Displaying 20 results from an estimated 5000 matches similar to: "Send INVITES and REFERs from OpenSIPS to Asterisk with multiple Contexts"
2020 Jan 29
0
Invitation for OpenSIPS Summit 2020 Call for Paper
Hello fellows VOIPer,
If you want to share with the rest of the VoIP & RTC community some
news, interesting or breaking through ideas, or even more, some
experience you had in terms of designing, integrating or operating
various solutions or platform based on Open Source Softwares, then you
should consider submitting a paper for the OpenSIPS Summit 2020 in May,
Amsterdam.
2013 Apr 09
1
[OpenSIPS-Users] 404 When BYE initiated by external callee
On Tue, Apr 9, 2013 at 1:22 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>wrote:
> **
> Hi Nick,
>
> The BYE is not properly formed and rejected by script - in the 200 OK of
> the INVITE, you can see that your opensips is doing Record-Routing, but the
> BYE does not contain the corresponding Route hdr, so SIP routing is
> impossible.
>
> Regards,
>
>
2014 Oct 15
0
OpenSIPS Summit Oct 21st before Astricon
Hello Everyone!
We wanted to let everyone coming to Astricon know that we will be
holding an OpenSIPS Summit on Tuesday Oct 21st, 2014 at the Suncoast
Casino & Spa.
Suncoast is about 10 minutes away from Red Rock and we will be provide
shuttle service to and from the Summit. For those of you that had to
book at Suncoast it should be really easy to find us!
Here are some things you can
2013 Mar 10
1
Register Free Opensips/Asterisk Integration
Hello Everyone,
I have gone through a few really good tutorials from the OpenSIPS
site, Asterisk resources etc.. The unanswered question (and final
piece of our puzzle) is if it's possible to have a register free
environment in an OpenSIPS/Asterisk integration. Most approaches have
OpenSIPS relay the UA's REGISTER request to Asterisk which has
"host=dynamic" set for the
2010 Apr 19
2
OpenSIPS with Asterisk Backend
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2009 Apr 13
0
opensips and asterisk canreinvite
Hi,
I'm using opensips as the registrar server for my users.
I am redirecting calls going out to pstn to my asterisk server.
call flow is basically:
ua --> opensips server --> * server --> sip gateway provider
if (uri=~"sip:00[0-9]*@sip\.myserver\.com") {
xlog("L_INFO", "Call to PSTN\n");
#strip(2);
#prefix("011");
2012 Jan 09
1
Asterisk as register server through OpenSIPS
Hi all,
I've been trying to register a SIP user agent to an Asterisk server using
OpenSIPS as SIP router. The functionality is working fine. However,
Asterisk uses the IP address of the OpenSIPS server as the peer IP address.
How can I use the original IP address of the peer without changing the
peer's nat=yes?
I appreciate any kind of help. Thanks!
Regards,
Ronald
-------------- next
2011 Mar 04
3
OT: OpenSIPS vs Kamailio -- which do you use and why?
I'm starting a new project similar to a previous project where I used
OpenSER to front a bunch of Asterisk servers.
Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely
candidates.
I'm leaning towards OpenSIPS because it's in EPEL so I can install it with
yum. Also, because I think the name sounds more 'professional' when
discussing architecture with clients :)
2009 Mar 20
1
Asterisk + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk
Hello All,
I have a little complicated question about the Dial command.
I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers.
Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers:
For each peer on Asterisk, "Addr->IP" is IP of the Proxy and "Reg. Contact" is the IP
2016 Jul 05
2
OpenSIPS or Kamailio based fronting for Asterisk?
