Displaying 20 results from an estimated 6000 matches similar to: "Asterisk as register server through OpenSIPS"
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 Aug 25
6
Breaking news, but what happened? 11.000 channels on one server
Hello Asterisk users around the world!
Recently, I have been working with pretty large Asterisk
installations. 300 servers running Asterisk and Kamailio (OpenSER).
Replacing large Nortel systems with just a few tiny boxes and other
interesting solutions. Testing has been a large part of these
projects. How much can we put into one Asterisk box? Calls per euro
invested matters.
So far,
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 :)
2010 Apr 19
2
OpenSIPS with Asterisk Backend
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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
2007 Apr 02
1
603 Error
Hi Guys,
I started getting this error only from one of our ITSP's once we upgraded from 1.2.16 to 1.2.17.
Can anyone shed light ?
--- (12 headers 0 lines) ---
Transmitting (NAT) to 209.212.93.53:5060:
SIP/2.0 603 Declined (no dialog)
Via: SIP/2.0/UDP XXX.XXX.XX.XX;branch=z9hG4bKf928.2b3b0de5.0;received=XXX.XXX.XX.XXX
Via: SIP/2.0/UDP
2008 Mar 28
2
wrong extension status when call-limit=1 is used
Without call-limit defined, when a sip extension calls
another sip extension then "show hints" shows that
both are InUse (as expected). When one of them hangs
up, both hints status become "Idle" (as expected).
With call-limit=1 for each SIP extension:
the caller is always Idle while the callee is InUse.
Is this behavior normal?
Doesn't sound right because if, during the
2009 Sep 04
1
Send 200 OK with SDP instead of 183 with SDP when ringing starts
Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through
which I connected an external PSTN line. I use it as carrier for VoIP
calls. I can make successfully calls, but there's one problem, I receive
200 OK with SDP with delay (sometimes more than 30 seconds).
So when I make a call through asterisk I receive intially:
- 100 Trying
- 183 Session Progress, with SDP
when the called
2010 Feb 17
3
sip.conf - sort order, does it matter
Does the sort order matter in sip.conf file?
I know sort order might effect:
allow=ulaw
allow=alaw
but does it matter where I place: insecure=invite ?
The reason I'm asking is that I've loaded almost two identical (sip.conf and extension.conf) files on the same asterisk server and with one set
insecure=invite is working correctly.
When I load the second set of dial plan (sip.conf and
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
2007 Apr 02
1
SIP 484 (Early Dial) and International Dialing
I'm building a dialplan for use with a bunch of GXP2000 desk sets. During
testing, we had some user issues surrounding the lack of an on-phone
dialplan. Users would hit 9 and sit there waiting for a redial tone, and
the GXP would time out, sending just '9' to *, which couldn't do much other
than spit back a 404 or play pbx-invalid.
I turned on the "early dial" option
2010 Mar 17
3
SIP codec negotiation / manipulation
We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation.
We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second
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?
2011 Nov 16
1
Server-to-server BLF
Hi all,
Do you have an idea on the best way on how to implement a system with
multiple Asterisk servers with BLF working in such a way that a peer on one
server can subscribe to another peer on the other server in a seamless
manner? Has anyone set-up a system like this before?
Thanks!
Regards,
Ronald
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2006 Apr 05
5
Dial Plan Logic Problem
Hi
I can't for the life of me work out why this is not
working. When in the campon contect if you hit a DTMF
key 2 you get moved to the exten => 2 defined in the
mainmenu context not the exten => 2 defined in the
campon context. What is wrong? The same happens if you
hit key 1.
[campon]
exten => _*1XXX,1,Answer
exten => _*1XXX,2,SetCallerID(${CALLERIDNUM})
exten =>
2004 Jul 05
2
Again Sip Registration Fail
Recently I wrote about this problem,
but it still exist and I can't dial my Xlite SIP Phone
So here is the Notice
Jul 5 17:14:07 NOTICE[65541]: chan_sip.c:6731 handle_request:
Registration from 'Damian Minkov <sip:damian@10.1.1.2>' failed for
'10.1.1.11'
The * box(10.1.1.2) and the PC(10.1.1.11) on which is the XLite are in
the same network
Here is part from sip
2009 Sep 03
3
GTalk functionality Asterisk
Hello
Previous context :- After Looking up sip and IAX2 that require
configuration at router level which may cause some problems like connection
break etc... so i left them ......... and start wondering if there is some
thing that dont require configuration at router layer. The task to
accomplish to make and recieve calls from outside local network using any
protocol whose soft phones are
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.