Mitel 3300 SIP Device (call transfer) 1

[DanielCMurray]

We currently have a Mitel 3300 R8.0 and have approx. 40 x Cisco 8841 phones we are using as SIP devices. We are looking for assistance in making the call transfer (transfer an active call from one extension to another) work on these phones.
Has anyone had any luck getting this to work?

 

[kwbMitel]

Try programming up a multicall key on the Mitel SIP phone.

Almost all SIP devices require a 2nd key to allow transfers.

 

[DanielCMurray]

I did that. When I press the transfer button the call is automatically put on hold and a new line is automatically selected, however when I dial an ext# and press the transfer button again, the action is just repeated. The call never physically transfers.

 

[jpruder]

Whhat does the sip device capability look like for these?

 

[Valamagules]

Have you tried FAC?

Clever men learns what Wise men shares!

 

[DanielCMurray]

Hi Valamagules. I looked at our Feature access codes in the system and did not see a FAC for internal transfer. I did see call forward but no transfer.

 

[DanielCMurray]

SIP Device Capabilities:
Call Routing and Administration Options
Outbound Proxy Server
Replace System based with Device based In-Call Features No
Allow MWI Notifications without Subscription Yes
Enable Digit Collection In Busy Or Alerting State Yes

Allow Device To Use Multiple Active M-Lines No
Allow Using UPDATE For Early Media Renegotiation No
AVP Only Device Yes
Enable Mitel Proprietary SDP No
Force sending SDP in initial Invite message No
Ignore SDP Answers in Provisional Responses No
Limit to one Offer/Answer per INVITE No
Prevent SDP Renegotiation If Peer Initiated Hold No
Prevent the Use of IP Address 0.0.0.0 in SDP Messages No
Renegotiate SDP To Enforce Symmetric Codec No
Repeat SDP Answer If Duplicate Offer Is Received No
Send Answer only after renegotiation is complete No
Suppress Use of SDP Inactive Media Streams Yes

Allow Display Update No
Disable Reliable Provisional Responses No
Disable Use of User-Agent and Server Headers No
Fail REFER To Keep Call Active On Mid-Call Feature No
If TLS use 'sips:' Scheme No
Multilingual Name Display No
Override Auto-Answer Headers No
Override Auto-Answer Headers With
Remove Anonymous User No
Require Reliable Provisional Responses on Outgoing Calls No
Suppress Redirection Headers No
Use P-Asserted Identity Header Yes
Use user=phone No

Registration Period Minimum 300
Session Timer 90
Session Timer: Local as Refresher No
Subscription Period 3600
Subscription Period Minimum 300
Subscription Period Refresh (%) 80
Invite Ringing Response Timer 0

 

[Valamagules]

Try FAC = Conference

Clever men learns what Wise men shares!