my own agi-bin for "patching" callerid and adding the name if the
number is found in a table (a csv that is mantained with a spreadsheet), it
adds the name taken from this table. Then you can see the name in the
display of the phones.

--
Alejandro Vargas
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RE: [Asterisk-Users] CallerID

[Colin Anderson]

I do the same as well. From my SQL server, I have my Asterisk box query my
customer and B2B contacts using ODBCSock and compose them as as CSV on the
Asterisk box; a script then parses the CSV and DBPut's them into Asterisk
itself. The nice thing about it is you can modify the CallerID with rich
data, for example, when a customer calls, I prepend the customer ID number
for our CRM into the CallerID so the staff member can type in the customer
ID in to the CRM before they pick up. 

I have an .awk that parses the CSV and DBPut's it into Asterisk, if you are
interested email me offlist. 

-Original Message-
From: Alejandro Vargas [mailto:[EMAIL PROTECTED]
Sent: Friday, April 07, 2006 2:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID


2006/4/7, Miles Scruggs <[EMAIL PROTECTED]>:
> Could you give me an example code of how this would work, and how to
> setup the database, I'm pretty new and while what you have written makes
> sense, and sounds like a good plan I'm not sure I can implement it.

I'm using my own agi-bin for "patching" callerid and adding the name
if the number is found in a table (a csv that is mantained with a
spreadsheet), it adds the name taken from this table. Then you can see
the name in the display of the phones.

--
Alejandro Vargas
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Re: [Asterisk-Users] CallerID

[Alejandro Vargas]

2006/4/7, Miles Scruggs <[EMAIL PROTECTED]>:
> Could you give me an example code of how this would work, and how to
> setup the database, I'm pretty new and while what you have written makes
> sense, and sounds like a good plan I'm not sure I can implement it.

I'm using my own agi-bin for "patching" callerid and adding the name
if the number is found in a table (a csv that is mantained with a
spreadsheet), it adds the name taken from this table. Then you can see
the name in the display of the phones.

--
Alejandro Vargas
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Re: [Asterisk-Users] CallerID

[Miles Scruggs]

Could you give me an example code of how this would work, and how to 
setup the database, I'm pretty new and while what you have written makes 
sense, and sounds like a good plan I'm not sure I can implement it.


Thanks

Miles

Waldo Rubinstein wrote:
AFAIK, you can use database lookups to fetch the "internal" caller id 
and "external" caller id depending on the channel that is placing the 
call. Then, simply set the corresponding caller id before placing the 
call. Alternatively, which is what I currently do, since I don't use 
account codes, I set the accountcode parameter in my sip peer 
definitions to the external caller id I want to show, and then I force 
the caller id to the ${CDR(accountcode)} variable before placing 
external calls.


I don't know if there are any other more efficient methods.

- Waldo

On Apr 6, 2006, at 3:02 AM, Miles Scruggs wrote:

how do you set two types of caller id one for internal calling and 
one for external?  Basically everyone calling out from asterisk from 
one context I want to assign a single callerid.  On all other 
contexts I want to assign a caller ID specific to each line for all 
calls going out to asterisk.


Finally for all calls that remain behind the asterisk box (ext to 
ext) the Caller ID is set to the specific extension of the caller.


Thanks

Miles

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Re: [Asterisk-Users] CallerID

[Waldo Rubinstein]

AFAIK, you can use database lookups to fetch the "internal" caller id  
and "external" caller id depending on the channel that is placing the  
call. Then, simply set the corresponding caller id before placing the  
call. Alternatively, which is what I currently do, since I don't use  
account codes, I set the accountcode parameter in my sip peer  
definitions to the external caller id I want to show, and then I  
force the caller id to the ${CDR(accountcode)} variable before  
placing external calls.


I don't know if there are any other more efficient methods.

- Waldo

On Apr 6, 2006, at 3:02 AM, Miles Scruggs wrote:

how do you set two types of caller id one for internal calling and  
one for external?  Basically everyone calling out from asterisk  
from one context I want to assign a single callerid.  On all other  
contexts I want to assign a caller ID specific to each line for all  
calls going out to asterisk.


Finally for all calls that remain behind the asterisk box (ext to  
ext) the Caller ID is set to the specific extension of the caller.


Thanks

Miles

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RE: [Asterisk-Users] callerid= in zapata.conf

[Nabeel Jafferali]

> Agreed, but if all else fails, set a different context for that PRI,
> and in that context, force the CallerID using SetCallerID before
> making the onward call.

Agreed, but I like "clean" configs.
 
> It is possible that the Zaptel "callerid=" field does not accept the
> "Name"  format (I'd have to look at the source to check, and
> you didn't say which veriosn you run). Have you tried just putting a
> number in.

