"CallerID Management" outbound CID using AMI

Tags: #<Tag:0x00007f24c4579ac8>

(Mo The Mighty) #1

Hello all, apologies for the long message –

I’m using FreePBX, Asterisk 16.15.1, and I have a question relating to using AMI and transferring calls.

For discussion purposes, I’m transferring from the extension ‘5555’. Our SIP provider requires you to present a number that is on the account, let’s say ‘12223334444’.

We are utilizing the “CallerID Management” module to set a prefix and associate it with a particular outbound CID. With that in mind, let’s say the prefix ‘*2100’ is associated with the number ‘12223334444’. For example, if I’m trying to transfer a call to ‘13334445555’ and I’m using a softphone, such as Zoiper, I can dial ‘*2’ or ‘##’, I get the audio prompt ‘Transfer’ and I can then dial the prefix and the number I’m trying to reach (*210013334445555). When I do this, it works great for both blind and supervised transfers. The call connects and I can see the correct outbound CID ‘12223334444’ on the trunk. However, when using Javascript and AMI I’m not having the same luck. I can use ‘dial=*210013334445555’ and it works great, but when using ‘transfer=*210013334445555’ or ‘supervisedtransfer=*210013334445555’ I keep getting an “All Circuits Are Busy Now”. So, I jump on the SIP provider portal and the number coming down the trunk is not the number associated with the prefix, which in this instance is ‘12223334444’. Instead I’m seeing +445555.

I have spent all day playing with the different dial plan configurations, but I’m still coming up empty handed. Is there a way that I can still get that outbound CID that is associated with the prefix to present down the trunk when using JS and AMI the same way as when I manually dial the transfer with a softphone or a WebRTC phone.

Any ideas are very much appreciated.

(David55) #2

Is this one of the providers that misuses Remote-Party-ID as the account name? If not, the account information is taken from the From: header, which can be forced to a particular value in sip.conf or pjsip.conf. Given that is so common, I assume that FreePBX supports it.