[Asterisk-Users] Passing CallerID to SIP phone from TDM400P

[James Bean]

Sorry, I explained this wrong.

I am wanting the callerid of the incoming caller from my analogue line on the TDM400P to be passed TO the sip phone so the sip phone display shows the phone number of the incoming caler from the call on the TDM400P.

It shows any callerid information from other sip phones or extension calls fine.

James 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Emilio Panighetti
Sent: Wednesday, 13 October 2004 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P

If the extension is a SIP Phone, it's up to the SIP Phone to pass the CallerID information. Some ATAs allow you to configure how's the Caller_ID being transmitted (like Cisco ATA-186). Others don't.

if you call from the console, the Caller ID information will say 'asterisk'. from your phones, it won't.

If the call originates, for example, from a SIP endpoint (phone, etc). 
it uses the callerid defined on sip.conf.

In your example, take the double quotes off (that seems to work in my
case):

> [bt-karen]
>  type=friend
>  secret=<password removed>
>  host=dynamic
>  callerid=Karen <691>
>  defaultip=192.168.69.251
> dtmfmode=inband
> mailbox=691

That would be what I would do.

On Oct 13, 2004, at 12:38 AM, James Bean wrote:

>
>
> Hi,
>
> Sorry, newbie, I want to pass the incoming callerid information 
> through to my sip phone but when an incoming call gets passed through 
> it says asterisk on the display instead of the number.
>
> Being in australia callerid information is passed through on the 
> second ring not the first, (hence my noop command doesn't currently
> work)
>
> James
>
> ----------------------------------------------------------
>
> /etc/asterisk/extensions.conf
>
> [pstn]
>
> exten => s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a 
> comment in the CLI for info.
>  exten => s,2,Dial(SIP/snom-james,45,t)  exten => s,3,Hangup  ;exten 
> => s,3,VoiceMail(u100)    ;Whatever box you want.
>
> [internal]
>
> exten => i,1,Playback(invalid)
> exten => i,2,Hangup
>  exten => t,1,Hangup
>
> exten => 099,1,Echo     ;simple echo test when you dial 099 on your 
> phone
>
> include => sip
>
> [sip]
>
> exten => 690,1,Dial(SIP/snom-james,30,tr)  exten => 
> 690,2,voicemail2,u900 exten => 690,102,voicemail2,b900
>
> exten => 691,1,Dial(SIP/bt-karen,30,tr)  exten => 
> 691,2,voicemail2,u901 exten => 691,102,voicemail,b901
>                                      
>
>  /etc/asterisk/sip.conf
>
> [general]
>
> port = 5060
>  bindaddr = 192.168.69.1
>  context = sip
>  disallow = gsm
> allow = alaw
>  disallow = ulaw
> srvlookup=no
>
> [snom-james]
>  type=friend
>  secret=<password removed>
>  host=dynamic
>  callerid="James" <690>
>  defaultip=192.168.69.250
> dtmfmode=inband
> mailbox=690
>
> [bt-karen]
>  type=friend
>  secret=<password removed>
>  host=dynamic
>  callerid="Karen" <691>
>  defaultip=192.168.69.251
> dtmfmode=inband
> mailbox=691
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