Hello,
I am beginning to front my Asterisk cluster with OpenSIPS/Kamailio and so
far my biggest issue is the complete lack of quick-start-like documentation
for either. Is there any place I can get a very simple HA configuration
(telling me where the config files are, for starters, is a good thing) for
OpenSIPS or Kamailio with the following features:
(a) Support an arbitrarily large number of
2009 Mar 20
3
OpenSIPS on CentOS
Hello,
I've been looking into OpenSIPS to see if it's a worthwhile addition to our setup. We're currently running a cluster, using Heartbeat, between two servers. It works well but I'm interested in seeing if we can improve it. My manager heavily uses RPM's for installations rather than source, particularly using yum to update. I'm trying to actually install OpenSips via
2005 Jun 17
0
Agents/Queues Contexts
Is there a way to define multiple contexts for agents/queues such
that in a multi-tenant environment, there could be two different,
say, Agents 1000?
I'm setting up a multi-tenant configuration and I'm giving each
tenant a web-based interface to define their own agents and I
wouldn't want to restrict one tenant from choosing an agent because
another tenant (which the tenant
2008 Dec 13
3
SER, OpenSER, Kamailio, OpenSIPS -- what are you using?
One of the above is frequently used to front-end Asterisk.
I used OpenSER to front-end a farm of Asterisk servers and was very happy
with it. The ability to take a box out of service or to route a specific
DNIS to a box for testing rocks.
Since OpenSER has died (I don't care about the
politics/personalities/trademarks), Kamailio and OpenSIPS have risen from
the ashes. What are you using?
2010 Dec 01
4
Asterisk with MySQL Cluster
I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant
High-Availability. I was wondering if it's possible for Asterisk to also
use multiple database servers for Realtime? Currently with Realtime I am
only able to point to a single IP address for a database. If that database
server goes down that Asterisk is pointed to then Asterisk won't be able to
do anything.
2005 Sep 07
2
asterisk, SIP, Re-INVITEs and different contexts
Hmmm... Folks, I beg you pardon, if I'm telling something which was said
before, but actually I have not found this anywhere, neither on
Voip-info.org or in several Asterisk's docs.
So, here is the statement:
If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them
will ALWAYS go via Asterisk.
I.e. Asterisk WILL NOT issue Re-INVITE even if:
1. Both UAs have
2010 Feb 24
3
Re-INVITE on BYE
Hi gurus,
In need of a little help here. I?m trying to do the Asterisk media release
by using canreinvite=yes. But I found weird behaviour when comes to BYE.
Below are my current setup:
Client A is registered to Opensips
Client B is registered to Asterisk
A ? Opensips ? Asterisk ? B
On hangup below are the SIP flow which I?ve notice from the Asterisk server
itself:
1. Opensips forward the BYE
2020 Oct 28
4
PJSIP tight loop on auth failure
Hi,
We're using Asterisk 13.17.0 with PJSIP 2.8 bundled.
I've found an issue when Asterisk tries to make a SIP call out using
auth, but has the wrong credentials and keeps getting returned a SIP
407, in this example to an OpenSIPs server requiring user auth.
Basically this happens:
1. Asterisk sends plain INVITE to OpenSIPs
2. OpenSIPs responds with SIP 407 auth required with a
2019 Feb 26
2
funnel shift, select, and poison
If I got poison propagation right, it's probably only by luck!
Hopefully, the funnel shift bug is fixed here:
https://reviews.llvm.org/rL354905
Nuno, IIUC this means that you do *not* need to change the funnel shift
semantics in Alive.
So I think that means we're still on track to go with John's suggestion
that only select and phi can block poison?
(I don't know of any
2018 Jan 26
0
RFC: Using link-time optimization to eliminate retpolines
Wouldn't a branch funnel open the door to a type 1 attack?
E.g. if the code looks like this, then a branch funnel basically turns into
a standard type 1 pattern AFAICT:
struct Base {
virtual int f(long) = 0;
};
struct A : Base {
int f(long x) override {
return 0;
};
};
struct B : Base {
int f(long x) override {
// As in listing 1 in
2009 May 15
1
Spiral SIP Request problem
Hello,
I am using OpenSIPS to register all the users and planning to use asterisk
for Auto Attendant, Queues, Voicemail and Conference Bridge.
I have a scenario where the signaling does not happen properly:
1) A user from Opensips dials an extension 7000 which is an
auto-attendant extension. The call is routed to asterisk to play the auto
attendant messages like Welcome and Dial the