External Transfert Not Working After Asterisk Update

Hello,

I migrated Asterisk from version 20, I believe, to version 22. I realized that external transfers no longer work. I configure it directly on my SIP phone.

If I dial directly from the phone, the call works. So my outgoing route is fine?

However, as soon as I activate the transfer, I immediately get the message: the number is not in service.

According to my initial research, this is due to an incorrect from-pstn context. Freepbx considers it to be an incoming call and blocks the call as spam?

Log attached in pastebin: Log freepbx - Pastes.io

Trunk-Diakelys is my operator, TRUNK-DEVILLE is my outgoing route, 2006 is the SIP extension where I configured the transfer, and 2007 is the extension from which I call 2006.

Can you help me?

Thank you and have a great day.

Best regards,

Nicolas

Hello,

UP, can you help me ?

The redirect (this is a SIP native, pre-Answer, blind transfer, using chan_pjsip, on unknown versions of Asterisk and FreePBX) appears to be correctly using the from-internal context. Is 0679963729 a valid number to call from an extension? Did you want the number handled by Asterisk, or did you want a URI to be used directly?

Hello, thank you so much for your help. I’m with freepbx 17, asterisk 22.

The number 0679963729 is my personal number, and it is valid because when I call it directly from the same phone used for the call forwarding, the call goes through. I don’t know which method is better, Asterisk or URI. I just want the transfer to work. Isn’t it normal to have a “from-pstn” context for call forwarding? Because FreePBX seems to think that the number 0679963729 is an incoming DID, so it says it’s not in service?

P.S. Same issue with a native installation of FreePBX 17 and Asterisk 22.

Have a good day

Now forwarding PJSIP/2007-0000004d to 'Local/0679963729@from-internal' (thanks to PJSIP/2006-0000004e)
    -- Not accepting call completion offers from call-forward recipient Local/0679963729@from-internal-00000016;1
    -- PJSIP/2006-0000004e Internal Gosub(app-missedcall-hangup,2006,1) start

Not accepting call completion offers from call-forward recipient Local/+33679963729@from-pstn-00000017;1
    -- PJSIP/Trunk_Diakelys-00000050 Internal Gosub(app-missedcall-hangup,Trunk_Diakelys,1) start

Executing [+33679963729@from-pstn:1] Set("Local/+33679963729@from-pstn-00000017;2", "__FROM_DID=+33679963729") in new stack
    -- Executing [+33679963729@from-pstn:2] NoOp("Local/+33679963729@from-pstn-00000017;2", "Received an unknown call with DID set to +33679963729") in new stack

No. from-pstn generally doesn’t allow any external numbers.

Thank you.

So how i can change the context ? Where is the settings ? It’s a trunk related settings ?

It’s being sent to the correct context. from-pstn would be the wrong context.

I’d have to trawl the Asterisk code and I would still need full “pjsip set logger on” tracing, to fully understand what is happening.

I suspect you have disabled forwarding somewhere.

I’d also suggest searching the fixed part of the Not forwarding message.

Thank you.

I will do the test with “pjsip longer on” on saturday.

Have a good day

What you mean “searching the fixed part of the Not forwarding message.” ?

The message says that the number 0679963729 is not in service (even though that’s my cell phone number)

I meant the “Not accepting” one.

Here the full logger on: https://pastes. io/JmtK3b2f

You just need to remove the space between “pastes.” and “io” because I couldn’t add that domain.

After doing some research, could this be an issue with the SIP header and the “From User” or “From Domain” fields? Is the message “Not accepting call completion offers from call-forward recipient Local/0679963729@from-internal-00000002;1” a response from the trunk or from Asterisk?

New test today: If I enable call transfer at the SIP line level rather than at the phone level (from UCP or via the feature code *72), it works. If I make an internal call between extensions 2007 and 2006, and then transfer from 2006 to an external number, it works. However, on any phone, if I enable call forwarding from the IP phone, it doesn’t work. I just set up a new VM with Asterisk 20 and have the same problem. Maybe it’s a configuration issue with my Asterisk that isn’t compatible with my trunk provider and is therefore being rejected?