Hi list,
Is there a possibility now to download 1.9 beta version ? I tried with svn
links provided on opensips.org, but, I just can get 1.8.2, no more...
Thanks a lot,
*
Bien cordialement,
Best Regards,
**Kevin MATHY*
*HEXANET*
*
--
*
Phone : +33 (0) 3 26 79 30 05
Tech Support : +33 (0) 3 51
Hi Kevin,
The 1.9 (or trunk) is still under development and available only via
SVN. You can download it with an SVN client (see
http://www.opensips.org/Resources/Downloads#svn):
|svn co https://opensips.svn.sourceforge.net/svnroot/opensips/trunk
opensips_head|
Regards,
Bogdan-Andrei Iancu
Hi Brett,
The events stuff works a bit the other way around - you do not decide
where to send the event, but the consumer entities subscribe to you in
order to receive events. So it is not push, but pull. The consumer
decides (via subscription process) what event to receive, for how long
and
+1 as well.
Good value if we want Opensips to be a drop in replacement for SBC
combine with the work that was done on B2BUA.
I suppose that using Sangoma we will have very little latency introduced on
the media.
On Mon, Oct 29, 2012 at 7:06 AM, SamyGo govoi...@gmail.com wrote:
+1for new
On Fri, Nov 2, 2012 at 3:49 PM, Rudy r...@dynamicpacket.com wrote:
Bogdan,
Its great to hear all these ideas for 1.9 from the community. One thing
that may also be useful, is some enhancements to the rabbit_mq module. It
would be great if this module could also push/send events to a any
Hello All,
Not sure if this is the right place to suggest changes/new features. I have
a request for a new function for the load_balancer module.
Load_balancer knows how many resources are available for each gateway and
keeps track of active calls for each resource to distribute the calls
Ali,
Just to add to the scenarios when you don't use load balancer but want to
keep up with load.
If you are load balancing to Asterisk servers but you don't use the
load_balance() function because you want an attended transfer or someone
is call for a Parked Call. You want to make sure the
On 1 Nov 2012, at 13:02, Bogdan-Andrei Iancu wrote:
Hi Dan,
Well, as Ovidiu said, it is prone. BUT, this is not something particular to
re-INVITE, but to whatever
in-dialog pinging you may have from a mid-proxy.
I never said otherwise.
On the other hand, as Ryan pointed out here, the
On 3 Nov 2012, at 7:39, Binan AL Halabi wrote:
Hi ALL,
According to 14.1 and 14.2 of RFC 3261 after receiving 491 the UAC will retry
again after 2.1-4.0 seconds if it owns the CALL-ID or 0.0.-2.0 if not.
While the RFC may provide a means to handle this, it's non-practical. When I
put a
Hi Ali,
Sound interesting and realistic your scenario - I will add this to the
1.9 wish list.
Regards,
Bogdan
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 11/05/2012 04:17 PM, Ali Pey wrote:
Hello All,
Not sure if this is the right place to
Hi Duane,
More or less is the same (forcing a call as load for a destination
without actually doing LB) - in your case, what is different is the way
you identify the destination - But if I'm not wrong, using the IP of the
destination should work in both cases (when call comes form GW with IP
Yeah. What you mention in your email to Ali would work for the scenarios I
mention if there was a function to add a call to a gateway's resource
(maybe based on the IP address) without calling the load_balance() function
On Mon, Nov 5, 2012 at 12:30 PM, Bogdan-Andrei Iancu
Hi Rudy,
Events are send to the server/queue described when you subscribed for an
event (via the MI command -
http://www.opensips.org/Resources/DocsCoreMi18#toc15). or ?
Regarding ACC, the module itself does not generate events for acc
records, but you can do that from the script (see the
right :)
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 11/05/2012 08:40 PM, Duane Larson wrote:
Yeah. What you mention in your email to Ali would work for the
scenarios I mention if there was a function to add a call to a
gateway's resource (maybe
Hi Dan,
I agree it is not a easy thing to do, neither bullet proofed, but the
big question is : does it do more good than harm ? also, what are the
other options we have here ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 11/05/2012
@lists.opensips.org
Skickat: torsdag, 1 november 2012 20:36
Ämne: Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0
major release
2012/11/1 Bogdan-Andrei Iancu bog...@opensips.org
Hi Inaki,
Please correct me if I'm wrong, but reading the draft and listing you guys
all, I would say
2012/11/2 Binan AL Halabi binanalhal...@yahoo.com:
Hi All,
If oversip uses Path extension OpenSIPS must support it.
