So the target of the refer is to another Asterisk or may be also back to
the carrier ?
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
On 02/07/2018 01:32 PM, Brian Southworth wrote:
Hi Bogdan,
The Cisco phone, generates the refer once you press the xfer button
when inside a call.
Caller àopensipsàasteriskàCarrier
(cisco)
Regards,
Brian Southworth
Communications Developer
cid:image001.png@01D22CAC.1DCB8580
111 Wilmslow Road
Handforth
Wilmslow
SK9 3ER
T: 03333 446677
W: www.clocom.uk <http://www.clocom.uk/>__
cid:image002.png@01CDDC62.D8483910 <http://www.facebook.com/clocom.uk>
Like us on Facebook
cid:image003.png@01CDDC62.D8483910
Follow us on Twitter
cid:image004.png@01CDDC62.D8483910 <https://twitter.com/clocom_uk>
Clocom is a *green* company. Think, do you need to print this email?
This message contains confidential information and is intended only
for the individual named. If you are not the named addressee you
should not disseminate, distribute or copy this e-mail. Please notify
the sender immediately by e-mail if you have received this e-mail by
mistake and delete this e-mail from your system. E-mail transmission
cannot be guaranteed to be secure or error-free as information could
be intercepted, corrupted, lost, destroyed, arrive late or incomplete,
or contain viruses. The sender therefore does not accept liability for
any errors or omissions in the contents of this message, which arise
as a result of e-mail transmission. If verification is required please
request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road,
Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/>__
*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 07 February 2018 09:38
*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,
Which party is 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>
<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,
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
_______________________________________________
Users mailing list
Users@lists.opensips.org
<mailto:Users@lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