I tried both. I even stopped and started Asterisk - I was not sure if
zapata.conf reloads on a 'reload'.

BTW I'm running 1.2.0.

> PS. While I'm here, is the A102d worth the money? I assume it works as
> easily as the A104d?

Yes, great card. Worked right away using the simple instructions at
http://sangoma.editme.com/. Contact me off-list if you need more info.

Nabeel

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Re: [Asterisk-Users] callerid= in zapata.conf

[Steve Davies]

On 3/21/06, Nabeel Jafferali <[EMAIL PROTECTED]> wrote:
> > try SetCallerId or set callerid=name <(xxx)xxx-> in sip.conf or
> > iax.conf (depending on what you are using)
>
> I am not using SIP or IAX2 clients. As mentioned in the original email, this
> is from PRI to PRI.
>
> I could use SetCallerID, but the zapata.conf method should work.
>

Agreed, but if all else fails, set a different context for that PRI,
and in that context, force the CallerID using SetCallerID before
making the onward call.

It is possible that the Zaptel "callerid=" field does not accept the
"Name"  format (I'd have to look at the source to check, and
you didn't say which veriosn you run). Have you tried just putting a
number in.

PS. While I'm here, is the A102d worth the money? I assume it works as
easily as the A104d?

Thanks,
Steve
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RE: [Asterisk-Users] callerid= in zapata.conf

[Nabeel Jafferali]

> try SetCallerId or set callerid=name <(xxx)xxx-> in sip.conf or
> iax.conf (depending on what you are using)

I am not using SIP or IAX2 clients. As mentioned in the original email, this
is from PRI to PRI.

I could use SetCallerID, but the zapata.conf method should work.

Nabeel

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Re: [Asterisk-Users] callerid= in zapata.conf

[Derek Whitten]

Nabeel Jafferali wrote:
> I have a PRI coming in from a legacy PBX to a Sangoma A102d. The other PRI
> connects to the telco.
> 
> Just before the channel= statements for the PRI connecting to the PBX, I
> have a callerid="Name"<416967>. In extensions.conf, all calls from the
> PBX PRI are sent out the telco PRI. However, the CLID is not set to the
> number I set.
> 
> In fact, the ${CALLERID} variable is empty (which is what the PBX is
> sending, but what I would have expected the callerid= setting to overwrite).
> 
> Any ideas?
> 
> Nabeel
> 
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try SetCallerId or set callerid=name <(xxx)xxx-> in sip.conf or
iax.conf (depending on what you are using)



-- 
.


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RE: [Asterisk-Users] Callerid on transfer

[Alexander Lopez]

This is not posible as a'standard' does not exist for 
rewritingg callerID after once a call is established.  We have C (in 
example given) hang up and than B does a blind transfer.
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Ronald 
  VoermansSent: Monday, March 13, 2006 4:05 AMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: 
  [Asterisk-Users] Callerid on transfer
  
  Hello,
   
  Suppose customer A 
  calls attendant. CallerID of A is displayed at the attendant. But, when 
  attendant does a consulted transfer to, let's say, B, the callerID of 
  attendant is displayed at B. When the consulted transfer is succesful, the 
  callerid of attendant is STILL displayed at B. Is it possible to, after a 
  successful transfer change the callerid of the attendant in the callerid of 
  A?
   
   
  Regard,
   
  Ronald
   
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RE: [Asterisk-Users] Callerid on transfer

[Ronald Voermans]

ok, thank you! 


Regards 

Ronald 

-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens C F
Verzonden: maandag 13 maart 2006 15:27
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [Asterisk-Users] Callerid on transfer

No, at least not yet.

On 3/13/06, Ronald Voermans <[EMAIL PROTECTED]> wrote:
>
> Hello,
>
> Suppose customer A calls attendant. CallerID of A is displayed at the 
> attendant. But, when attendant does a consulted transfer to, let's 
> say, B, the callerID of attendant is displayed at B. When the 
> consulted transfer is succesful, the callerid of attendant is STILL 
> displayed at B. Is it possible to, after a successful transfer change 
> the callerid of the attendant in the callerid of A?
>
>
> Regard,
>
>
> Ronald
>
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>
>
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Re: [Asterisk-Users] Callerid on transfer

[C F]

No, at least not yet.

