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nighthawk

@albaniaonline.net

[Equipment] Issue with call transfer in Linksys PAP2t

Hello,
I am having an issue with transferring incoming calls from my pap2t to other extensions.
After answering the call, I press the flash button, then enter *98 and the extension number. I get the two beeps, than the call hungs up. I have tried also to dial ## + extension, but it is the same.
The provider allows this service.
Help is very appreciated.
Thanks in adavance

Stewart

join:2005-07-13
kudos:26
Can you do an attended transfer (flash, dial extension, wait for answer, hang up)? Can you transfer by hanging up on a conference (flash, dial extension, wait for answer, flash again, have three-way call, hang up)? If these don't work, make sure that Blind Attn-Xfer Enable, Xfer When Hangup Conf, Three Way Call Serv, Three Way Conf Serv, Attn Transfer Serv and Unattn Transfer Serv are all set to yes, and try both settings of Refer-To Target Contact. If no luck, try using SIP Debug to see what is going wrong.

Unlike the above features which use the SIP REFER request, ## (Asterisk blind transfer) is implemented entirely on the server. You press ## without flashing. Asterisk will interpret your key presses and transfer the call by itself -- the PAP2T doesn't do anything (except disconnect when Asterisk hangs up on it). If that's not working, the server-side setup is incorrect, or incompatible with some other enabled feature (such as re-invite).


Nighthawk

@opera-mini.net
said by Stewart:

Can you do an attended transfer (flash, dial extension, wait for answer, hang up)? Can you transfer by hanging up on a conference (flash, dial extension, wait for answer, flash again, have three-way call, hang up)? If these don't work, make sure that Blind Attn-Xfer Enable, Xfer When Hangup Conf, Three Way Call Serv, Three Way Conf Serv, Attn Transfer Serv and Unattn Transfer Serv are all set to yes, and try both settings of Refer-To Target Contact. If no luck, try using SIP Debug to see what is going wrong.

Unlike the above features which use the SIP REFER request, ## (Asterisk blind transfer) is implemented entirely on the server. You press ## without flashing. Asterisk will interpret your key presses and transfer the call by itself -- the PAP2T doesn't do anything (except disconnect when Asterisk hangs up on it). If that's not working, the server-side setup is incorrect, or incompatible with some other enabled feature (such as re-invite).

Hello Stewart,
I have tried doing both the attended transfer and the conference mode,but it either hangs up after dialling the extension or it stays on hook,until call is canceled.
I have tried to set all the settings that you mentioned to yes,but only on the refer. I will have to check them on the target too.
The server is not Asterisk-based,so i assume that the ## option falls.
I am pretty sure that the server supports these features,because it works with IP phones, but I will try to debug it,and let you know. Thanks

Stewart

join:2005-07-13
kudos:26
reply to nighthawk
IMO, the easiest case to debug is from a conference. If you can't get the three-way call established, the trouble is not related to transfer; report what happens and what the capture shows.

When you hang up (on a successful three-way call), you should see an in-dialog REFER request sent to the server and a 202 Accepted reply. Shortly thereafter, the server should send a NOTIFY request with a Sipfrag showing a 200 OK, and the ATA responds with a 200 OK. The server will also send a BYE for the transferred-to call, which the ATA will acknowledge with a 200 OK. A few seconds later, the ATA will send BYE for the original incoming call, which the server will acknowledge.

With luck, there will be an error indication in the response to REFER, or an error status in the Sipfrag.

If you still have trouble, compare to what your working IP phone does, or post some details here.


nighthawk

@albaniaonline.net
reply to nighthawk
I did a debug on server, and I see everything that you mentioned, except the 202 acceptance coming from server, as I see when I try to do the same with an IP phone, for the same line.
I did a better check, and the server offers some service activation codes, including * and # to activate three way calls and call transfers. I tried to do a three-way call, and it works, but the transfer still doesn't. Maybe the ATA's dial plan doesn't support it. Something that confuses me is that I don't have to dial them on the IP phone. I will have to ask for the assistance the company that did the server's implementation, as it is based on a proprietary's OS. Thank you very much for your help.