Re: [OpenSIPS-Users] t_relay

yeah you are right. I changed my script to go to sems.  Caller is tls, sems is udp. Hence I need to use udp socket. The problem was that I forgot to call rewriteuri. Below is the solution.  Maybe it helps for somebody in the future.     rewriteuri("5556@x.y.z.t:5060");    

Re: [OpenSIPS-Users] t_relay

Hi Johan, The socket used for receiving the REGISTER is stored in the OpenSIPS registration. And used/forced when doing the lookup(). So, if the REGISTER was received via TLS, the TLS interface should be forced after the lookup. So how comes that opensips tries TLS (TLS Registration??)

[OpenSIPS-Users] t_relay

Hi, when a user is not reachable, I want to reroute my call to sems. the user is on tls, but sems is on udp. as I use rtpengine after doing lookup(location), I need to re relay the original message on the tls connection. However this fails. how can I do t_relay to tls socket ? Sep  7

Re: [OpenSIPS-Users] t_relay and forward accepts only literals

Thanks Liviu. To be precise it only worked when I changed both $ru and $du. I'm not sure if this is a good use. Also, because of that, now, when it comes from the other endpoint I have to force it to do the opposite (force to use UDP), what wasn't necessary before. Does it make any sense to

Re: [OpenSIPS-Users] t_relay and forward accepts only literals

That's because I just copy-pasted your example and adapted it without thinking. It should be: force_send_socket(sctp:10.12.8.108:5060 ) $ru = "sip:" + $od + ":" + $op + ";transport=sctp"; forward(); Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On

Re: [OpenSIPS-Users] t_relay and forward accepts only literals

Hi Liviu. I did the change and Opensips started up ok. But when I run my tests, no message get to the UAS. When I check the dump there's a lot of Destination Unreachable error for messages in UDP. It didn't change the transport protocol. Thanks anyway. Atenciosamente, Daniel Moreira

Re: [OpenSIPS-Users] t_relay and forward accepts only literals

t; Trainee > > > > *Von:* users-boun...@lists.opensips.org [mailto: > users-boun...@lists.opensips.org] *Im Auftrag von *Daniel Moreira Yokoyama > *Gesendet:* Dienstag, 22. März 2016 21:24 > *An:* OpenSIPS users mailling list <users@lists.opensips.org> > *Betreff:* [OpenSIP

Re: [OpenSIPS-Users] t_relay and forward accepts only literals

Hi Daniel, It's not a limitation, but rather a performance optimization which kills some of the user experience :) You can still do stateless forwarding to various destinations by setting the Request-URI ahead: force_send_socket(sctp:10.12.8.108:5060) $ru = "sctp:" + $od + ":" + $op;

Re: [OpenSIPS-Users] t_relay and forward accepts only literals

: Dienstag, 22. März 2016 21:24 An: OpenSIPS users mailling list <users@lists.opensips.org> Betreff: [OpenSIPS-Users] t_relay and forward accepts only literals Hi everyone. I'm still trying to undestand why I can't use t_relay(dest) or forward(dest) passing a variable instead of a literal.

[OpenSIPS-Users] t_relay and forward accepts only literals

Hi everyone. I'm still trying to undestand why I can't use t_relay(dest) or forward(dest) passing a variable instead of a literal. In my scenario, all I have to do is to receive a UDP message from a client and relay it on SCTP to the remote endpoint. The way I'm trying to achieve this is by:

[OpenSIPS-Users] T_Relay Re-Invite

Hi everyone, I have a problem with t_relay. When the session is established (last 200 OK) the phone receive a in-dialog invite from the PBX directly and since there every packet don’t go thought OpenSIPS but goes directly to the PBX meaning that I have no way to do anything on the active

Re: [OpenSIPS-Users] T_Relay Re-Invite

s.org>> Date: Wednesday, January 13, 2016 at 3:44 PM To: OpenSIPS users mailling list <users@lists.opensips.org<mailto:users@lists.opensips.org>> Subject: Re: [OpenSIPS-Users] T_Relay Re-Invite You need to record route the initial request so that your proxy stays in the signalli

Re: [OpenSIPS-Users] t_relay

Gary, Maybe take a look at the 0x02 flag for t_relay() : http://www.opensips.org/html/docs/modules/1.12.x/tm.html#id294571 by default, t_relay() is internally sending a negative reply if not able to send the request out (like DNS failure, bad IP, network error, etc). Regards, Bogdan-Andrei

