Does asterisk support the SIP UPDATE method?

Hi all,

I’m trying to understand if Asterisk ( I’m using V. Supports the SIP UPDATE method.
Documentation is not exactly clear.

According to some documents I read Asterisk should now support the UPDATE method.

During my Testing I’m using a cisco 7940 with one of the last SIP firmware, I can see the telephone " announcing" the method in the allowed list in the SIP INVITE header. When the INVITE is forwarded by the Asterisk to other endpoint that also supports SIP UPDATE it is stripped. This way the endpoint that replies to the INVITE, should it need it, will not use the UPDATE method even though the other endpoint supports it.

Right now my asterisk is configured to also force the call media through itself.

We also tested the same situation when the Asterisk is receiving calls from an ACME Packet Session Border Controller on the Asterisk SIP trunk side. The behaviour is the same.

Asterisk is stripping the UPDATE method from the allowed method list when received from an endpoint that supports it.

Below I include links to a couple of documents that I found. unfortunately they are not up to date. … 51524.html … ion=195589

Can you help me understanding if the SIP UPDATE method is now supported and what configuration is needed to enable it if possible ?

Thank you forehand.