Displaying 20 results from an estimated 1000 matches similar to: "Invitation for OpenSIPS Summit 2020 Call for Paper"
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
2010 Apr 19
2
OpenSIPS with Asterisk Backend
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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
2010 Oct 27
0
Send INVITES and REFERs from OpenSIPS to Asterisk with multiple Contexts
I currently have OpenSIPS set up with users and most of my call handling.
OpenSIPS won't be able to handle things like Call Park, Hunt Groups, ACD,
etc. So I want to send these types of requests to Asterisk. I also want to
set Asterisk up as Multi Tenant. So my question is
How can I send requests to Asterisk and have them funnel into the specific
context for that specific Tenant? So if
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
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
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?
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
2015 Nov 20
2
SIP calls dropping at 15 minutes
I have a problem where SIP calls from some providers are dropping at 15
minutes.
The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS
sends calls to an Asterisk server.
Below,
'Client' is the IP address of the client's host (running
FPBX-2.8.1(1.8.20.0)
'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls
'Asterisk' is the IP
2010 Sep 17
0
need help with IVR dialplan
Hi list
i setup successfull asterisk version 1.4 + opensips,
Opensips is the Registrar Server, Asterisk is the IVR server
the call flow
IP phone ---INVITE 1001----> opensips -----> ASterisk ----INVITE
5001--->opensips ---> Busy|cancel|404..--->asterisk---wait 10s to bye --->IP
phone (5000)
my case is:
1/ IP phone(5000) --->Opensips
2/ IVR number : 1001
3/ IP
2020 Oct 29
0
PJSIP tight loop on auth failure
Hi,
What if some fail2ban magic could keep OpenSIPs response from hitting
Asterisk after N attempts ?
Le mer. 28 oct. 2020 à 18:32, Kingsley Tart - Barritel Ltd <
kingsley.tart at barritel.com> a écrit :
> 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
2015 Jan 21
1
PJ SIP realtime with Kamailio / opensips
Hi all,
I saw Matt Jordan's recent Kamailio world talk and was interested in the
idea he proposed of stripping out authentication and registration from
asterisk and letting Kamailio handle it.
All of the tutorials I've seen (e.g. on asipto) show Kamailio forwarding
registrations to asterisk.
In order to do what Matt suggested would I be correct in assuming I would
have to use the
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
2010 Sep 25
0
Asterisk Cluster Scenario
Hello folks,
my company has experience in setting up single asterisk setup, but
recently one of our customers asked us to set up an asterisk cluster,
that must be High Availability and Load Balanced.
So I wrote here to have some hint or advice about the configuration we thought.
First of all I'll explain you the scenario:
The asterisk cluster must serve as Call Center
2009 Feb 26
1
incoming call problem
Dear All,
I have created an inbound context in SIP .conf that forward incoming call to
opensips server...The problem appears as soon as I enable t38pt_udptl =
yes...The Asterisk negotiate the SIP session with OpenSIPS without adding
voice codec to INVITE packet...It just contains T.38 protocol...When
t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with
OpenSIPS and cal