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
____________ Virus checked by G DATA AntiVirusKit Version: AVK 16.7010 from 25.04.2006 Virus news: www.antiviruslab.com
this is quit difficult: Which SIP phones? Which version of Asterisk? ...
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.
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
Virus checked by G DATA AntiVirusKit Version: AVK 16.7010 from 25.04.2006 Virus news: www.antiviruslab.com
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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
Virus checked by G DATA AntiVirusKit Version: AVK 16.7010 from 25.04.2006 Virus news: www.antiviruslab.com
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
____________ Virus checked by G DATA AntiVirusKit Version: AVK 16.7010 from 25.04.2006 Virus news: www.antiviruslab.com
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
Virus checked by G DATA AntiVirusKit Version: AVK 16.7010 from 25.04.2006 Virus news: www.antiviruslab.com
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Virus checked by G DATA AntiVirusKit Version: AVK 16.7010 from 25.04.2006 Virus news: www.antiviruslab.com
I too have seen this problem using both Cisco and Polycom phones.
I have tried a very slim openser config in order to eliminate as many variables and still no success.
Cannot get inbound PSTN calls to "warm-transfer" from UA1 to UA2. SIP to SIP transfer to PSTN is fine.
Interestingly, on Polycom (and I suspect Cisco too), when a warm transfer is attempted, the transferring party cannot retrieve the call after the transfer key is hit the second time.
I have many traces but would be happy to do quite a bit more testing and post results if anyone has additional advice on some steps to investigate.
F
--- Klaus Darilion klaus.mailinglists@pernau.at wrote:
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
Virus checked by G DATA AntiVirusKit Version: AVK 16.7010 from 25.04.2006 Virus news: www.antiviruslab.com
Users mailing list Users@openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
Virus checked by G DATA AntiVirusKit Version: AVK 16.7010 from 25.04.2006 Virus news: www.antiviruslab.com
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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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
____________ Virus checked by G DATA AntiVirusKit Version: AVK 16.7061 from 28.04.2006 Virus news: www.antiviruslab.com
This looks fine, except that asterisk does not know how to call the new destination: (Refer-To: sip:00043551191@192.168.0.91:5060;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
Virus checked by G DATA AntiVirusKit Version: AVK 16.7061 from 28.04.2006 Virus news: www.antiviruslab.com
-
Bastian Schern -
Frogger -
Klaus Darilion