1- Sending Path header values in 200 ok REGISTER response
Even if this is required by RFC, the fact is that it's not needed at
all for this stuff to work.
2- Path header
2012/11/2 Bogdan-Andrei Iancu bog...@opensips.org:
Yes the patch is there, we need to review it. Also Inaki sugested some
tunings in the PATH module (to fully support websocket).
Well, there is no special needs for the PATH module to support
WebSocket. In fact, nothing must be done if it
On 11/02/2012 02:17 PM, Iñaki Baz Castillo wrote:
2012/11/2 Bogdan-Andrei Iancubog...@opensips.org:
Yes the patch is there, we need to review it. Also Inaki sugested some
tunings in the PATH module (to fully support websocket).
Well, there is no special needs for the PATH module to support
Bogdan,
Its great to hear all these ideas for 1.9 from the community. One thing
that may also be useful, is some enhancements to the rabbit_mq module. It
would be great if this module could also push/send events to a any rabbitmq
queue or exchange (maybe based on an avp) . Another thing that may
-Devel] [RELEASES] Planing OpenSIPS 1.9.0
major release
Hi Dan,
Well, as Ovidiu said, it is prone. BUT, this is not something particular
to re-INVITE, but to whatever in-dialog pinging you may have from a
mid-proxy.
On the other hand, as Ryan pointed out here, the need to check the
dialog health
Hi Inaki,
Please correct me if I'm wrong, but reading the draft and listing you
guys all, I would say the right approach is to : (1) use OverSIP as gw
(to extract SIP traffic from WebSocket) and (2) make OpenSIPS to support
SIP traffic resulted from websocket extraction.
If so, OpenSIPS has
On Nov 1, 2012, at 11:38 AM, Bogdan-Andrei Iancu wrote:
Hi Inaki,
Please correct me if I'm wrong, but reading the draft and listing you guys
all, I would say the right approach is to : (1) use OverSIP as gw (to extract
SIP traffic from WebSocket) and (2) make OpenSIPS to support SIP
Hi Dan,
Well, as Ovidiu said, it is prone. BUT, this is not something particular
to re-INVITE, but to whatever in-dialog pinging you may have from a
mid-proxy.
On the other hand, as Ryan pointed out here, the need to check the
dialog health from proxy side, without relying on special end-UA
2012/11/1 Bogdan-Andrei Iancu bog...@opensips.org
Hi Inaki,
Please correct me if I'm wrong, but reading the draft and listing you guys
all, I would say the right approach is to : (1) use OverSIP as gw (to extract
SIP traffic from WebSocket) and (2) make OpenSIPS to support SIP traffic
Hi Inaki,
Thanks for the confirmation and details.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 11/01/2012 09:36 PM, Iñaki Baz Castillo wrote:
2012/11/1 Bogdan-Andrei Iancubog...@opensips.org
Hi Inaki,
Please correct me if I'm wrong, but
Till: Bogdan-Andrei Iancu bog...@opensips.org
Kopia: OpenSIPS users mailling list users@lists.opensips.org
Skickat: torsdag, 1 november 2012 20:36
Ämne: Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0
major release
2012/11/1 Bogdan-Andrei Iancu bog...@opensips.org
Hi
users@lists.opensips.org
*Skickat:* torsdag, 1 november 2012 20:36
*Ämne:* Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS
1.9.0 major release
2012/11/1 Bogdan-Andrei Iancu bog...@opensips.org
Hi Inaki,
Please correct me if I'm wrong, but reading the draft and listing you
Hi Saul,
OK, aside the TCP part (which anyhow is scheduled for fixing) and some
extra parsing, does supporting WebRTC imply something more on the
OpenSIPS side ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 10/29/2012 12:42 PM, Saúl
On Oct 31, 2012, at 12:52 PM, Bogdan-Andrei Iancu wrote:
Hi Saul,
OK, aside the TCP part (which anyhow is scheduled for fixing) and some extra
parsing, does supporting WebRTC imply something more on the OpenSIPS side ?
