Displaying 20 results from an estimated 1000 matches similar to: "Asterisk + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk"
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
2009 Oct 23
2
How to generate 183 Session Progress
Hello everybody,
I have 2 users connected on the same Asterisk server that are connected with 2 different Asterisk servers.
For outgoing calls, one in receiving 183 Session Progress and the other not! Do you have any idea why?
Thanks.
I have tried to understand by myself and in their INVITE they have almost the same Allow and Supported SIP Headers
The one that works:
Allow: INVITE, ACK,
2009 Apr 01
2
Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten => s,n,ResetCDR(vw)
exten => s,n,NoCDR()
So I retrieve
2009 Jun 12
1
Asterisk + TC400B - Clock Trouble
Hello all, I have a TC400B Digium card in order to deal with transcoding and
I have some trouble using it, I have a timer synchronisation problem!
I would be very grateful if you have any idea to help me?
It seems that the card is not correctly synchronised to the system because
when I speak to one side, the sound takes 5 seconds to go to the other side,
and increasing, after 30 seconds of call,
2009 Jul 14
2
Asterisk 1.4.26 final release - What is blocking?
Hello everybody,
I was wondering what is postponing the 1.4.26 release? I thought it was
scheculed for last week.
Is there something we can do to help to release this version?
There is no more issue reported on https://issues.asterisk.org/ for the time
being.
Best Regards,
-- --
Marc LEURENT
lftsy at leurent.eu
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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
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 :)
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
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2009 Sep 09
1
CLI file convert from alaw to g729 with TC400B transcoding card results to an empty file
Good afternoon,
I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card.
The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty
file as you can see below...
CLI> file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw
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
2010 Apr 19
2
OpenSIPS with Asterisk Backend
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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
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.
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
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?
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,
>
>
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 Jun 21
1
Unable to connect to CIFS host
Hello,
I'm using samba 3.5.11 to connect a Windows 2003 Active Directory.
With cups, samba is an part of a print server used to print to windows
desktop shared printers.
DNS are Active Directory Integrated.
Network is both IPV4 and IPV6, IPV6 for Linux and Windows Vista and above.
Some times, some users are not able to print. In logs of cups, I see to
thinks "Unable to connect CIFS
2007 Nov 13
1
route INVITE sip:s@sip.test.fr
Good evening!
I was wondering one thing,
I'm using freepbx to configure my asterisk server and I have a problem
with some inbound calls.
When I receive a call to an INVITE sip:01xxxxxx at myip.com I an set an
inbound route! It matches a DID number.
How can I route an INVITE sip:s at myip.com? The number only appear in the
To: Section.
Thanks!
Example:
With this one, I cannot route it
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