Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
Yes, the chain is: carrier -> LB -> B2B -> Asterisk. And when Asterisk
generates an REFER (for call transfer) , the REFER will be received by
B2B which will handle it (it will generate a new call leg) without
changing anything on the leg connecting the b2b to 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 02/12/2018 04:41 PM, Brian Southworth wrote:
Hi Bogan,
Thanks for the reply, so are you saying the load balancer will send
the call over to the B2B and then to asterisk ?
Again sorry for my lack of knowledge there is still a lot I don’t
understand or know.
Regards,
Brian Southworth
Communications Developer
*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 12 February 2018 14:23
*To:* Brian Southworth ; OpenSIPS users
mailling list
*Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no
socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Brian,
In this case, I guess that the OPenSIPS B2B (handling the REFER)
should sit between the OpenSIPS LB and Asterisk - again , this is the
case only if the result of the transfer is a call to an Asterisk box
too. If the call may be redirected back to a carrier, the OpenSIPS B2B
should sit in front of the LB.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
On 02/07/2018 03:38 PM, Brian Southworth wrote:
Opensips handles the refer sending it to the asterisk box
Regards,
Brian Southworth
Communications Developer
111 Wilmslow Road
Handforth
Wilmslow
SK9 3ER
T: 0 446677
W: www.clocom.uk <http://www.clocom.uk/>__
__
*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 07 February 2018 11:36
*To:* Brian Southworth
<mailto:brian.southwo...@clocom.uk>; OpenSIPS users mailling list
<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]
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
T: 0 446677
W: www.clocom.uk <http://www.clocom.uk/>__
*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 07 February 2018 09:38
*To:* Brian Southworth
<mailto:brian.southwo...@clocom.uk>; OpenSIPS users mailling
list <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,
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
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Bogan, Thanks for the reply, so are you saying the load balancer will send the call over to the B2B and then to asterisk ? Again sorry for my lack of knowledge there is still a lot I don’t understand or know. Regards, Brian Southworth Communications Developer From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: 12 February 2018 14:23 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] Hi Brian, In this case, I guess that the OPenSIPS B2B (handling the REFER) should sit between the OpenSIPS LB and Asterisk - again , this is the case only if the result of the transfer is a call to an Asterisk box too. If the call may be redirected back to a carrier, the OpenSIPS B2B should sit in front of the LB. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com> OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam <http://www.opensips.org/events/Summit-2018Amsterdam> On 02/07/2018 03:38 PM, Brian Southworth wrote: Opensips handles the refer sending it to the asterisk box Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org <mailto:bog...@opensips.org> ] Sent: 07 February 2018 11:36 To: Brian Southworth <mailto:brian.southwo...@clocom.uk> ; OpenSIPS users mailling list <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] 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 <http://www.opensips-solutions.com> OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam <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 àopensipsà asteriskàCarrier (cisco) Regards, Brian Southworth T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org <mailto:bog...@opensips.org> ] Sent: 07 February 2018 09:38 To: Brian Southworth <mailto:brian.southwo...@clocom.uk> ; OpenSIPS users mailling list <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, 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 <http://www.opensips-solutions.com> OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam <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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Brian,
In this case, I guess that the OPenSIPS B2B (handling the REFER) should
sit between the OpenSIPS LB and Asterisk - again , this is the case only
if the result of the transfer is a call to an Asterisk box too. If the
call may be redirected back to a carrier, the OpenSIPS B2B should sit in
front of the LB.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
On 02/07/2018 03:38 PM, Brian Southworth wrote:
Opensips handles the refer sending it to the asterisk box
Regards,
Brian Southworth
Communications Developer
111 Wilmslow Road
Handforth
Wilmslow
SK9 3ER
T: 0 446677
W: www.clocom.uk <http://www.clocom.uk/>__
<http://www.clocom.uk/>__
*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 07 February 2018 11:36
*To:* Brian Southworth ; OpenSIPS users
mailling list
*Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no
socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
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
T: 0 446677
W: www.clocom.uk <http://www.clocom.uk/>__
__
*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 07 February 2018 09:38
*To:* Brian Southworth
<mailto:brian.southwo...@clocom.uk>; OpenSIPS users mailling list
<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,
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
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Bogdan, The Cisco phone, generates the refer once you press the xfer button when inside a call. Caller opensips asteriskCarrier (cisco) Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter 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 ; OpenSIPS users mailling list 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.openutions.com <http://www.opensips-solutions.com> OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam <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 <mailto:bog...@opensips.org> ] Sent: 05 February 2018 15:47 To: Brian Southworth <mailto:brian.southwo...@clocom.uk> ; OpenSIPS users mailling list <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 <http://www.opensips-solutions.com> OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam <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 caller A places call on hold and dials caller B caller B picks up caller 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 <mailto:bog...@opensips.org> ] Sent: 02 February 2018 16:50 To: Brian Southworth <mailto:brian.southwo...@clocom.uk> ; OpenSIPS users mailling list <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
