The transfer app sends the same call to some other extension at some other dialplan within the switch.
We are not a sip specific switch so we have to abstract many concepts.
If you are trying to redirect the call we have 2 ways
the redirect application
<action application="redirect" data="sip:user@host"/>
This will result in a 302 redirect in the case of a sip channel
<action application="deflect" data="sip:user@host"/>
This will result in a REFER to the specified uri
That is the only 2 ways we support, if you are looking to divert like a proxy does
remember we are only a b2bua and do not perform proxy features, see openser for that.
Hi all at third today :-)
I'm trying to make call-forward (on all failures, but it is not
relevant, I think) but don't know how...
I used "transfer" application but it didn't work like i expected.
The scenario was as follows:
Really simple dialplan
<extension name="Eeee">
<condition field="destination_number" expression="^(23)$">
<action application="set" data="continue_on_fail=true"/>
<action application="bridge" data="sofia/default/$1%$${domain}"/>
<action application="transfer" data="22"/>
</condition>
</extension>
And than user 21 calls number 23 which is not registered. Call
succesfully rings on user 22 but:
1. There is no "181 Call is being forwarded" sent.
2. INVITE to user 22 lacks "Diversion" header.
If someone interested I have full pcap dump.
Could someone point me to right solution how to properly implement call
forwarding?
I mean 181 send back, and INVITE with Diversion header with proper
forwarding reason and forward counter. When I looked at
Channel_Variables I didn't find anything suitable...
Any suggestion is really appreciated :-)
Best regards,
kokoska.rokoska
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