Original incoming number displayed after call is tranferred

Hey all,

I have a problem with transferring calls on our SIP phones (Cisco SPA504G). They transfer correctly, but after the transfer is complete we would like the external callers phone number to be displayed on the phone (like it is when the incoming call rings).

Is this possible?


Depends on the capability of the phone and the version of Asterisk.

It may also depend on whether this is a SIP transfer or a features one, and, for SIP transfers, whether it is blind or attended, and, for attended, whether the REFER is directed at party A or party C. (Some phones actually do attended SIP transfers even when the UI claims a blind one.)

I believe Asterisk 1.8 introduced a feature that should update the displayed numbers, in a suitable environment.


I’m currently using asterisk but would upgrade to 1.8 if this was available.

This seems like a similar issue but is not using Asterisk (and isn’t the exact same model of phone): https://supportforums.cisco.com/thread/2109530. How should I go about finding out if it’s a SIP transfer or a features one? Wireshark and a softphone?


Your reference seems to relate to SIP transfers. Their problem seems to be that it is working the way you want! (I didn’t read it in depth, but it is possible that Asterisk is being bypassed.)

If you don’t know whether you have a SIP or features transfer, ask the person who configured Asterisk and the phone.

Normally if the word Transfer appears on the phone’s UI, you have a SIP transfer, in which case, if it is actually an attended transfer, the call to C is made, and therefore the initial CLID is set before Asterisk has any idea that a transfer is coming up. You need the 1.8 changes, and a phone that supports them, for there to be any possibility of the CLID being updated when the transfer is completed.

As noted before, some phones make an attended transfer, even if their user interface indicates a blind one. An actual blink one should show party A’s CLID, as Asterisk does know that it is a transer when C is first called.


Unfortunately I’m the one who set up the whole system!

The only sign of “transfer” in the phones settings is under “Vertical Service Activation Codes” in the Regional Settings and it’s called “Blind Transfer Code” and is set to “*98”.

I have a trace saved now. What should I be looking for?

Also, is it possible for the caller Id to be the transferring extension initially and then once the transfer is accepted the incoming caller id replaces this? This would be ideal.

Sounds like I should upgrade to 1.8 before testing further.


A features transfer will simply have DTMF in the RTP stream, for the transferror.

A blind transfer will have a REFER sip method, mid call, but only one call.

An attended transfer will have a second, outgoing, call to the destination, from the transferror, followed by a REFER, with Replaces, which could be on either the transferree call or the final destination call.

Okay, in my trace I was only monitoring UDP port 5060 on the destination phone. All I saw was a SIP INVITE, a Status 100, Status 180, Status 200 and then an ACK.

I’ll run a better trace and I’ll run a trace on the asterisk server also.