Re: [OpenSIPS-Users] From a moment OpenSIPS ignores one end of the communication

Hi, another detail, when this happen Opensips continues to handle calls, it only ignores packets from this thread or call and when BYE arrive, work again. Hope someone can help me. Very Thanks. Regards. De: Users en nombre de Jorge Luis Ortea Enviado: lunes

[OpenSIPS-Users] From a moment OpenSIPS ignores one end of the communication

Hi all, Very strange case, OpenSIPS in the middle SIP transaction ignores one end of the communication. The schema is very simple, OpenSIPS exchanges SIP traffic between another OpenSIPS (with a passthrough T38 Asterisk behind) and a SBC, during a T38 fax call. The sniffer on the Opensips

[OpenSIPS-Users] UDP send failed

Hi all, I have a OpenSIPS between SBC and Asterisk PBX. This SIP Proxy relay outbound calls from PBX to SBC. Some calls when it is answered SIP Proxy receives from SBC a REINVITE or INVITE in-dialog, sometimes those REINVITES make fail SIP Proxy and it doesn't relay. - In log: ERROR: core:

Re: [OpenSIPS-Users] RV: Very Strange Case

ns.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 09/19/2017 12:14 PM, Jorge Luis Ortea wrote: Hi all, I've been using OpenSIPS for a long ti

[OpenSIPS-Users] RV: Very Strange Case

Hi all, I've been using OpenSIPS for a long time and I have never seen anything like it. The schema is very simple, OpenSIPS exchanges SIP traffic between SBC and Asterisk PBX, during a fax call in response a REINVITE packet a 200 Ok is ignore by OpenSIPS, not even writes in log, as if this

[OpenSIPS-Users] Audio problems

Hi, I am using several rtpproxies with Opensips, I have detected problems with the audio of some calls and when i show relays list can see this: # /usr/local/opensips/sbin/opensipsctl fifo nh_show_rtpp udp:10.201.200.53:9000:: set=1 index:: 2 disabled:: 1 weight:: 1

Re: [OpenSIPS-Users] ignoring IP Connection Information

Solved, r flag in rtpproxy_offer. Very Thanks, Regards. From: dar...@hotmail.com To: users@lists.opensips.org Date: Mon, 23 Feb 2015 17:53:18 +0100 Subject: Re: [OpenSIPS-Users] ignoring IP Connection Information One question, by default, any INVITE request managed by OpenSIPS, it sends RTP

Re: [OpenSIPS-Users] ignoring IP Connection Information

One question, by default, any INVITE request managed by OpenSIPS, it sends RTP flow to IP source signaling or IP (c=) in SDP ¿? In this case OpenSIPS public interface is in the same network than SBC, no nat and OpenSIPS has not nat_traversal implemented. When a INVITE is received from SBC with

[OpenSIPS-Users] ignoring IP Connection Information

Hi all, I am using OpenSIPS 1.8.2 + RTPProxy on bridge mode, I force a destination and a socket between 2 points: (public net)(private net) [SBC] --- [OpenSIPS+RTPProxy] --- [PBX] The signaling manage

[OpenSIPS-Users] False dialogs due to network failures

Hi all, I'm using OpenSIPS 1.8. with several Asterisk. When Proxy SIP manages a call keeps a dialog with (called,caller,Asterisk) through store_dlg_value function from dialog module. Later through get_dialog_info function it match calls and resolves transfers issue. I have the following