[OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]

Fri Feb 2 12:03:28 EST 2018

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:bogdan at opensips.org] 
Sent: 02 February 2018 16:50
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]

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:bogdan at opensips.org <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 <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:bogdan at opensips.org <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 <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

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