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 <brian.southwo...@clocom.uk>; OpenSIPS users mailling list 
<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 <users@lists.opensips.org> 
<mailto:users@lists.opensips.org> ; Brian Southworth 
<brian.southwo...@clocom.uk> <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







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