On 3/13/06, Ronald Voermans <[EMAIL PROTECTED]> wrote:
>
> Hello,
>
> Suppose customer A calls attendant. CallerID of A is displayed at the
> attendant. But, when attendant does a consulted transfer to, let's say, B,
> the callerID of attendant is displayed at B. When the consulted transfer is
> succesful, the callerid of attendant is STILL displayed at B. Is it possible
> to, after a successful transfer change the callerid of the attendant in the
> callerid of A?
>
>
> Regard,
>
>
> Ronald
>
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RE: [Asterisk-Users] CallerID popup

[Mimmus]

> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Iuri Gomes Diniz
> Sent: Friday, February 03, 2006 7:55 PM
> 
> On Fri, 3 Feb 2006 11:41:53 +0100
> Giovanni Miano <[EMAIL PROTECTED]> wrote:
> 
> > Link event
> 
> For me, Link event only occurs when the called number pickup the call.
> 
> I prefer 'Newchannel' event when the 'State' are equal to 'Ringing'

I prefer "Link" (thanks Giovanni!) because I need also to open a CRM app,
thus only when an operator answers.

Mimmus

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Re: [Asterisk-Users] CallerID popup

[Andrew Kohlsmith]

On Monday 06 February 2006 10:54, Facundo Ameal wrote:
> doing) generate a little "database" in XML in which you would put
> jabberid and extension so if you know the extension, you know the
> jabberid... what do you think about that?

Yeah the little thing I whipped up a couple years back has no such provisions.

-A.
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Re: [Asterisk-Users] CallerID popup

[Peter Bowyer]

YAC is a nice popup application (for Windows) to eat alerts just like
the one below.

http://sunflowerhead.com/software/yac/

Peter

On 06/02/06, Facundo Ameal <[EMAIL PROTECTED]> wrote:
> If you wnt to do it quick, I've seen this in another post of this
> list, and I think is good:
>
> exten => s,1,System(/bin/echo -n -e "'${CALLERIDNAME}
> ${CALLERIDNUM}'"| nc -w 1 192.168.1.16 10629)
>
> then you have tyo be monitoring that port and capture the information,
> you can do that in VB.
>
> 2006/2/6, Facundo Ameal <[EMAIL PROTECTED]>:
> > First, about the Jabber library: I'm using Asterisk Perl and the
> > Jabber module for Perl.
> > About dinmically loading the jabberid list, welll that's the problem I
> > had and now I'm developing that. I thought about (and it's what I'm
> > doing) generate a little "database" in XML in which you would put
> > jabberid and extension so if you know the extension, you know the
> > jabberid... what do you think about that?
> >
> > 2006/2/3, Andrew Kohlsmith <[EMAIL PROTECTED]>:
> > > On Friday 03 February 2006 10:21, Facundo Ameal wrote:
> > > > I 'm developing something similar. It a perl script which tells you
> > > > who is calling but it do it by sendind a jabber message.
> > > > it's my first perl script so it's not finished yet.
> > > > i'll share it so you can contribute if you want...
> > >
> > > http://www.mixdown.ca/~andrew/astbot
> > >
> > > -A.
> > > ___
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> > >
> > > Asterisk-Users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> > --
> > Facundo Ameal.
> > famealgmailcom
> > Linux User #395088
> >
> > FWD: 741664
> > MSN: asadolamorcillacomar
> > ICQ: 74005793
> >
> >
> > Open your mind, use open source.
> >
>
>
> --
> Facundo Ameal.
> famealgmailcom
> Linux User #395088
>
> FWD: 741664
> MSN: asadolamorcillacomar
> ICQ: 74005793
>
>
> Open your mind, use open source.
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>


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Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
VoIP: [EMAIL PROTECTED]
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Re: [Asterisk-Users] CallerID popup

[Facundo Ameal]

If you wnt to do it quick, I've seen this in another post of this
list, and I think is good:

exten => s,1,System(/bin/echo -n -e "'${CALLERIDNAME}
${CALLERIDNUM}'"| nc -w 1 192.168.1.16 10629)

then you have tyo be monitoring that port and capture the information,
you can do that in VB.

2006/2/6, Facundo Ameal <[EMAIL PROTECTED]>:
> First, about the Jabber library: I'm using Asterisk Perl and the
> Jabber module for Perl.
> About dinmically loading the jabberid list, welll that's the problem I
> had and now I'm developing that. I thought about (and it's what I'm
> doing) generate a little "database" in XML in which you would put
> jabberid and extension so if you know the extension, you know the
> jabberid... what do you think about that?
>
> 2006/2/3, Andrew Kohlsmith <[EMAIL PROTECTED]>:
> > On Friday 03 February 2006 10:21, Facundo Ameal wrote:
> > > I 'm developing something similar. It a perl script which tells you
> > > who is calling but it do it by sendind a jabber message.
> > > it's my first perl script so it's not finished yet.
> > > i'll share it so you can contribute if you want...
> >
> > http://www.mixdown.ca/~andrew/astbot
> >
> > -A.
> > ___
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> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> --
> Facundo Ameal.
> famealgmailcom
> Linux User #395088
>
> FWD: 741664
> MSN: asadolamorcillacomar
> ICQ: 74005793
>
>
> Open your mind, use open source.
>


