[OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
Bogdan-Andrei Iancu
bogdan at opensips.org
Wed Feb 7 04:38:06 EST 2018
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:bogdan at opensips.org]
> *Sent:* 05 February 2018 15:47
> *To:* Brian Southworth <brian.southworth at clocom.uk>; OpenSIPS users
> mailling list <users at 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:bogdan at opensips.org]
> *Sent:* 02 February 2018 16:50
> *To:* Brian Southworth <brian.southworth at clocom.uk>
> <mailto:brian.southworth at clocom.uk>; OpenSIPS users mailling list
> <users at lists.opensips.org> <mailto:users at 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:bogdan at opensips.org]
> *Sent:* 02 February 2018 14:20
> *To:* Brian Southworth <brian.southworth at clocom.uk>
> <mailto:brian.southworth at clocom.uk>; OpenSIPS users mailling
> list <users at lists.opensips.org> <mailto:users at 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:bogdan at opensips.org]
> *Sent:* 02 February 2018 11:29
> *To:* OpenSIPS users mailling list
> <users at lists.opensips.org>
> <mailto:users at lists.opensips.org>; Brian Southworth
> <brian.southworth at clocom.uk>
> <mailto:brian.southworth at 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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>
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>
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>
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