Re: [OpenSIPS-Users] t_relay

at 8:37 AM From:Bogdan-Andrei Iancu bog...@opensips.org To:OpenSIPS users mailling list users@lists.opensips.org, Gary Nyquist g...@gmx.us Subject:Re: [OpenSIPS-Users] t_relay Gary, Maybe take a look at the 0x02 flag for t_relay() : http://www.opensips.org/html/docs/modules/1.12.x/tm.html#id294571

[OpenSIPS-Users] t_relay

Hi, Looking for some insight into the t_relay(). I am using the t_relay() to route INVITE to the Callee device (sip client running on a mobile device). I was expecting thet_relay() to fail (returning a -ve number), when the sip client is not running on the mobile device. But I find

Re: [OpenSIPS-Users] t_relay behavior with 477 send failed

Bogdan, I set the flag with t_relay and it resolved the issue. I am able to handle failure from t_relay now. Thanks, Amit On Mon, Mar 21, 2011 at 10:33 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Amit, have you set the 0x02 flag for t_relay ? See:      

Re: [OpenSIPS-Users] t_relay behavior with 477 send failed

Hi Amit, have you set the 0x02 flag for t_relay ? See: http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id293378 This flag will prevent the t_relay() function do internally deal with errors (by sending back a negative reply). Regards, Bogdan Amit Sharma wrote: I tried the fix

Re: [OpenSIPS-Users] t_relay behavior with 477 send failed

I tried the fix from the 1.6 branch. The issue I still face is that now t_relay returns with a value of 1 even when the relaying has failed. How do you figure out if t_relay has failed (!t_relay()) does not work in this case? Does a return value 1 also signify an error? -Amit On Tue, Mar 15,

Re: [OpenSIPS-Users] t_relay behavior with 477 send failed

Hi Amit, Following your report, Anca did a fix on SVN - see: http://lists.opensips.org/pipermail/devel/2011-March/007893.html Please update from SVN and try again. Regards, Bogdan Amit Sharma wrote: I am facing an issue similiar to the one outlined in the thread

[OpenSIPS-Users] t_relay behavior with 477 send failed

I am facing an issue similiar to the one outlined in the thread http://lists.opensips.org/pipermail/users/2010-April/011783.html I am using the latest stable version of opensips (1.6.4) and tried the solution outlined in the thread above. The issue I have observed is that t_relay function

[OpenSIPS-Users] t_relay(tcp:)

t_relay() default to udp. How can force it to tcp without specifying the host? Thanks, Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] t_relay(tcp:)

On Tue, Mar 8, 2011 at 12:23 AM, Dave Singer dave.sin...@wideideas.com wrote: Jeff, I like using the core rewritable variables to control things. They allow you to easily change just one part of the request and many times the equivalent function does not accept variables as arguments. $rP

Re: [OpenSIPS-Users] t_relay() not relaying payload

Hi Thamer, actually I see a 408 TIMEOUT for the second re-INVITE - but I see not outgoing re-INVITE in the second step (only the inbound message) Looking at the 2 re-INVITEs, I would say you have a routing issue as the first re-INVITE has as RURI a public IP

Re: [OpenSIPS-Users] t_relay() not relaying payload

Hi Bogdan, Sorry for not getting back sooner. I've updated my config a bit. I'm including what our reinvite handling looks like and the two reinvites that pass through opensips. The second one as you can see has no payload (ngrep shows ...) I have verified this as well under wireshark.

Re: [OpenSIPS-Users] t_relay() not relaying payload

Hi Thamer, Could you post the first re-INVITE (received and sent) to see what could be the problem? Regards, Bogdan Thamer Alharbash wrote: We currently have opensips setup to route through another carrier for certain calls. All signaling and media works well except for reinvites.

[OpenSIPS-Users] t_relay() not relaying payload

We currently have opensips setup to route through another carrier for certain calls. All signaling and media works well except for reinvites. if (has_totag()) { # sequential request within a dialog should # take the path determined by record-routing

[OpenSIPS-Users] t_relay() for Tel-URI

Hi OpenSIPS users, I have a UAC sending OpenSIPS a SIP INVITE with a SIP RURI. I need OpenSIPS to rewrite the SIP RURI into a TEL RURI and statefully relay this message to the UAS. Can I use the t_relay() function to forward the modified INVITE to the UAS? Since the TEL RURI does not

Re: [OpenSIPS-Users] t_relay() for Tel-URI

2009/6/3 David Roberge drobe...@xittel.net: Hi OpenSIPS users, I have a UAC sending OpenSIPS a SIP INVITE with a SIP RURI.  I need OpenSIPS to rewrite the SIP RURI into a TEL RURI and statefully relay this message to the UAS. Can I use the t_relay() function to forward the modified INVITE