Hello,
we are finally going to redesign our Asterisk-Setup, which has grown
quite complex. We have five sites with a total of 400 users, 15 SIP
registrations and 3 IAX registrations. We do not use any
VoIP-hardware, so it's all software-based. But we make heavy use of
features, including
Can Asterisk do virtual hosting? While I want/need the sites to be
hosted by the same instance (so that e.g. calls can be transferred
easily), I don't want to have to name my peers [site1-john], and
I want people to be able to SIP-dial j...@site1.example.org and
j...@site2.example.org and trust
I am using 1.4.43 currently.
I am using the AMI to originate a call over a SIP Trunk to my cell
XXX506. works fine.
when the call is active I do a core show channels concise and I get:
SIP/testsystem-0ad0!smvoice-dialout!callprogress!4!Up!AGI!smvoice!0!!3!24!(None)
My AGI is called
On 11/06/2012 09:45 PM, Michael L. Young wrote:
- Original Message -
From: sean darcy seandar...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 6, 2012 7:51:04 PM
Subject: [asterisk-users] 11.0.1: more sip registry woes
Upgrade to 11. This worked on 10.X.X
Lionel BEAUDOIN wrote:
Hello,
Hola,
I saw your email in a forum message, can you help me, I try to use
SIPML5 with an Asterisk 11 server ?
My Asterisk server is installed on a Debian server.
I have download all the sources from sipml5.org
Please ensure you have followed the instructions
Since you're using sip, use sip show channels and pick the call-id from
there.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Wednesday, November 07, 2012 7:27 AM
To: Asterisk Users Mailing
- Original Message -
From: sean darcy seandar...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Wednesday, November 7, 2012 9:20:58 AM
Subject: Re: [asterisk-users] 11.0.1: more sip registry woes
On 11/06/2012 09:45 PM, Michael L. Young wrote:
- Original Message -
On Wed, Nov 7, 2012 at 7:27 AM, Jerry Geis ge...@pagestation.com wrote:
I am using 1.4.43 currently.
I am using the AMI to originate a call over a SIP Trunk to my cell
XXX506. works fine.
when the call is active I do a core show channels concise and I get:
snip
How do I lookup my
martin f krafft wrote:
Can Asterisk do virtual hosting? While I want/need the sites to be
hosted by the same instance (so that e.g. calls can be transferred
easily), I don't want to have to name my peers [site1-john], and
I want people to be able to SIP-dial j...@site1.example.org and
also sprach Joshua Colp jc...@digium.com [2012.11.07.1831 +0100]:
Peer names have to be distinct, this is just a fundamental design
element of chan_sip. What a lot of people end up doing is instead of
treating peers as people they treat them as devices. The peer name
becomes the MAC address of
Dear list,
we would really like to be able to invite a third and fourth party
to our current one-on-one call. At the moment, we have to agree to
dial into MeetMe 10 minutes later, then make calls to the third
parties, and hope it all works out.
I have found a couple of examples on the Internet
martin f krafft wrote:
also sprach Joshua Colpjc...@digium.com [2012.11.07.1831 +0100]:
Peer names have to be distinct, this is just a fundamental design
element of chan_sip. What a lot of people end up doing is instead of
treating peers as people they treat them as devices. The peer name
On 11/07/2012 01:06 PM, Joshua Colp wrote:
martin f krafft wrote:
also sprach Joshua Colpjc...@digium.com [2012.11.07.1831 +0100]:
Peer names have to be distinct, this is just a fundamental design
element of chan_sip. What a lot of people end up doing is instead of
treating peers as people
With our Polycom 501 phones we can put a third and fourth party on our
two-way call without using the meetme app. You just hit the conference
button on the phone during a call and add your third person and do the same
for your fourth person.
-Original Message-
From:
On 11/7/2012 2:01 PM, martin f krafft wrote:
Dear list,
we would really like to be able to invite a third and fourth party
to our current one-on-one call. At the moment, we have to agree to
dial into MeetMe 10 minutes later, then make calls to the third
parties, and hope it all works out.
I
I use the ChannelRedirect function to redirect the desired channel to the meetme roomChristian SavinovichVoIP Telephony Consultant646-982-3572
Original Message
Subject: Re: [asterisk-users] Impromptu conferencing
From: James Sharp ja...@fivecats.org
Date: Wed, November 07,
2012-11-07 20:49, Jeff LaCoursiere skrev:
Just to chime in, if you REALLY want multi-tenant, it is super easy and
surprisingly efficient to use kernel level virtualization to run
multiple instances of asterisk (and even FreePBX). We use LXC to do
this. The host runs an instance that has the
Hello,
I have noticed some occasional one way audio on a specific sub set of calls
in my system. First, let me be more specific about what I mean about occasional
one way audio. Unlike most of the posts I've seen (where the end fix was either
NAT'ing or RTP issues) the calls in question
I experience random crash of machine (full hang, requiring a hard reset)
after trying to test run Asterisk 11.
