Displaying 20 results from an estimated 1200 matches similar to: "OpenSIPS on CentOS"
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
2011 May 11
2
Asterisk SIP Trunking with Cisco UC 560
Hello,
I'm interested in knowing if anyone out there has successfully connected
Asterisk to a Cisco UC 560 via SIP trunking? We have a client of ours that
we put in an Asterisk install, one of their sister companies who we don't
control is putting in a Cisco UC 560. From my looking I think it can be
done, but the vendor is telling them it can't. Thought I'd ask around here
and see
2009 Oct 06
2
Transfers from Queue Calls
Hello,
I thought to post this here before my manager starts his own coding
project to give us a workaround. My situation I'm running into is as
follows:
1. A call comes into our Asterisk system, it's trunked from one office
to another via IAX.
2. Call enters a queue and is picked up by one of the agents.
3. That agent has to transfer the call, could be for a number of
reasons the client
2009 May 26
2
Converting Cisco 7961 to SIP
As part of a project to move a clients Cisco phones to SIP to work with an Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk setup. Now, I've gotten the firmware files from the site, the latest version is 8.4 which contains the following files:
apps41.8-4-3-16.sbn
cnu41.8-4-3-16.sbn
cvm41sip.8-4-3-16.sbn
dsp41.8-4-3-16.sbn
jar41sip.8-4-3-16.sbn
2009 Jun 18
2
Voicemail Password
Does anyone know of a way to force the voicemail password for users to be of a certain length? We've setup operator=yes within our voicemail.conf and want to have the users use a long password to prevent possible guessing by external parties. I'm not seeing any such option in my research. If it doesn't exist it might be a decent feature. Thanks.
Running: 1.4.25, on CentOS 4.7
2009 Mar 12
2
Timeout for Queue
Hello,
We had an incident recently where a call was in queue for an extended period of time. We use queuemetrics for reporting, and it reports that the call was waiting for 20 minutes. The different thing about it is that the disconnect reason is stated as Timeout. Is there a set maximum time a call will wait in the queue before being automatically disconnected? I tried looking through the code
2009 Jan 08
1
Goto Question
Hello,
I'm running three asterisk boxes, spread across three different countries. One of the offices is running Asterisk 1.2.18 on the Druid Telephony Platform(not my choice, has been in before I started and haven't had the time to remove it).
My situation I have is based on the contexts already in place, particularly for outbound calls, I need to do a Goto sending the call back into 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 :)
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
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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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
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 Jun 10
1
External PRI Appliance
Hello,
I'm supporting an Asterisk setup in the Cayman Islands(I'm in Canada), which is running a PRI to a local telecom provider. We are looking at improving the setup, setting up high availability etc. My manager is interested in putting a TDMOE device in place, so we can easily switch the line remotely. He's looking at the following device:
2009 Jan 20
1
Setting up an outgoing trunk group
Hi All,
I'm confused! My Asterisk system has a Zap trunk and three SIP trunks.
I'd like to configure the dialplan to route via the first trunk in a
list and if that's not available or it's busy, fall over to the
second, then to the third, etc.
AIUI Dial(Zap/1&SIP/out1&SIP/out2/${EXTEN}) rings all the trunks in
the list and bridges to the first to answer. Unfortunately,
2006 Oct 03
1
a domain VTx with the VNIF does hang.
Hi all, my name is Hirofumi Tsujimura.
We are porting and testing a PV-on-HVM in the IPF.
This is a first time to send mail.
I probably found the problem when I tested the VNIF.
My operation for the test is following.
1. create a domain VTx and attach the VNIF in it.
2. create a domain U.
3. send a packet to the domain VTx from the domain U with ping
command.
Then, the domain VTx
2009 Apr 01
2
Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten => s,n,ResetCDR(vw)
exten => s,n,NoCDR()
So I retrieve
2009 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,
2013 Feb 02
3
Running XEN 4.1 64 bit on Ubuntu desktop without VTx
Dear all,
I am quite new in this virtualization area. I am want to do some experiment
with live migration using xen. However, I got problem since my server
didn''t support VTx. I am using Ubuntu desktop 12.04 64 bit with Xen 4.1
Amd64. But when I reload the machine it wont start, since the XEN website
its doesn''t matter using Paravirtualization without VTx support I dont know
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