Re: [asterisk-users] Managing complex setups with Asterisk

2012/11/7 Jeff LaCoursiere > >> 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 dahd

Re: [asterisk-users] Managing complex setups with Asterisk

also sprach Logan Bibby [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 keep scouting f

Re: [asterisk-users] Managing complex setups with Asterisk

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 is

Re: [asterisk-users] Managing complex setups with Asterisk

also sprach Paul Belanger [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 am only talking about con

Re: [asterisk-users] Random crash of the machine ? due to Asterisk 11

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 wrote: > Hello, > thanks for your reply. > No , the daddi is not running on my asterisk server, > Do you think it is neces

Re: [asterisk-users] Random crash of the machine ? due to Asterisk 11

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 wrote: > are you run

Re: [asterisk-users] Random crash of the machine ? due to Asterisk 11

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 >To: Asterisk Users Mailing List - Non-Commercial Discussion > >Sent

[asterisk-users] (problem in Integrate asterisk through LDAP (Invalid credential

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 http://www.asteriskdocs.org/en/3rd_Edition/

Re: [asterisk-users] Random crash of the machine ? due to Asterisk 11

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 wrote: > I experience random crash of machine (full hang,

Re: [asterisk-users] How to determine if AMI MessageSend succeeded?

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 wrote: > I would recommend two things. Number one would be to tweak your logger.conf > to separate out e

Re: [asterisk-users] TE820 hardware detection

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

[asterisk-users] TE820 hardware detection

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 cont

Re: [asterisk-users] Managing complex setups with Asterisk

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 as

Re: [asterisk-users] Managing complex setups with Asterisk

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 this

Re: [asterisk-users] Managing complex setups with Asterisk

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

[asterisk-users] Random crash of the machine ? due to Asterisk 11

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? --

[asterisk-users] Occasional one way audio.

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 will

Re: [asterisk-users] Managing complex setups with Asterisk

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

Re: [asterisk-users] Impromptu conferencing

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 Date: Wed, November 07,

Re: [asterisk-users] Impromptu conferencing

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 h

Re: [asterisk-users] Impromptu conferencing

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: asterisk-users-boun.

Re: [asterisk-users] Managing complex setups with Asterisk

On 11/07/2012 01:06 PM, Joshua Colp wrote: martin f krafft wrote: also sprach Joshua Colp [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

Re: [asterisk-users] Managing complex setups with Asterisk

martin f krafft wrote: also sprach Joshua Colp [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 addre

[asterisk-users] Impromptu conferencing

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

Re: [asterisk-users] Managing complex setups with Asterisk

also sprach Joshua Colp [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 the device t

Re: [asterisk-users] Managing complex setups with Asterisk

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 j...@site2.

Re: [asterisk-users] how to lookup a call

On Wed, Nov 7, 2012 at 7:27 AM, Jerry Geis 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: > > How do I "lookup" my call so I can "ha

Re: [asterisk-users] 11.0.1: more sip registry woes

- Original Message - > From: "sean darcy" > 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 - > >> From: "se

Re: [asterisk-users] how to lookup a call

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 Lis

Re: [asterisk-users] Can you help me to use SIPML5 with Asterisk ?

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 at

Re: [asterisk-users] 11.0.1: more sip registry woes

On 11/06/2012 09:45 PM, Michael L. Young wrote: - Original Message - From: "sean darcy" 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 sip.conf: register=>:@n

[asterisk-users] how to lookup a call

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

Re: [asterisk-users] Managing complex setups with Asterisk

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 tha

[asterisk-users] Managing complex setups with Asterisk

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 voicemail