Forward no answer trough sip devices don't work 1

[Striep]

Hello outthere,
I have the following issue:
On headoffice is a 500V2 running 9.1.7.0.163.
On 4 other locations is a Yeastar TA3200 media converter installed to conect the extension on site witch are connected with old wiring.
Sip extension 51813 has a forward on no answer and on busy set to sip extension 51812.
If a external call hits 51813 i see in the SSA that the call is forwarded to 51812 after the no aswer timer, however 51813 keeps ringing instead of 51812 starts ringing.
A wireshark trace shows that the invit hits 51812, however in the message header the "to" still hits 51813.

Anyone a fiks or known issue for this?

 

[janni78]

Would be interesting to see the SIP INVITE but probably the TA3200 doesn't handle it correctly.

"Trying is the first step to failure..." - Homer

 

[Striep]

It isn't going wrong in the TA3200, see the uploaded picture of the trace

 

[janni78]

If it works when you have 2 softphones/SIP phones connected directly to the IPO then it would be the TA3200 handling it wrong.

Set the TA3200 to look at Request URI instead of To.

"Trying is the first step to failure..." - Homer

 

[Striep]

Hello Janni78,

i just have the TA3200 and no other diveces with me.
I changed the config so the TA3200 looks to the "invite" field and it works fine now.

I still belief it isn't correctly send from the ip office.

Thanks for now, i will close this thread.

 

[janni78]

Usually the Request URI is the destination you wanna reach and the To header is the original destination, the Request URI changes for example when the call is forwarded but the To header remains the same.

You can read the SIP RFC if you're bored =)

What is the difference between request URI and to tag in SIP request messages?
In the context of telephony applications, the To header usually represents the dialled number along with the first UAS in the chain (and usually* remains the same between hops), while the Request-URI header identifies the routed number on the next UAS in the chain (and will almost certainly change for each hop)



"Trying is the first step to failure..." - Homer

 

[intrigrant]

Don't use the settings in the SIP phone, use the divert settings in Avaya manager or use the shortcodes which are designed for it:
*03 Forward On Busy On
*04 Forward On Busy Off
*05 Forward On No Answer On
*06 Forward On No Answer Off
*57*N# : forward on busy/no answer