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
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: 0 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
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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 ; OpenSIPS users
mailling list
*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
<mailto:brian.southwo...@clocom.uk>; OpenSIPS users mailling list
<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
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
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 ; OpenSIPS users
mailling list
*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
<mailto:brian.southwo...@clocom.uk>; OpenSIPS users mailling list
<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
<mailto:brian.southwo...@clocom.uk>; OpenSIPS users mailling
list <mailto:users@lists.opensips.org>
*Subject:* Re: [OpenSIPS-Users] [15066]
WARNING:rr:after_strict: n
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
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 ; OpenSIPS users mailling list
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 <http://www.opensips-solutions.com>
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
<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 à caller A places call on hold and dials caller B à
caller B picks up à caller 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
<mailto:bog...@opensips.org> ]
Sent: 02 February 2018 16:50
To: Brian Southworth
<mailto:brian.southwo...@clocom.uk> ; OpenSIPS users mailling list
<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 <http://www.opensips-solutions.com>
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
<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
<mailto:bog...@opensips.org> ]
Sent: 02 February 2018 14:20
To: Brian Southworth
<mailto:brian.southwo...@clocom.uk> ; OpenSIPS users mailling list
<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 <http://www.opensips-solutions.com>
OpenSI
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 instancesin 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 ; OpenSIPS users
mailling list
*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
<mailto:brian.southwo...@clocom.uk>; OpenSIPS users mailling list
<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
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
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 à caller A places call on hold and dials caller B à
caller B picks up à caller 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 ; OpenSIPS users mailling list
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 <http://www.opensips-solutions.com>
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
<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
<mailto:bog...@opensips.org> ]
Sent: 02 February 2018 14:20
To: Brian Southworth
<mailto:brian.southwo...@clocom.uk> ; OpenSIPS users mailling list
<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 <http://www.opensips-solutions.com>
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
<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 à asterisk gateway à carrier
Scenario:
Inbound call comes into the phone like so: carrier à ast à proxy à phone A
Phone A needs to transfer call to phone B: Phone A dials phone B à phone B
picks up à phone A presses xfer button and call is dropped.
Any help would be appreciated.
Regards,
Brian Southworth
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org
<mailto:bog...@opensips.org> ]
Sent: 02 February 2018 11:29
To: OpenSIPS users mailling list
<mailto:users@lists.opensips.org> ; Brian Southworth
<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
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 ; OpenSIPS users
mailling list
*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
<mailto:users@lists.opensips.org>; Brian Southworth
<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
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
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 ; OpenSIPS users mailling list
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 <http://www.opensips-solutions.com>
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
<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 à asterisk gateway à carrier
Scenario:
Inbound call comes into the phone like so: carrier à ast à proxy à phone A
Phone A needs to transfer call to phone B: Phone A dials phone B à phone B
picks up à phone A presses xfer button and call is dropped.
Any help would be appreciated.
Regards,
Brian Southworth
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org
<mailto:bog...@opensips.org> ]
Sent: 02 February 2018 11:29
To: OpenSIPS users mailling list
<mailto:users@lists.opensips.org> ; Brian Southworth
<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 <http://www.opensips-solutions.com>
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
<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
<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
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Brian, Maybe thatwarning points toa 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 ; Brian Southworth *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
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
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 à asterisk gateway à carrier Scenario: Inbound call comes into the phone like so: carrier à ast à proxy à phone A Phone A needs to transfer call to phone B: Phone A dials phone B à phone B picks up à phone 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 ; Brian Southworth 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 <http://www.opensips-solutions.com> OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam <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 <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
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 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