--
Facundo Ameal.
famealgmailcom
Linux User #395088

FWD: 741664
MSN: asadolamorcillacomar
ICQ: 74005793


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Re: [Asterisk-Users] CallerID popup

[Facundo Ameal]

First, about the Jabber library: I'm using Asterisk Perl and the
Jabber module for Perl.
About dinmically loading the jabberid list, welll that's the problem I
had and now I'm developing that. I thought about (and it's what I'm
doing) generate a little "database" in XML in which you would put
jabberid and extension so if you know the extension, you know the
jabberid... what do you think about that?

2006/2/3, Andrew Kohlsmith <[EMAIL PROTECTED]>:
> On Friday 03 February 2006 10:21, Facundo Ameal wrote:
> > I 'm developing something similar. It a perl script which tells you
> > who is calling but it do it by sendind a jabber message.
> > it's my first perl script so it's not finished yet.
> > i'll share it so you can contribute if you want...
>
> http://www.mixdown.ca/~andrew/astbot
>
> -A.
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--
Facundo Ameal.
famealgmailcom
Linux User #395088

FWD: 741664
MSN: asadolamorcillacomar
ICQ: 74005793


Open your mind, use open source.
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Re: [Asterisk-Users] CallerID popup

[Iuri Gomes Diniz]

On Fri, 3 Feb 2006 11:41:53 +0100
Giovanni Miano <[EMAIL PROTECTED]> wrote:

> Link event

For me, Link event only occurs when the called number pickup the call.

I prefer 'Newchannel' event when the 'State' are equal to 'Ringing'

-- 
Iuri Gomes Diniz 
Network Admin and Programmer [http://clx.digi.com.br]
DIGINET [http://www.digi.com.br]
Natal - RN - Brazil.



-- 
Iuri Gomes Diniz 
Network Admin and Programmer [http://clx.digi.com.br]
DIGINET [http://www.digi.com.br]
Natal - RN - Brazil.


-- 
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Re: [Asterisk-Users] CallerID popup

[Andrew Kohlsmith]

On Friday 03 February 2006 10:21, Facundo Ameal wrote:
> I 'm developing something similar. It a perl script which tells you
> who is calling but it do it by sendind a jabber message.
> it's my first perl script so it's not finished yet.
> i'll share it so you can contribute if you want...

http://www.mixdown.ca/~andrew/astbot

-A.
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Re: [Asterisk-Users] CallerID popup

[Facundo Ameal]

I 'm developing something similar. It a perl script which tells you
who is calling but it do it by sendind a jabber message.
it's my first perl script so it's not finished yet.
i'll share it so you can contribute if you want...

2006/2/3, C F <[EMAIL PROTECTED]>:
> You should write a proxy and not connect directly, the reasons are as follows:
> 1. You don't want asterisk to crash because of problems with the
> manager app over the network, which Asterisk is known not to handle
> very well (as per the wiki).
> 2. Security, if you have every computer connecting to asterisk manager
> over the network, then you are giving the users a way to login to the
> system to do much more than they need, with a proxy however, you can
> always validate (and you should make sure to do that) everything
> before its submitted to asterisk.
>
>
> On 2/3/06, Mimmus <[EMAIL PROTECTED]> wrote:
> >
> > It works. Thanks a lot.
> > With 15/20 users, is it better to use a manager proxy or to connect directly
> > to the Asterisk server?
> >
> > Thanks
> >
> >
> >  
> >  From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf
> > Of Giovanni Miano
> > Sent: Friday, February 03, 2006 11:42 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] CallerID popup
> >
> >
> > Link event
> >
> >
> > ___
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> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
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--
Facundo Ameal.
famealgmailcom
Linux User #395088

FWD: 741664
MSN: asadolamorcillacomar
ICQ: 74005793


Open your mind, use open source.
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Re: [Asterisk-Users] CallerID popup

[C F]

You should write a proxy and not connect directly, the reasons are as follows:
1. You don't want asterisk to crash because of problems with the
manager app over the network, which Asterisk is known not to handle
very well (as per the wiki).
2. Security, if you have every computer connecting to asterisk manager
over the network, then you are giving the users a way to login to the
system to do much more than they need, with a proxy however, you can
always validate (and you should make sure to do that) everything
before its submitted to asterisk.


