[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 06:35:57 EST 2018


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
>
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> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* 07 February 2018 09:38
> *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,
>
> 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:bogdan at opensips.org]
>     *Sent:* 05 February 2018 15:47
>     *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,
>
>     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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