Steve Davies
2013-Jul-29 15:52 UTC
[asterisk-users] Connected Line presentation in 1.8.x upwards
Hi, I've searched the asterisk.org and voip-info wiki sites, but not found an answer that seems to match. Hopefully this is a simple question. COLP is working very well on our system - Unfortunately it is working a bit TOO well in some circumstances. We have some "untrusted" trunks. On these trunks, an initial CallerID can be used, but any redirected caller numbers, COLP updates etc are not safe to accept. Sadly I cannot find how to cause COLP updates to be ignored for a trunk. I need solutions for SIP, IAX and DAHDI, what options do I have? This applies to both in- and out-bound calls. Are there some variables that I can set just before dialling an outbound call, and immediately on receiving an inbound call to determine what the callerID values will be for the entire duration of the call? (ie. old-style pre-COLP behaviour for specific trunks) Thanks for any pointers. Regards, Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130729/bbae0e5a/attachment.htm>
Kevin Larsen
2013-Jul-29 15:55 UTC
[asterisk-users] Connected Line presentation in 1.8.x upwards
From: Steve Davies <davies147 at gmail.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>, Date: 07/29/2013 10:53 AM Subject: [asterisk-users] Connected Line presentation in 1.8.x upwards Sent by: asterisk-users-bounces at lists.digium.com Hi, I've searched the asterisk.org and voip-info wiki sites, but not found an answer that seems to match. Hopefully this is a simple question. COLP is working very well on our system - Unfortunately it is working a bit TOO well in some circumstances. We have some "untrusted" trunks. On these trunks, an initial CallerID can be used, but any redirected caller numbers, COLP updates etc are not safe to accept. Sadly I cannot find how to cause COLP updates to be ignored for a trunk. I need solutions for SIP, IAX and DAHDI, what options do I have? This applies to both in- and out-bound calls. Are there some variables that I can set just before dialling an outbound call, and immediately on receiving an inbound call to determine what the callerID values will be for the entire duration of the call? (ie. old-style pre-COLP behaviour for specific trunks) Thanks for any pointers. Regards, Steve I believe what you are looking for in Dial is the 'I' option. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130729/9ce50cfe/attachment.htm>