On 2/3/06, Mimmus <[EMAIL PROTECTED]> wrote:
>
> It works. Thanks a lot.
> With 15/20 users, is it better to use a manager proxy or to connect directly
> to the Asterisk server?
>
> Thanks
>
>
>  
>  From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf
> Of Giovanni Miano
> Sent: Friday, February 03, 2006 11:42 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] CallerID popup
>
>
> Link event
>
>
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>
>
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RE: [Asterisk-Users] CallerID popup

[Mimmus]

I would be happy to share everything but actually I'm working only at a
feasibility study. 
In addition, I'm a system admin and development job is made by someone else!

In principle, it's simple: open a socket to manager port, login and wait for
"right" event.
Ideal target is a small, traybar application giving chance to
login/logoff/pause to the agent (in my opinion, it's better than doing it by
phone) and pointing the CRM application to the right caller card.

In addition, peraphs, if number of agents is more than a few, it's better to
use a manager proxy.


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jonathan k. Creasy
> Sent: Friday, February 03, 2006 2:13 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] CallerID popup
> 
> I have been planning to do the same thing but never got 
> around to it, I actually did write a nice class to wrap the 
> interface to the manager but it isn't complete. 
> 
> Would you be willing to share your work?
> -Jonathan
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mimmus
> Sent: Friday, February 03, 2006 5:19 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] CallerID popup
> 
> Hi,
> I'm trying to write a small Visual Basic app to throw a popup 
> with CallerIDNum when a call center agent answers a queue call.
> Does anyone know what is the right manager event to intercept?
> 
> Thanks
> Mimmus
> 

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RE: [Asterisk-Users] CallerID popup

[Jonathan k. Creasy]

I have been planning to do the same thing but never got around to it, I
actually did write a nice class to wrap the interface to the manager but
it isn't complete. 

Would you be willing to share your work?
-Jonathan



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mimmus
Sent: Friday, February 03, 2006 5:19 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] CallerID popup

Hi,
I'm trying to write a small Visual Basic app to throw a popup with
CallerIDNum when a call center agent answers a queue call.
Does anyone know what is the right manager event to intercept?

Thanks
Mimmus

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Re: [Asterisk-Users] CallerID popup

[asterisk183]

Dial event, in Asterisk 1.2:203->201Event: Dial  Privilege: call,all  Source: SIP/203-8467  Destination: SIP/201-45d9  CallerID: 203  CallerIDName: 203  SrcUniqueID: asterisk-1912-1138197095.3769  DestUniqueID: asterisk-1912-1138197095.3771Mimmus <[EMAIL PROTECTED]> ha scritto:   Hi,I'm trying to write a small Visual Basic app to throw a popup withCallerIDNum when a call center agent answers a queue call.Does anyone know what is the right manager event to intercept?ThanksMimmus___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] CallerID popup

[Mimmus]

It works. Thanks a lot.
With 15/20 users, is it better to use a manager proxy or to 
connect directly to the Asterisk server?
 
Thanks

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Giovanni 
  MianoSent: Friday, February 03, 2006 11:42 AMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] CallerID popup
  Link event
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Re: [Asterisk-Users] CallerID popup

[Giovanni Miano]

Link event2006/2/3, Mimmus <[EMAIL PROTECTED]>:
Hi,I'm trying to write a small Visual Basic app to throw a popup withCallerIDNum when a call center agent answers a queue call.Does anyone know what is the right manager event to intercept?ThanksMimmus
___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano
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RE: [Asterisk-Users] Callerid Name

[Alexander Lopez]

Look at
 http://bugs.digium.com/view.php?id=1192

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> John Bittner
> Sent: Thursday, February 02, 2006 11:27 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Callerid Name
> 
> Anyone know why zaptel would ignore a facility message from 
> an ISDN PRI.
> I am trying to get Callerid name to work. The carrier says it 
> on and I see it in the pri debug but asterisk never gets it.
> 
> Any help would be appreciated.
> 
> Thanks
> 
> John Bittner
> Simlab.net
> 
> 
> 
>  Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 572/0x23C) (Terminator) Message type: 
> > ALERTING (1) [1e 02 81 88]I> Progress Indicator (len= 4) [ Ext: 1  
> > Coding: CCITT (ITU) standard (0) 0:
> 0   Location: Private network serving the local user (1)
> >   Ext: 1  Progress Description: Inband
> information or appropriate pattern now available. (8) ]
> -- SIP/69.60.198.130-5119 is ringing < Protocol 
> Discriminator: Q.931 (8)  len=36 < Call Ref: len= 2 
> (reference 572/0x23C) (Originator) < Message type: FACILITY 
> (98) < [1c 1d 9f 8b 01 00 a1 17 02 01 01 02 01 00 80 0f 42 55 
> 4c 4b 41 4e 27 53 2c 48 45 41 4c 54 48] < Facility (len=31, 
> codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x17, 0x02, 0x01, 
> 0x01, 0x02, 0x01, 0x00, 0x80, 0x0f, 'BULKAN', 0x27, 'S', 
> 0x2c, 'HEALTH' ]
> -- Processing IE 28 (cs0, Facility)
> Handle Q.932 ROSE Invoke component
> 
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Re: [Asterisk-Users] CallerID Problem

[Gary Richardson]

Num and Number are aliases, I believe.

