I am trying to handle the REFER method for call-transfer and I need some assistance.
What is the openser.cfg structure to handle a refer from ua1 to ua2 so that ua2 can call ua3?
Basically, I am trying to establish a new INVITE with the REFER-To for ua2 calling ua3. Also, I need to clean up the initial transaction between ua1 and ua2.
BTW, I read the rfc on REFER and I have a good sense of whats supposed to happen but I lack the mechanics to make it so.
All the guidance will be very helpful. Thanks, F
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REFER is an end-2-end feature. Thus, After ua1 sends REFER to ua2 (Refer-To: ua3), ua2 will create a new INVITE to ua3. You can not intercept the REFER in the SIP proxy and create an INVITE.
The only thing you might want to do in the proxy is to screen the URI in the Refer-To - e.g. to avoid that your subscribers are referred to costly service numbers.
regards klaus
Frogger wrote:
I am trying to handle the REFER method for call-transfer and I need some assistance.
What is the openser.cfg structure to handle a refer from ua1 to ua2 so that ua2 can call ua3?
Basically, I am trying to establish a new INVITE with the REFER-To for ua2 calling ua3. Also, I need to clean up the initial transaction between ua1 and ua2.
BTW, I read the rfc on REFER and I have a good sense of whats supposed to happen but I lack the mechanics to make it so.
All the guidance will be very helpful. Thanks, F
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So ua2 should handle the REFER, create and new invite to ua3 and then ua2 should then BYE to ua1 once the new call is active.
Whats interesting though is that in my implementation, SIP to SIP calls are able to transfer to other ua's and outbound to the PSTN gw. However Inbound PSTN calls (routed to a registered ua) cannot be transferred from SIP-ua1 to SIP-ua2.
In this case (PSTN-ua1 to SIP-ua2 trx SIP-ua3) ua1 is the gw and it should issue a new invite?
So in this case, would the issue be the Cisco gateway?
Can openser intercept the REFER-TO, issue a new invite from PSTN-ua to SIP-ua3?
Also, I like the idea of screening the REFER-to. In fact I plan on setting up domain variables using avps to enable or diable transfers.
Thank you very much for your feedback.
FR
--- Klaus Darilion klaus.mailinglists@pernau.at wrote:
REFER is an end-2-end feature. Thus, After ua1 sends REFER to ua2 (Refer-To: ua3), ua2 will create a new INVITE to ua3. You can not intercept the REFER in the SIP proxy and create an INVITE.
The only thing you might want to do in the proxy is to screen the URI in the Refer-To - e.g. to avoid that your subscribers are referred to costly service numbers.
regards klaus
Frogger wrote:
I am trying to handle the REFER method for call-transfer and I need some assistance.
What is the openser.cfg structure to handle a
refer
from ua1 to ua2 so that ua2 can call ua3?
Basically, I am trying to establish a new INVITE
with
the REFER-To for ua2 calling ua3. Also, I need to clean up the initial transaction between ua1 and
ua2.
BTW, I read the rfc on REFER and I have a good
sense
of whats supposed to happen but I lack the
mechanics
to make it so.
All the guidance will be very helpful. Thanks, F
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protection around
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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My gateway is receiving the refer correctly but apparently not acting on it.
The refer is properly formed and the SIP URL contains:
Refer-To: sip:111@sip.domain.com
and all of the other headers.
Anyone know how a cisco gateway is supposed to handle refers?
Thanks in advance for your assistance.
FR
--- Juha Heinanen jh@tutpro.com wrote:
Frogger writes:
Also, I like the idea of screening the REFER-to.
In
fact I plan on setting up domain variables using
avps
to enable or diable transfers.
permissions module has function allow_refer_to.
-- juha
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I have a dial-peer that matches the "name" portion of the URI.
For example, the refer-to is 3125551212@sip.refer.com
I have a dial peer as follows:
dial-peer voice 312 pots destination-pattern 312....... direct-inward-dial prefix 312 port 1/0:3:D
It matches using - sh dialplan number 3125551212
Yet it doesnt go on the refer.
I am concerned about the "@sip.refer.com". I am not sure how the gateway is handling this.
Any ideas? I am sure its a simple setup on the gw.
--- Juha Heinanen jh@tutpro.com wrote:
Frogger writes:
Anyone know how a cisco gateway is supposed to
handle
refers?
cisco gw does support refers. you just need to have a dial peer that matches to the refer-to uri.
try debugging sip and dial plans on your gw.
-- juha
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I have done a lot of investigation and debugging. Here is where I am so far.
I am able to get transfers working in the following situation:
++This works++ ============== SIP-ua1 calls pstn phone through gw
SIP-ua1 transfers pstn phone to final-callee: SIP-ua2 ==============
**This does NOT work** ============== PSTN phone calls SIP-ua1
SIP-ua1 transfers pstn phone to final-callee: SIP-ua2 ==============
Basically, if sip-ua1 is the primary caller, the transfer works. If pstn gw is the primary caller, the transfer does not work.
Any thoughts?
Please, even the slightest comment can be helpful to crack this case.
Thank you! FR
--- Juha Heinanen jh@tutpro.com wrote:
Frogger writes:
I am concerned about the "@sip.refer.com". I am
not
sure how the gateway is handling this.
my understanding is that cisco doesn't look host part at all. as i said, debug your dial plan when refer comes in.
-- juha
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Hi,
take a look on the network trace and see if the REFER is properly routed also in the second case.
regards, bogdan
Frogger wrote:
I have done a lot of investigation and debugging. Here is where I am so far.
I am able to get transfers working in the following situation:
++This works++
SIP-ua1 calls pstn phone through gw
SIP-ua1 transfers pstn phone to final-callee: SIP-ua2
**This does NOT work**
PSTN phone calls SIP-ua1
SIP-ua1 transfers pstn phone to final-callee: SIP-ua2
Basically, if sip-ua1 is the primary caller, the transfer works. If pstn gw is the primary caller, the transfer does not work.
Any thoughts?
Please, even the slightest comment can be helpful to crack this case.
Thank you! FR
--- Juha Heinanen jh@tutpro.com wrote:
Frogger writes:
I am concerned about the "@sip.refer.com". I am
not
sure how the gateway is handling this.
my understanding is that cisco doesn't look host part at all. as i said, debug your dial plan when refer comes in.
-- juha
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Frogger writes:
**This does NOT work**
PSTN phone calls SIP-ua1
SIP-ua1 transfers pstn phone to final-callee: SIP-ua2
Basically, if sip-ua1 is the primary caller, the transfer works. If pstn gw is the primary caller, the transfer does not work.
are you sure that your pstn ge support refer? if yes, turn debug on and see what it is doing when it receives refer.
-- juha
-
Bogdan-Andrei Iancu -
Frogger -
Juha Heinanen -
Klaus Darilion