Hi Brian,
Which partyis generating the REFER ? the asterisk boxes from behind the
LB ? or the carrier side ?
and yes, see you in Amsterdam !!
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
On 02/05/2018 05:52 PM, Brian Southworth wrote:
I think I get it now thank you Bogdan.
So I would forward the traffic using the opensips proxy, using the if
(is_method(“refer”)) to the opensips box that would be the B2BUA? To
bridge the call ?.
Also look forward to Opensips summit in may 😊ill see you all there
got it booked Saturday 😊
Regards,
Brian Southworth
*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 05 February 2018 15:47
*To:* Brian Southworth <brian.southwo...@clocom.uk>; OpenSIPS users
mailling list <users@lists.opensips.org>
*Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no
socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Brian,
Keep in mind that you cannot make opensips act in the same time as
proxy (as required by the load balancer) and as a end-point (as
required by the B2BUA). Ideally is to run the two services (LB and
B2B) on two opensips instances in a chain.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
On 02/02/2018 07:03 PM, Brian Southworth wrote:
Sorry my apologies.
So from the beginning opensips acts as an authorization proxy
which passes the call on to an asterisk box based on load (using
load balancer).
I am trying to get the opensips proxy to handle call transfers and
I thought the b2bua would be the best way. Initially the refer was
sent to the asterisk box.
On inbound calls
The call comes in from the carrier goes to asterisk, asterisk then
passes the sip invite to the proxy which then rings the sip phone.
What I wish to achieve is a way to transfer an inbound call to an
internal extension or external number.
Example:
Caller A receives call àcaller A places call on hold and dials
caller B àcaller B picks up àcaller A presses cisco xfer and call
is passed to caller B
I was hoping to achieve this using the proxy or asterisk box if
possible.
I hope this helps.
Regards,
Brian Southworth
*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 02 February 2018 16:50
*To:* Brian Southworth <brian.southwo...@clocom.uk>
<mailto:brian.southwo...@clocom.uk>; OpenSIPS users mailling list
<users@lists.opensips.org> <mailto:users@lists.opensips.org>
*Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict:
no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
I'm a bit confused. The original report was on a record_route() /
loose_route() matter. But you say you have opensips as B2B, so the
RR mechanism must not be used in such a case - you act either as a
end-point, either as a proxy - you cannot be both for the same call.
Now you have this b2b error, during a call transfer scenario. and
you mentioned LB also :)...so I'm a bit confused - could please
try to put all these pieces together, so I can understand what you
are doing ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
On 02/02/2018 04:27 PM, Brian Southworth wrote:
Maybe I am doing this wrong but I wanted the B2BUA module to
handle the refer and bridge the calls.
I have the B2bUA working now. However my issue is that its not
able to send the replies.
incoming reply
b2b_reply (B2B.222.7591351.1517580641)
Feb 2 14:10:47 [22664] ERROR:tm:_reply_light: failed to
generate 408 reply when a final 200 was sent out
Feb 2 14:10:47 [22664] ERROR:b2b_entities:b2b_send_reply:
failed to send reply with tm
Feb 2 14:10:47 [22664]
ERROR:b2b_logic:b2b_logic_notify_reply: Sending reply failed -
408, [B2B.452.342.1517580641]
Do you need anything else to help me debug this ? I am not
sure why its failing to pass the reply with tm, I have enabled
the param:
modparam("tm", "pass_provisional_replies", 1)
I should also note that I am using the load balancer module
also. This normally deals with all call distribution. In and out.
Regards,
Brian Southworth
*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 02 February 2018 14:20
*To:* Brian Southworth <brian.southwo...@clocom.uk>
<mailto:brian.southwo...@clocom.uk>; OpenSIPS users mailling
list <users@lists.opensips.org> <mailto:users@lists.opensips.org>
*Subject:* Re: [OpenSIPS-Users] [15066]
WARNING:rr:after_strict: no socket found to match RR
[1][XXX.XXX.XXX.XXX:5060]
Hi Brian,
Maybe that warning points to a routing error that prevents the
REFER to be route to carrier - make a sip capture to be sure
the REFER from A is properly routed and accepted by the carrier.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
On 02/02/2018 01:38 PM, Brian Southworth wrote:
Hi Bogdan,
Thank you very much, so this doesn’t have any impact on
why the call being transferred are dropped ?
I am trying to transfer a call using the refer method as
that is what the cisco phones use.
The network is setup like so opensips proxy àasterisk
gateway àcarrier
Scenario:
Inbound call comes into the phone like so: carrier àast
àproxy àphone A
Phone A needs to transfer call to phone B: Phone A dials
phone B àphone B picks up àphone A presses xfer button and
call is dropped.
Any help would be appreciated.
Regards,
Brian Southworth
*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 02 February 2018 11:29
*To:* OpenSIPS users mailling list
<users@lists.opensips.org>
<mailto:users@lists.opensips.org>; Brian Southworth
<brian.southwo...@clocom.uk>
<mailto:brian.southwo...@clocom.uk>
*Subject:* Re: [OpenSIPS-Users] [15066]
WARNING:rr:after_strict: no socket found to match RR
[1][XXX.XXX.XXX.XXX:5060]
Hi Brian,
That warning means OpenSIPS found a Route header (while
doing loose_route) that is suppose to be of its own, but
the network information from the header does not match any
of the OpenSIPS SIP listeners.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
On 02/02/2018 11:14 AM, Brian Southworth wrote:
I get this when trying to transfer calls using the B2BUA:
[15066] WARNING:rr:after_strict: no socket found to
match RR [1][xxx.xxx.xxx.xxx:5060]
When I try looking on the mailing list there are no
other similar posts, could you please shed some light
on what maybe causing this please.
Regards,
Brian Southworth
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