Re: [OpenSIPS-Users] Functional Difference B2B_Bridge between OpenSIP 1.8 and 2.4.2

2019-07-31 Thread Louis Rochon
I have gone over the release notes for 2.4.2 to 2.4.6. I will upgrade and retest. Louis From: Users On Behalf Of Ben Newlin Sent: July 31, 2019 10:34 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Functional Difference B2B_Bridge between OpenSIP 1.8 and 2.4.2 WARNING:

Re: [OpenSIPS-Users] Functional Difference B2B_Bridge between OpenSIP 1.8 and 2.4.2

2019-07-31 Thread Ben Newlin
I’m not sure about the B2B behavior, but is there a reason you are not migrating to 2.4.6, which is the latest supported 2.4 release? It’s possible this problem has already been reported and fixed. Ben Newlin From: Users on behalf of Louis Rochon Reply-To: OpenSIPS users mailling list

[OpenSIPS-Users] Functional Difference B2B_Bridge between OpenSIP 1.8 and 2.4.2

2019-07-31 Thread Louis Rochon
Trying to upgrade my platform to OpenSIPS 2.4.2 (yea, about time, I know). We use this to move established calls from one server to another: /usr/local/sbin/opensipsctl fifo b2b_bridge sip:555@1.2.3.4 0 This works fine with OpenSIPS 1.8. The above causes this: 1. Invite (reinvite) to

Re: [OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread Johan De Clercq
And no need for ice=force. You can drop that. Also check your sdp settings. On Wed, 31 Jul 2019, 15:15 David Villasmil, wrote: > Hello, > > You need to do this for every leg of the call. This means: > > Call from SRTP client TO non-SRTP: > Remove the ICE, etc. > > When the REPLY with the 200

Re: [OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread David Villasmil
Hello, You need to do this for every leg of the call. This means: Call from SRTP client TO non-SRTP: Remove the ICE, etc. When the REPLY with the 200 SDP comes back FROM the non-SRTP, you need to ADD ICE, etc. Hope that makes sense David On Wed, 31 Jul 2019 at 14:03, Dragomir Haralambiev

Re: [OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread Dragomir Haralambiev
Hi, When change the answer flag to $var(rtpengine_flags) = " RTP/SAVP rtcp-mux-offer ICE=force"; rtpengine_answer("$var(rtpengine_flags)"); Call is connected but UAC1 not send and receive voices. Regards, Dragomir На ср, 31.07.2019 г. в 15:53 ч. Sasmita Panda написа: > Hi Dragomir, > > I

Re: [OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread Sasmita Panda
Hi Dragomir, I had mentioned to modify this according to your requirement . If your phone only support RTP/SAVP then change the flag what I have mentioned while answering . *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Wed,

Re: [OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread Johan De Clercq
Use rtp/savp On Wed, 31 Jul 2019, 14:40 Dragomir Haralambiev, wrote: > Hi, > > Thanks for your replay, but this not working. > > UAC1 receive 183 session progress with: > receive audio 50106 UDP/TLS/RTP/SAVP 0 8 18 101 > > UAC1 send to Opensips CANCEL. > > I make test with MicroSips latest

Re: [OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread Dragomir Haralambiev
Hi, Thanks for your replay, but this not working. UAC1 receive 183 session progress with: receive audio 50106 UDP/TLS/RTP/SAVP 0 8 18 101 UAC1 send to Opensips CANCEL. I make test with MicroSips latest version. Best regards, Dragomir На ср, 31.07.2019 г. в 15:04 ч. Sasmita Panda написа:

Re: [OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread Sasmita Panda
Hi , You have to do something like below wherever you are calling rtpengine_offer/rtpengine_answer. $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; rtpengine_offer("$var(rtpengine_flags)"); $var(rtpengine_flags) = "UDP/TLS/RTP/SAVP rtcp-mux-offer

[OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread Dragomir Haralambiev
Hello, I have 2 applications connected to Opensips+rtpengine: UAC1 -use encryption always. SRTP (RTP/SAVP) UAC2 - never use encryption . RTP (RTP/AVP) How to setup Opensips to make follow call: UAC1 SRTP -> Opensips+rtpengine ---> UAC2 RTP Thanks, Dragomir

[OpenSIPS-Users] CacheDB_Redis authentication

2019-07-31 Thread Yury Kirsanov
Hi, Could you please let me know correct syntax of modparam for cachedb_redis for password authentication? Or is it not supported at all? I've tried different variants but can't get it to work, always getting an error: Jul 31 19:22:56 ERROR:cachedb_redis:redis_set: Redis operation failure -

Re: [OpenSIPS-Users] mhomed=1 and force_send_socket()

2019-07-31 Thread Vitalii Aleksandrov
When user is registered everything is OK. The problem appears when I send a call to another proxy or carrier. With UDP transport opensips correctly detects an outgoing interface and forwards a call. With TCP/TLS transport opensips is not that smart. Found the place in code which detects an

[OpenSIPS-Users] Loglevel priority prefix in /var/log/messages

2019-07-31 Thread Denys Pozniak
Hello! Please help me to configure loglevel prefix priority (like INFO, NOTICE, WARNING, ERROR) for /var/log/messages. As you can see below OpenSIPS does not display it, but for example, Rtpengine does. ### Global Settings log_level=3 log_stderror=no log_facility=LOG_LOCAL0 ...

[OpenSIPS-Users] extra_fields not working in opensips 2.4.x

2019-07-31 Thread J E H A N Z A I B
Hi there, I used to have old version 1.11.x in the past and extra fields in the acc module used to work perfectly fine. I have updated my opensips to 2.4 now and extra_fields are not working. according to the documentation the from uri and to uri should be put in by default but its not. also the