[OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
Brian Southworth
brian.southworth at clocom.uk
Fri Feb 2 09:27:08 EST 2018
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>; 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,
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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