I just tried it and I got the same error..

Thanks.

On 2/1/06, Jason Adams <[EMAIL PROTECTED]> wrote:
> Have you tried this:
>
> exten => _9.,1,Set(CALLERID(num)=)
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gary
> Richardson
> Sent: Wednesday, February 01, 2006 12:47 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] CallerID Problem
>
> Hey,
>
> I'm using a Cisco 2811 to make calls out to a PRI. My asterisk box
> connects to it using SIP. The asterisk version is 1.2.0.
>
> In my sip.conf, I set callerid="First Last" 
>
> When I make a an outbound call with the following macro:
>
> exten => _9.,1,Dial(SIP/${EXTEN}@,,w)
> exten => _9.,2,Congestion()
>
> The caller id is set to the extension that's defined in sip.conf.
>
> If I try something like:
>
> exten => _9.,1,Set(CALLERID(number)=)
> exten => _9.,2,Dial(SIP/${EXTEN}@,,w)
> exten => _9.,3,Congestion()
>
> I get the following error:
>
> -- Got SIP response 488 "Not Acceptable Media" back from 
>
> It all works fine if I don't set the caller id.. Any ideas on why this
> may be happening?
>
> Thanks.
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RE: [Asterisk-Users] CallerID Problem

[Jason Adams]

Have you tried this:

exten => _9.,1,Set(CALLERID(num)=)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
Richardson
Sent: Wednesday, February 01, 2006 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] CallerID Problem

Hey,

I'm using a Cisco 2811 to make calls out to a PRI. My asterisk box
connects to it using SIP. The asterisk version is 1.2.0.

In my sip.conf, I set callerid="First Last" 

When I make a an outbound call with the following macro:

exten => _9.,1,Dial(SIP/${EXTEN}@,,w)
exten => _9.,2,Congestion()

The caller id is set to the extension that's defined in sip.conf.

If I try something like:

exten => _9.,1,Set(CALLERID(number)=)
exten => _9.,2,Dial(SIP/${EXTEN}@,,w)
exten => _9.,3,Congestion()

I get the following error:

-- Got SIP response 488 "Not Acceptable Media" back from 

It all works fine if I don't set the caller id.. Any ideas on why this
may be happening?

Thanks.
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Re: [Asterisk-Users] CallerID info needed

[Doug Lytle]

C F wrote:


Look at this post:
http://lists.digium.com/pipermail/asterisk-users/2005-December/139952.html


 

Actually, I don't think the caller ID number is being sent in my 
situation, I am wondering what I can't manually set it.


Thanks for the reply!

Doug

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Re: [Asterisk-Users] CallerID info needed

[C F]

Look at this post:
http://lists.digium.com/pipermail/asterisk-users/2005-December/139952.html


On 12/28/05, Doug Lytle <[EMAIL PROTECTED]> wrote:
> Hey everybody,
>
> I'm trying to figure out a problem with Caller-ID info coming in from
> one of our facilities.  The Caller-ID name is all that comes across.  I
> figured out that I probably could do a database lookup against the name
> and set the Caller-ID number to their extension.  I'm using Asterisk
> SVN-trunk-r7230 on a PRI connected to a Definity PBX.
>
> When testing, my Polycom IP501 still shows unknown.  A bit from the log
> below:
>
> -- Accepting call from '' to '4288' on channel 0/2, span 1
> -- Executing DBget("Zap/2-1", "CIDINFO=name/Ballard, Lance") in new
> stack
> -- DBget: varname=CIDINFO, family=name, key=Ballard, Lance
> -- DBget: set variable CIDINFO to 4300
> -- Executing NoOp("Zap/2-1", "Setting CallerID Number to: 4300") in
> new stack
> -- Executing Set("Zap/2-1", "CALLERID(Name)=Ballard, Lance") in new
> stack
> -- Executing Set("Zap/2-1", "CALLERID(Number)=4300") in new stack
> -- Executing SetGroup("Zap/2-1", "Max_Calls") in new stack
> -- Executing NoOp("Zap/2-1", "Active Calls: 1") in new stack
> -- Executing GotoIf("Zap/2-1", "0?103") in new stack
> -- Executing Dial("Zap/2-1",
> "IAX2/bc.asterisk:[EMAIL PROTECTED]/4288||t") in new stack
> -- Called bc.asterisk:[EMAIL PROTECTED]/4288
> -- Call accepted by 192.168.102.15 (format gsm)
> -- Format for call is gsm
> -- IAX2/liv.asterisk-2 is ringing
>
> And my code snip:
>
> exten => _42XX,1,Dbget(CIDINFO=name/${CALLERIDNAME})
> exten => _42XX,2,NoOp(Setting CallerID Number to: ${CIDINFO})
> exten => _42XX,3,Set(CALLERID(Name)=${CALLERIDNAME})
> exten => _42XX,4,Set(CALLERID(Number)=${CALLERIDNUM}${CIDINFO})
>   ; Both variables with never have data at the same
> time
> exten => _42XX,5,SetGroup(Max_Calls)
> exten => _42XX,6,NoOP(Active Calls: ${GROUP_COUNT(Max_Calls)})
> exten => _42XX,7,GotoIf($[ ${GROUP_COUNT(Max_Calls)} > 4 ]?103)
> exten =>
> _42XX,8,Dial(IAX2/bc.asterisk:[EMAIL PROTECTED]/${EXTEN},,t)
> exten => _42XX,103,Congestion()
>
> Any suggestions on a fix (if it's possible)?
>
> Thanks!
>
> Doug
>
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Re: [Asterisk-Users] Callerid

