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

[Paul Hales]

James - I have the same problem, and tried a some of the same ideas. No
result.

But at least we both know that a few people in Australia are using Asterisk!

Later,

PaulH
 

-----Original Message-----
From: James Bean [mailto:james at hdcs.com.au] 
Sent: Wednesday, 13 October 2004 4:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P

>> 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.
>
>I'm not sure, but try the following:
>
>a) Ensure you actually have the callerid service provided to your line,
this is usually an extra charge from telstra (AFAIK)

Yep my analog handset on the line (not through asterisk) displays the
callerid of the incoming call (just as a double check).

>b) Add a wait(3) (maybe a 2 or 4 seconds is needed) before the noop

Took it out to Wait(5), and made sure that the callerid was being displayed
on my analog handset before the wait times out in asterisk to do the noop.
Still no go.

SIP handset still displays Asterisk on it when the call is patched through.

>c) Patch asterisk with this patch (I'm still waiting to be able to do
this from a config file. This is what I use to allow asterisk to pass
callerid *to* my analog FXS extensions. I assume it is the same for FXO
lines.
>
>diff -ur asterisk/channels/chan_zap.c asterisk.mine/channels/chan_zap.c
>--- asterisk/channels/chan_zap.c        Wed Sep 22 18:24:18 2004
>+++ asterisk.mine/channels/chan_zap.c   Wed Sep 22 18:24:41 2004
>@@ -89,7 +89,7 @@
> /* #define ZAP_CHECK_HOOKSTATE */
>
> /* Typically, how many rings before we should send Caller*ID */
-#define DEFAULT_CIDRINGS 1
>+#define DEFAULT_CIDRINGS 2
>
> #define CHANNEL_PSEUDO -12
>
>Obviously after the last one, you need to re-compile and re-install
asterisk, and then re-start asterisk.
>
>Regards,
>Adam

Yes I had found this patch previous and it was already compiled into my
current build, asterisk 1.0.1...

Thanks for the reply though it did open my eyes to a few things.

Unfortunately no callerid from the incoming analog line call on my TDM400P.

James
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
CAUTION: This email message and accompanying data may contain information
that is confidential. If you are not the intended recipient, you are
notified that any use, dissemination, distribution or copying of this
message or data is prohibited. If you have received this email message in
error, please notify us immediately and erase all copies of this message and
attachments. Thank you.

CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you.