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. On Sun, Apr 6, 2008 at 2:13 PM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
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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