Can anyone advise whether theres some method to reliably determine which SIP device is the party performing a transfer (REFER or using features.conf), and to whom theyre transferring?Weve been analysing the AMI AttendedTransfer events and of course..
Author : David Cunningham
Home » Articles posted by David Cunningham (paged 3)
We have an issue where using ChanSpy with the B option to talk to both parties doesnt work in some call scenarios (calling out to a T-mobile device). The problem is that the spying party can talk to the called person but not the caller. The Aster..
Im familiar with Pickup/PickupChan for taking a ringing call, but does anyone know how a phone can steal an already answered call from another phone? Our users have decided that call parking is too long-winded and dont want to use that.For example: ph..
Does anyone know if its possible to get the non-terminating DTMF keys that may have been pressed while using the Record command in Asterisk?The intended purpose is that a caller can say a name or enter it using DTMFkeys. If we can get the keys in a varia..
We are generating AOC messages via the AMI and trying to deliver them to various brands of SIP phone.When snom_aoc_enabled = yes in sip.conf then a message is set in the Snom format correctly.Were not having much luck sending any other type of AOC mess..
Does anyone know how to send a SIP INFO with custom data on a specified channel from Asterisk? The intention is to provide an Advice Of Charge.Asterisk has its own AOC function but it only seems to support Snoms format, not the more general XML format.Search..
Is there any equivalent of the CONNECTEDLINE function which can be called from an application using the AMI?Thanks for ..
We are seeing an issue where a call will be stuck for roughly a minute or so and in the Asterisk log give many Expected to acknowledge tickswarnings as below. Can anyone suggest a cause? The system is running11.21.2. Thanks in advance.[Mar 10 08:00:..
Is it possible to mix PJSIP realtime and flat file configuration in pjsip,conf?What we want is to set up endpoints in the ps_endpoints table with some columns set but most being NULL, and then allow end-customers to optionally add configuration by add..
Can anyone advise on the status of SRV lookups in Asterisk 11?(specifically 11.17.1)Is there any difference given how the Dial is done, and how supported are weights and priorities?Thanks i..