It requires that OpenSIPS is able to use SIP over a WebSocket
Hi Bogdan,
Saul is correct. The key thing is to support WebSocket transport. WebRTC is
becoming quite popular and seems to be the thing of future. It is already
added in asterisk (version 11 released last week) and it is being added to
Kamalio. Using OverSIP and the OpenSIPS would make things
Hi guys,
Thanks for this - I will take a look at this websocket to see what about
- is there any RFC or similar ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 10/31/2012 03:53 PM, Ali Pey wrote:
Hi Bogdan,
Saul is correct. The key thing
Hi Bogdan,
On Oct 31, 2012, at 5:05 PM, Bogdan-Andrei Iancu wrote:
Hi guys,
Thanks for this - I will take a look at this websocket to see what about - is
there any RFC or similar ?
Here you go: http://datatracker.ietf.org/doc/draft-ietf-sipcore-sip-websocket/
Regards,
--
Saúl Ibarra
Asterisk 11 has some early support for SIP over websockets, but that's far from
being compatible with WebRTC. The standards for WebRTC are still evolving and
require much more. It's a good step forward, but the ASterisk team is not there
yet... :-)
SIP over websockets is currently a draft that
2012/10/31 Ali Pey ali...@gmail.com
Using OverSIP and the OpenSIPS would make things just more complex
specially for larger deployments.
Really? IMHO it makes things MUCH MORE simple. Use OverSIP as an Outbound
EDGE Proxy and you are done. Otherwise wait for a proper TCP/TLS and
WebSocket
Which one sounds simpler? Having a new layer of proxies and extra hardware
on different software packages with their own set of configurations,
limitations and bugs than having WebSocket enabled on opensips and control
your routing logic all in one place off of same DB.
Regards,
Ali Pey
On Wed,
2012/10/31 Ali Pey ali...@gmail.com
Which one sounds simpler? Having a new layer of proxies and extra hardware
on different software packages with their own set of configurations,
limitations and bugs than having WebSocket enabled on opensips and control
your routing logic all in one place
31 okt 2012 kl. 18:17 skrev Ali Pey ali...@gmail.com:
Which one sounds simpler? Having a new layer of proxies and extra hardware on
different software packages with their own set of configurations, limitations
and bugs than having WebSocket enabled on opensips and control your routing
On 29 Oct 2012, at 12:11, Bogdan-Andrei Iancu wrote:
Hi Saul,
We were thinking at re-INVITE pinging from OpenSIPs level towards the caller
and callee. There will be 2 modes (at least this is the current plan).
1) remember all the time the last SDPs from each side and re-use them when
On Wed, Oct 31, 2012 at 3:03 PM, Dan Pascu d...@ag-projects.com wrote:
On 29 Oct 2012, at 12:11, Bogdan-Andrei Iancu wrote:
Hi Saul,
We were thinking at re-INVITE pinging from OpenSIPs level towards the caller
and callee. There will be 2 modes (at least this is the current plan).
1)
+1for new module to drive SANGOMA cards in for transcoding (similar to
driving the rtpproxy or mediaproxy)
Regards,
Sammy
On Sat, Oct 27, 2012 at 4:41 AM, Ali Pey ali...@gmail.com wrote:
I do also see a lot of value in sip over websocket. WebRTC is pretty much
here and it makes much more
Hi Saul,
We were thinking at re-INVITE pinging from OpenSIPs level towards the
caller and callee. There will be 2 modes (at least this is the current
plan).
1) remember all the time the last SDPs from each side and re-use
them when pining the other sides (just to trick the SDP
Hi Ali,
I have to admit I'm not really familiar with the WebRTC and what are the
requirements for this to work directly over OpenSIPS - there is pending
patch for compatibility with WebRTC (parsing and detecting more VIA
params, specific to WebRTC), but we will need to have a clear view on
Hi Bogdan,
On Oct 29, 2012, at 11:26 AM, Bogdan-Andrei Iancu wrote:
Hi Ali,
I have to admit I'm not really familiar with the WebRTC and what are the
requirements for this to work directly over OpenSIPS - there is pending patch
for compatibility with WebRTC (parsing and detecting more VIA
Hi Brett,
I have also done some research on NoSQL backends and I discovered that MongoDB
is a much better suited candidate for storing accounting records an SIP traces
than key/value store databases.
http://cdrtool.ag-projects.com/projects/cdrtool/wiki/MongoDB
Adrian
On Oct 29, 2012, at
I do also see a lot of value in sip over websocket. WebRTC is pretty much
here and it makes much more sense to be able to support it on one proxy
server rather than having to use OverSIP and then OpenSIPS. WebRTC seems to
be very popular and the thing of tomorrow and it will be very important for
46 matches
Mail list logo
