Displaying 20 results from an estimated 3000 matches similar to: "Access PBX from internet - best practice"
2014 Jan 14
2
Asterisk QOS
I asked this on the list over the weekend, and likely missed a few people inboxes.
I'm having a problem pulling data from RTPAUDIOQOS. For testing purposes I have asterisk sending QOS data to the console. It seems I get QOS data only if the caller hangs up, with the variable being empty if the callee (or asterisk) hangs up.
Any idea why I would see this?
exten =>
2014 Jan 31
1
Asterisk as a media gateway
I'm playing around in a lab, and I was wondering if its possible to have Asterisk act similar to that of a Avaya PBX, where we have media gateways do the heavy lifting.
This is what I was thinking of trying.
1. One asterisk server will contain the logic of the phone system (ex: queues, extensions...etc).
2. The mains server will not handle RTP traffic, it will send the RTP traffic to
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
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
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
<body bgcolor="#ffffff" background="https://img.web.de/v/p.gif" class="bgRepeatYes" style="background-repeat: repeat; ; background-color: rgb(255, 255, 255); color: rgb(0, 0, 0); font-family: verdana,geneva; font-size: 9pt; padding-left: 0px;"><div style="min-height: 200px; background-image: url(https://img.web.de/v/p.gif); background-repeat:
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
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
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
2012 Jun 22
2
SIP over SSL TCP or SRTP?
Hello,
Which one of these ensures that SIP packets are sent and received in a
secure format so that users using public wifi don't allow MITM type of
attacks or others can't read the plaintext SIP packet info. VPN is not an
option. Looking for 2nd most secure to VPN.
P.S. Are both options part of the configs of Asterisk or need modules to be
selected and installed before doing the
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 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
2009 Mar 01
1
Help T.38
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
under General context...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
2009 Feb 11
2
OPTIONS packets
Hi all,
I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is
OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy
is not replying back...The issue is the UNKNOWN username that reside in the
OPTIONS packet as you can see in the captured packets as you can see below:
1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060
2. OPTIONS sip:OPENSIPS_IP
2009 Dec 30
2
multiple instances of asterisk on same machine
hi all,
I have a little problem.... I'm using asterisk with opensips as opensips
dispatches calls to asterisk. I have to use multiple asterisk servers but
since for the time being im using 1 machine for testing i want to run
different instances of asterisk running on 1 pc listening to different
ports. Can someone please guide me how to do this? I'll be very thankful
Regards,
Saeed
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
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
2009 Feb 24
1
Incoming call
Dera All,
I have the following scenario,
A customer dial a DID number...The call is routed to a PSTN GW that send the
call to asterisk...
On asterisk I created an AGI Script that send the call to an extension
registered on OpenSIPS server...
The extension is ringing successfully, but as soon as I accept the call on
OpenSIPS side the call is hangd up...
I checked rhe SIP debug and it seems that I
2020 Jan 29
2
Centos 7: UPD packet checksum verification?
On Wednesday, January 29, 2020 6:52:50 PM CET Nataraj wrote:
[...]
> By burst, I mean that you don't have a bandwidth commitment with an SLA
> from your provider. A bandwidth commitment means that you are paying a
> provider to guarantee you so many MB or GB of bandwidth and this is
> guaranteed to you. This means it is allocated to you in their network
> allotments and you