[asterisk-bugs] [Asterisk 0010451]: Sending caller ID to ZAP Extension fails if sendcalleridafter=0 or sendcalleridafter=1

noreply at bugs.digium.com noreply at bugs.digium.com
Sun Aug 19 20:30:53 CDT 2007


The following issue has been UPDATED. 
====================================================================== 
http://bugs.digium.com/view.php?id=10451 
====================================================================== 
Reported By:                rjenkins
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   10451
Category:                   Channels/chan_zap
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.10.1  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             08-14-2007 15:20 CDT
Last Modified:              08-19-2007 20:30 CDT
====================================================================== 
Summary:                    Sending caller ID to ZAP Extension fails if
sendcalleridafter=0 or sendcalleridafter=1
Description: 
UK Caller ID setup is (or shoudl be):
cidsignalling=v23
cidstart=polarity
sendcalleridafter=0

This setup always fails to properly send the caller ID; logged error is:
WARNING[14416] chan_zap.c: Didn't finish Caller-ID spill. Cancelling.

(It still fails with sendcalleridafter=1, it must be 2 to avoid an
error.)

Ideally, with this setup the first actual ring should be deferred until a
short time after the cidspill has been completed (or the cidspill data
should include a 200mS silence at the end, allowing ring to follow
immediately).

The UK (British Telecom) standard timing sequence is:
Polarity reversal,
300mS delay (which may include other signalling),
V23 caller ID data burst, starting with 45 - 75mS MARK tone,
Minimum 200mS silence from end of data burst to start if ringing voltage.

[The idea is that caller ID can be received and processed before ring
voltage arrives, so unwanted calls can be muted without any ringing]

The formal timing diagram is on page 8 of the BT document here:
http://www1.btwebworld.com/sinet/242v2p3.pdf

====================================================================== 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
08-19-07 20:30  file           Asterisk Version         1.4.10  => 1.4.10.1 
08-19-07 20:30  file           Category                 Addons/General =>
Channels/chan_zap
======================================================================




More information about the asterisk-bugs mailing list