[Abdul Lateef]

Hi,

I am using SIPS softphoe. and i tested with another
SIP Gatekeeper and i can see callerid in plain format.
But when i am trying using Asterisk it is apearing
"callerid", .

So i don't think this is from client side or
softphone.




Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
Tel: +974 - 4883068
ICQ: 276994704
YM!: abdul_zu
Fax: +974 - 4883063
Doha Qatar
http://www.hatif.com



__ 
Yahoo! DSL – Something to write home about. 
Just $16.99/mo. or less. 
dsl.yahoo.com 

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Re: [Asterisk-Users] Callerid

[Mark Phillips]

Assuming its a SIP based device

[110001]
user=something
allow=whatever
callerid= lateef




Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com


Code Lover wrote:

Hi all,

How i can change the CallerId format in plan id?
for the example i can see the value of CALLERID variable like

"lateef" <110001>

I want to let asterisk do in plain id like

lateef


any idea?

--
Thank You,
Code Lover
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Re: [Asterisk-Users] CallerID/Extension Matching with Realtime Extensions

[Kevin P. Fleming]

Douglas Garstang wrote:

which matches when a user with callerid 5551212 dials 8000. 
This doesn't work with realtime extensions. or does it? Does someone know how it's done?

The following doesn't work. Asterisk can't find the number.


The docs for Realtime extensions clearly state that CallerID matching is 
not supported.

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Re: [Asterisk-Users] CallerID/Extension Matching with RealtimeExtensions

[Brian Capouch]

Douglas Garstang wrote:

Well, according to something I just read, dated JULY, this isn't supported yet. 
This is so damn frustrating. Why does Digium keep releasing this stuff before 
it's ready? Why can't they friggin document this? Why is it every time I check 
out a new feature I have to drop it because of some deal breaker like this?



Hmm.  Your post would indicate that you are under the impression that 
Asterisk is a commercial product that was released by Digium.


It is actually an Open Source project, under development by lots and 
lots of people located all over the world.


So Digium didn't release the code; the Asterisk developers released it.

The wonderful thing about Open Source--the door swinging both ways--is 
that those who GRIPE about features that are lacking can actually get 
the source code and FIX the thing that is causing their gripe.


In your case, it might provide you an opportunity to come back to the 
list someday with something positive to say!!


B.

p.s. Honey will get you a lot further than vinegar.  'Specially if 
you're just a lowly enduser (like me), not a coder. . .

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RE: [Asterisk-Users] CallerID/Extension Matching with RealtimeExtensions

[Douglas Garstang]

Well, according to something I just read, dated JULY, this isn't supported yet. 
This is so damn frustrating. Why does Digium keep releasing this stuff before 
it's ready? Why can't they friggin document this? Why is it every time I check 
out a new feature I have to drop it because of some deal breaker like this?

-Original Message- 
From: Douglas Garstang 
Sent: Thu 12/15/2005 10:50 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion; 
asterisk-users@lists.digium.com 
Cc: 
Subject: [Asterisk-Users] CallerID/Extension Matching with 
RealtimeExtensions



In a regular extensions.conf, you can specify an extension as 
callerid/extension. For example

5551212/8000 => dosomething.

which matches when a user with callerid 5551212 dials 8000.
This doesn't work with realtime extensions. or does it? Does 
someone know how it's done?
The following doesn't work. Asterisk can't find the number.

mysql> select * from extensions_table order by context, exten, 
priority;   

++--+---+--+--+--+
| id | context  | exten | priority | app  | appdata 
 |

++--+---+--+--+--+
| 14 | context2 | 1001/1401 |1 | Answer   | 
 |
| 15 | context2 | 1001/1401 |2 | Playback | 
hang-on-a-second-angry   |
| 16 | context2 | 1001/1401 |3 | Hangup   | 
 |

Doug.



