This looks fine, except that asterisk does not know how to call the new
destination: (Refer-To:
sip:[EMAIL PROTECTED]:5060;line=fnhf1aep?Replaces=3c267e322bf2-hcvubihsgoqa%40snom360-000413230066%3Bto-tag%3Dgpij52sisv%3Bfrom-tag%3Djhu81hu4vn
)
watch asterisks log files, make sure asterisk is using the proper
context when looking in extensions.conf.
google for asterisk transfer context
regards
klaus
Bastian Schern wrote:
I made an dump of a working scenario (97-96-91.html) and a dump of the
problematic scenario (PSTN-96-91.html).
Regards
Bastian
Klaus Darilion schrieb:
then we will need some more SIP dumps to help you.
"ngrep -d any port 5060" on the SIP proxy.
regards
klaus
On Tue, April 25, 2006 20:00, Bastian Schern said:
Klaus Darilion schrieb:
this is quit difficult: Which SIP phones? Which version of Asterisk?
...
I use snom 360 and 200 phones, Asterisk 1.2.7.1 and OpenSER 1.0.1
You have to make sure that Asterisk and the SIP phones are
"compatible".
There are several ways how to make a call transfer.
Also an often seen problem is the different dialing plans on openser
and
Asterisk. Asterisk must be able to call B in the same way (same request
URI) then A calls B.
Of course Asterisk is able to call A or B in the same way.
Regards
Bastian
regards
klaus
Bastian Schern wrote:
Hello,
does anybody got a working configuration to make an "attended call
transfer" with a call through an Asterisk gateway?
Example:
PSTN --> Asterisk --> SER --+-- A
|
+-- B
The call will come from the PSTN Network and will go through "A". A
sets the call on "Hold" and calls "B". After A is connected with B, A
hangup an B got the call from PSTN.
This in _not_ working at the moment.
Attended call transfer only with OpenSER and only SIP-Phones is no
Problem. But if the is an Asterisk as PSTN-GW in the game it will not
work.
Regards
Bastian
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