Re: [OpenSIPS-Users] multiple retransmissions on 4XX after ACK

2016-10-07 Thread Richard Robson
The ack is from the cp. opensips is sending the 4xx. I'll try the same on another provider. Thanks for the info On 7 Oct 2016 18:27, "Newlin, Ben" wrote: > The most likely cause is that there is something wrong in the format of > the ACK which is causing the far end to not recognize it as being

Re: [OpenSIPS-Users] multiple retransmissions on 4XX after ACK

2016-10-07 Thread Newlin, Ben
The most likely cause is that there is something wrong in the format of the ACK which is causing the far end to not recognize it as being the ACK for that 4XX response. So the far end will continue to retransmit the 4XX response. On the OpenSIPS side, these are recognized as retransmissions so t

[OpenSIPS-Users] multiple retransmissions on 4XX after ACK

2016-10-07 Thread Richard Robson
Hi Guys, Not sure if this a problem or normal behaviour or I'm doing something wrong. after a 4XX is sent and the ACK recieved I can see 3 retransmissions of the 4XX message all with corresponding ACKs. Whywould it re transmit after an ACK in the logs in the reply route I only see one. Is t

[OpenSIPS-Users] t_on_failure and 401 Unauthorized

2016-10-07 Thread James Thomas
Hi, I'm trying to have my front end opensips server fail over to different backend registrar servers when one registrar server is . I thought I could make t_on_failure trigger after the timeout waiting for the response from server 1. Unfortunately when server 1 is UP, the 401 challenge is being tr

Re: [OpenSIPS-Users] BYE with different transport

2016-10-07 Thread SamyGo
Yes record route headers are set just as the default config file has them. Also do note that the A party and B party are not registered users. The setup is also behind NAT as well. Regardless of these two the calls work completely just flawless in case the caller side is on UDP. I do see BYE reac

Re: [OpenSIPS-Users] FW: Asynchronous operation for REST queries

2016-10-07 Thread Bogdan-Andrei Iancu
Hi Agalya, Fixes are pushed also in minor releases. So all the fixes done in trunk will also be part of the 2.2.2 minor release. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 03.10.2016 22:08, Ramachandran, Agalya (Contractor) wrote: Hi te

Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

2016-10-07 Thread Newlin, Ben
No problem. Thanks, but I am not using 2.2 and not using the topology_hiding module. I am using the Dialog module with the topology_hiding function in 1.11. Ben Newlin From: Bogdan-Andrei Iancu Date: Friday, September 30, 2016 at 4:39 AM To: "Newlin, Ben" , OpenSIPS users mailling list Subj

Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

2016-10-07 Thread Bogdan-Andrei Iancu
Ben, In 1.11, if you do TH, you should use match_dialog() function and not loose_route() at all. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 30.09.2016 16:21, Newlin, Ben wrote: No problem. Thanks, but I am not using 2.2 and not using t

Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

2016-10-07 Thread Newlin, Ben
Yes, that is what you suggested before. My comments below were stating that that does not work. Specifically, “the match_dialog function must do loose routing on its own” because even when I only call match_dialog() from the script, I can still see the loose_route processing being triggered in t

Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

2016-10-07 Thread Bogdan-Andrei Iancu
Ben, The match_dialog() does not uses or rely on the loose_route() functionality- I briefly checked the code in 1.11; but if you have some logs to show it otherwise, do not hesitate to share with me (of course, if you are 100% sure you do not also call loose_route() from scritp ;) ). Regards