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RE: [Asterisk-Users] CallerID Transfer

[Rushowr]

Use the o flag to force the original callerid, not the num2 
callerid.
 
example:
 
exten = s,1,Dial(SIP/200,30,ortT)
 
SKM


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
asterisk183Sent: Monday, December 12, 2005 2:59 AMTo: 
asteriskSubject: [Asterisk-Users] CallerID 
Transfer

 When I receveid a call (num1) in the my office (num2), 
I transfer the call at the num3, but the callerid is num2, in the 
telephone3.What can I doing for show the callerid 
num1?Thanks


Yahoo! 
Mail: gratis 1GB per i messaggi, antispam, antivirus, 
POP3
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Re: [Asterisk-Users] callerid international-format

[Florian Overkamp]

Florian Meister wrote:

Hi,

Is it possible to send international format (+435572999888) with asterisk. I 
have the following problem:


When I set the calleridnum to the format above, the telephone (grandstream ata 
with a siemens gigaset) does not display the "+". So I send it now with "00" 
instead of the "+" for the international prefix, but it would be nice if it 
would possible to make the "+"-thing work.


You could try messing with the type of callerid the ATA is sending. In 
DTMF you cannot send a '+' symbol, but maybe in Bellcore it can work ?


(For the record: I doubt that this is possible, but feel free to try)

Florian
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Re: [Asterisk-Users] CallerID not passing through to Polycom 500 (SOLVED, sort of)

[Adam Goryachev]

On Fri, 2005-11-25 at 13:08 -0500, Gary MacKay wrote:
> After playing around with the CALLERID(number) and 
> CALLERID(name) variables and things, I find that asterisk is sending
> the "name" to my phone and the name is "unknown". I added a line
> exten => _X.,Set(CALLERID(name)=${CALLERIDNUM})  and now it shows
> the number. Is this the right way to do this?

Nope, the correct method is probably to upgrade the firmware in your
polycom phone. I recall one of the bugs fixed in 1.5.2 was to display
CID name + number instead of name only...

Regards,
Adam


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Re: [Asterisk-Users] CallerID

[JP Carballo]

Eric "ManxPower" Wieling wrote:



My point is that CALLERID(number) is ALWAYS the same as ${CALLERIDNUM} 
so setting one to the other is pointless.  It's like setting 2=2.  
Same with the CallerIDName stuff.

___


Point taken.
Well, between our posts, (and a few minutes testing), I believe there 
was enough info to figure out what was needed.


--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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Re: [Asterisk-Users] CallerID not passing through to Polycom 500

[Mojo with Horan & Company, LLC]

If this is the problem you're having, you would simply replace something 
like:


exten => s,1,Dial...

with

exten => s,1,Wait(1)
exten => s,2,Dial...

Gary MacKay wrote:

How do I make it wait? For how long? I watched the logs but did not see 
anything that related to this.



Check your logs, make sure you are waiting long enough before sending
the call to the polycom.

Uf asterisk sees the CID, it should send it and it should show up on the
polycom.

Greg
-Original Message-
From: asterisk-users-bounces at lists.digium.com 

[mailto:asterisk-users-bounces at lists.digium.com 
] On Behalf Of Gary
MacKay
Sent: Thursday, November 24, 2005 11:19 AM
To: asterisk-users at lists.digium.com 

Subject: [Asterisk-Users] CallerID not passing through to Polycom 500

I have a basic system working, except for callerid. The Polycom 500 just
shows call from "Business Line" on the screen. "Business Line" is the
name of the context that line is in. How do I get it to show the
callerID on the screen instead? Yes, I have CallerID on that line and it
works on a standard analog phone.




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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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RE: [Asterisk-Users] CallerID not passing through to Polycom 500

[gw]

 
Check your logs, make sure you are waiting long enough before sending
the call to the polycom.

Uf asterisk sees the CID, it should send it and it should show up on the
polycom.

Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
MacKay
Sent: Thursday, November 24, 2005 11:19 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] CallerID not passing through to Polycom 500

I have a basic system working, except for callerid. The Polycom 500 just
shows call from "Business Line" on the screen. "Business Line" is the
name of the context that line is in. How do I get it to show the
callerID on the screen instead? Yes, I have CallerID on that line and it
works on a standard analog phone.
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