Hello,
I have a problem with Asterisk 13 (freepbx installation on ubuntu 14.04) and call transfer (external call -> phone1 to phone2)…
when I receive a call and I answer the phone1 and I want to transfer on the phone2 line falls (internal call is ok)…
How can I solve this problem?
Thank You very much
Luca
Please note that FreePBX is not supported here.
There is insufficient information to debug it as pure Asterisk, but details of the transfer method used (features, SIP attended, SIP blind, SIP blind simulated by attended, DAHDI register recall, etc.,) would help.
Hello,
thanks for the reply,
Use two telephones ip cisco 303 with 1 internal for phone setup…
I have one server on digitalocean.com…
no register call
queue yes
cal flow control with day/night
Thanks and sorry for my bad english
skinny or SIP?
Features or native SIP transfers?
Hello,
SIP, with feature (queue, and “ringall” on extension)…
Thanks!
By feature, I mean to you dial a code to initiate the transfer, as against using the phones’s built-in transfer capability?
Hello,
I Use the key “xfer” on the cisco phone…
xfer -> extension number -> xfer
Thanks
I think we are talking about a SIP attended transfer. Are you doing it as a full attended transfer, and if so, can you talk to the new target before you try to complete the transfer. If you are doing a “blonde transfer”, in Asterisk terms (completing the transfer whilst the called party is still ringing), what happens if you do it as a fully attended transfer?
What does the log show, at verbosity 5, for the enquiry part of the transfer?