I am working on setting up call forwarding. Currently, a user calls PBX A (not my PBX) from the PSTN, and then PBX A forwards the call to my Asterisk PBX. PBX A and my Asterisk PBX are connected via a SIP trunk. (PBX A registered in Asterisk PBX pjsip as a endpoint) ; pjsip in Asterisk PBX [PBX_A] …

https://docs.asterisk.org/Asterisk_20_Documentation/API_Documentation/Dialplan_Applications/Transfer/

Thanks for reply. I tried Transfer(), but it cannot transfer the call to the PBX A endpoint. It will make the call end. Here is my code: ; when user click #, user will goto this Transfer function exten = 1234,1,Transfer(pjsip/5555@PBX_A) I used WireShark to catch the SIP package, nothing happen…

chan_pjsip requires a full URI after the /. Also note that, to use REFER, the call must have been answered, otherwise a 302 status will be used.

Sorry I am new in Asterisk. May I ask for an example of ‘full URI after the /’ ? If the IP address is XX.XXX.XX.XX, Is it look like this? pjsip/XX.XXX.XX.XX:5060/5555 Thanks

See SIP-URI and SIPS-URI, towards the middle of this page RFC 3261: SIP: Session Initiation Protocol

However, you cannot set even valid headers on REFER, and CTI_Data is not a valid header name, as it is not marked as being non-standard, and there appears to be no RFC defining it.

Do you mean that I cannot set CTI_Data in the Refer with Asterisk?

[image] PJSIP Transfer with Headers Asterisk Dialplan How do I attached SIP headers to a Transfer (SIP Refer)? According to the second note PJSIP_HEADER wiki, I’m probably attaching the header to the incoming channel. But Transfer doesn’t have a pre-dial handler to attach …

Seem it can transfer the call, but the call changed to on hold, after a few seconds it pop up Time out request PBX_A still did not receive anything. Am I missing anything? Here is my extensions.conf in Asterisk [image] [image] Here is my Refer package, which 10.11.30.80 is my Asterisk PBX loca…

Please provide the complete REFER transaction, as plain text, not an image.

Sorry about it, here is the plain text I saw there are a 5524 NOTIFY after the call forward. In this case, call flow: 1001@10.11.30.26 → (call) → 1000@10.11.30.80 → (call forward) → sip:5555@XX.XXX.XX.XX <--- Transmitting SIP request (706 bytes) to UDP:10.11.30.26:56759 ---> REFER sip:1001@10.11…

Call forward with refer and CTI data

Asterisk Asterisk SIP

david551 August 7, 2024, 6:06pm 9

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Call forward with refer and CTI data

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