Hi All,
I'm happy to announce that we have released rtpproxy v2.0.0.
You can review the release notes here:
https://github.com/sippy/rtpproxy/releases/tag/v2.0.0
-sobomax
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on restarting opensips I got the following errors in my log.
```
Mar 9 16:54:18 alpha /usr/local/sbin/opensips[24254]:
ERROR:usrloc:parse_phostport: bad protocol in ws:104.236.248.128:8080
Mar 9 16:54:18 alpha /usr/local/sbin/opensips[24254]: ERROR:usrloc:dbrow2info:
bad socket
Mar 9 16:54:1
I am sending an initial invite to opensips from a webrtc user agent, sip.js, to
a tcp registered user through opensips.
opensips gets the invite from sip.js, sends 100 trying ... then does nothing
and eventually the call gets timed out in sip.js.
I am pretty sure i got a invite from opensips li
Branch: refs/heads/master
Home: https://github.com/OpenSIPS/opensips
Commit: aba0f907153bb13932886cd059a8f5c6ab6ebd49
https://github.com/OpenSIPS/opensips/commit/aba0f907153bb13932886cd059a8f5c6ab6ebd49
Author: Vlad Paiu
Date: 2015-03-09 (Mon, 09 Mar 2015)
Changed paths:
I test above patch and it works!!! Superb! Many thanks to make it work!
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I see a strange behaviour using the route_to_carrier and use_next_gw function:
If I use a sip client whichs adds :5060 to the ruri and I receive a 503 Service
unavailable from the first gateway, then the ruri sent to the next gateway
(using the use_next_gw() function) is mixed up. The first pref
Branch: refs/heads/master
Home: https://github.com/OpenSIPS/opensips
Commit: 122ace5040ab495fff3cbcdf28fcbf08e5e7a940
https://github.com/OpenSIPS/opensips/commit/122ace5040ab495fff3cbcdf28fcbf08e5e7a940
Author: Liviu Chircu
Date: 2015-03-09 (Mon, 09 Mar 2015)
Changed paths:
Branch: refs/heads/master
Home: https://github.com/OpenSIPS/opensips
Commit: e25e0295f741e369e77fbbf76d7d9030d3dc46c9
https://github.com/OpenSIPS/opensips/commit/e25e0295f741e369e77fbbf76d7d9030d3dc46c9
Author: Razvan Crainea
Date: 2015-03-09 (Mon, 09 Mar 2015)
Changed path
I think we should also ask the question: "Should it work?". Because the pattern
parameter of algorithm 9 only supports a unique type of scripting variable: it
must be in shared memory, and it must be global (not related to any tm
structs), since concurrent reads from multiple processes must retu
It would be good to have some predefined logwatch-rules that automatically
parse the logfiles for OpenSIPS-related things so that a server administrator
easily can see if something unusual happens.
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Branch: refs/heads/master
Home: https://github.com/OpenSIPS/opensips
Commit: 3e440cbd42ea1d906072621700d28c58d6c7eb72
https://github.com/OpenSIPS/opensips/commit/3e440cbd42ea1d906072621700d28c58d6c7eb72
Author: Vlad Paiu
Date: 2015-03-09 (Mon, 09 Mar 2015)
Changed paths:
Branch: refs/heads/master
Home: https://github.com/OpenSIPS/opensips
Commit: 7903f2cb2011a6669b4affa347ba4fa5076ff265
https://github.com/OpenSIPS/opensips/commit/7903f2cb2011a6669b4affa347ba4fa5076ff265
Author: Razvan Crainea
Date: 2015-03-09 (Mon, 09 Mar 2015)
Changed path
force_send_socket() does not work for this case at all by the way.
If I use force_send_socket() it actually reverts to going to the A-record,
sending a tcp connection there with "SIP/2.0/UDP" in the Via header.
Tested with:
```
xlog("DBG: Forcing tcp socket for outbound");
force_send_socket(tcp:
Branch: refs/heads/master
Home: https://github.com/OpenSIPS/opensips
Commit: 178b0cc26b05a81947de150fe1c2df36d600ccaa
https://github.com/OpenSIPS/opensips/commit/178b0cc26b05a81947de150fe1c2df36d600ccaa
Author: Razvan Crainea
Date: 2015-03-09 (Mon, 09 Mar 2015)
Changed path
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