The machine is a centos 5.8 32 bits pc with 1G ram. Asterisk is compiled
from the source and no other software has been installed
Anyone experience similar situation?
--
On 12-11-07 05:41 AM, martin f krafft wrote:
Hello,
we are finally going to redesign our Asterisk-Setup, which has grown
quite complex. We have five sites with a total of 400 users, 15 SIP
registrations and 3 IAX registrations. We do not use any
VoIP-hardware, so it's all software-based. But we
On 11/07/2012 02:16 PM, Johan Wilfer wrote:
2012-11-07 20:49, Jeff LaCoursiere skrev:
Just to chime in, if you REALLY want multi-tenant, it is super easy and
surprisingly efficient to use kernel level virtualization to run
multiple instances of asterisk (and even FreePBX). We use LXC to do
On 11/07/2012 05:20 PM, Jeff LaCoursiere wrote:
On 11/07/2012 02:16 PM, Johan Wilfer wrote:
2012-11-07 20:49, Jeff LaCoursiere skrev:
Just to chime in, if you REALLY want multi-tenant, it is super easy and
surprisingly efficient to use kernel level virtualization to run
multiple instances of
I just installed a TE820 octal span T1 card, and it's not showing up in
dahdi_hardware output. This was installed into a test machine that already has
a TDM800P card in it, and that one is showing up and working fine. Is there
some kernel module that I'm missing?
Lspci:
05:04.0 Ethernet
On Wed, Nov 07, 2012 at 05:02:57PM -0800, Justin Killen wrote:
I just installed a TE820 octal span T1 card, and it's not showing
up in dahdi_hardware output. This was installed into a test
machine that already has a TDM800P card in it, and that one is
showing up and working fine. Is there
Thanks for your help! Do you know if there's a way to read the sip
debug messages without opening the log file on the disk, such as
through AMI?
On Tue, Nov 6, 2012 at 3:59 PM, Danny Nicholas da...@debsinc.com wrote:
I would recommend two things. Number one would be to tweak your logger.conf
are you running dahdi ?
We're using 11, System uptime: 3 weeks, 22 hours, 42 minutes, 19
seconds, 231452 calls processed
We did, however, have a problem with dahdi freezing the machine
Julian
On 7 November 2012 22:32, asterisk asterisk aster...@ck-lee.com wrote:
I experience random crash of
Hello all,
I am going to register asterisk sip users through active directory accounts
LDAP (that is a separated server with ip : 192.168.11.17)
So I have followed the below link as well:
https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver
Hello,
thanks for your reply.
No , the daddi is not running on my asterisk server,
Do you think it is necessary ? and the problem on LDAP is associate with dahdi?
From: Julian Lyndon-Smith aster...@dotr.com
To: Asterisk Users Mailing List - Non-Commercial
No, I put it in Xen VPS with Centos 5.8. Only things I added are skype
support using siptosis and java.
Asterisk 11 is complied with no issue, siptosis and skype call no issues.
But hangs unexpectedly.
Any clue is welcome?
On Thu, Nov 8, 2012 at 2:10 PM, Julian Lyndon-Smith
No, we removed dahdi (some hardware issues) and not had a problem since.
No idea on the ldap side (you never mentioned ldap at all)
On 8 November 2012 06:22, Samira Hosseini samiramhosse...@yahoo.com wrote:
Hello,
thanks for your reply.
No , the daddi is not running on my asterisk server,
Do
also sprach Paul Belanger paul.belan...@polybeacon.com [2012.11.07.2340
+0100]:
What is your point of pain? Right now we do most of the
configuration, provisioning, and system management outside of
asterisk.
My systems are already managed automatically, thankfully no longer
with Puppet. ;)
I
What about just setting up a database which stores your data however you
want then generate static files from that data or creating views for
realtime (where appropriate)?
That's how I do it with my company's system.
To keep things not so complicated, I have AGI scripts. Keeps things clean
and
also sprach Logan Bibby lo...@keobi.com [2012.11.08.0747 +0100]:
What about just setting up a database which stores your data
however you want then generate static files from that data or
creating views for realtime (where appropriate)?
Sure, I could do that. First, however, I would like to
2012/11/7 Jeff LaCoursiere j...@sunfone.com
Just to chime in, if you REALLY want multi-tenant, it is super easy and
surprisingly efficient to use kernel level virtualization to run multiple
instances of asterisk (and even FreePBX). We use LXC to do this. The
host runs an instance that has
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