[asterisk-users] Extensions routing

Greetings! I've been playing around with clustering some Asterisk servers for sake of fail-over and load balancing with DNS round-robin, and came to one problem. If I have, say, 2 servers, and clients register either on 1 or 2, how can I route extensions between them? I mean, if today user

Re: [asterisk-users] Extensions routing

The way I accomplish this is by having an active/passive cluster. The two or more servers have individual IP addresses and running heartbeat creates a clustered IP address. The active server uses the cluster IP address. If the active server should fail then the cluster IP address moves to

[asterisk-users] Slow AMI Originate

Hello, We use AMI to originate calls. Sometimes, lately every morning, the AMI Originate process operates extremely slowly. I cannot see the calls in core show channels verbose, I don't know where they are, what state they are in, after 2-3 minutes the calls go through one after the other. As

Re: [asterisk-users] Fwd: RTP stats explaination

Hi Dave, On Fri, May 18, 2012 at 11:27 PM, Dave Platt dpl...@radagast.org wrote: In our app we do not forward packet immediately. After enough packet received to increase rtp packetization time (ptime) the we forward the message over raw socket and set dscp to be 10 so that this time packets

Re: [asterisk-users] Extensions routing

On Saturday 19 May 2012, Mikhail Lischuk wrote: I've been playing around with clustering some Asterisk servers for sake of fail-over and load balancing with DNS round-robin, and came to one problem. If I have, say, 2 servers, and clients register either on 1 or 2, how can I route extensions

[asterisk-users] Testing for media?

I'm in the process of setting up an asterisk box that will stand between PBX's and the SIP providers. So a trunking server. How can I 'test' to see if this trunking server is stepping out of the media path during calls? Thanks David -- -- www.ringfree.biz 828-575-0030 --

Re: [asterisk-users] Testing for media?

tcpdump and wireshark would help I guess. Just sniff for sip traffic and look out for what's happening there. My 2 cents Sent from my iPhone On May 19, 2012, at 8:33 PM, David Wessell da...@ringfree.biz wrote: I'm in the process of setting up an asterisk box that will stand between PBX's and

Re: [asterisk-users] Best practices to route calls according holidays

Olivier wrote: Hi, At the moment, I'm mostly using a Day/Night toggle button to let users deal with week-ends, holidays and opening hours. As Asterisk 1.8 introduces Calendar capabilities, I'm wondering if better alternatives now exist. Is it possible, safe, reliable and easy to refer from

[asterisk-users] make and receive call from dial-up modem

is it possible to use data voice dial-up modem to make and receive call for ivr system.. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

[asterisk-users] Realtime peers and trunks coming from the same IP

I use an SBC to protect my pool of asterisk servers and as trunking endpoint with SIP Telcos. Now I'm trying to implement SIP phone registration to be delegated through the SBC, as a proxy. It doesn't work. It just works when I don't use realtime peers at the asterisk servers. Using realtime SIP

Re: [asterisk-users] make and receive call from dial-up modem

Mahendra Dobariya wrote: is it possible to use data voice dial-up modem to make and receive call for ivr system.. This should help: http://www.voip-info.org/wiki/view/X100P+clone Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety,

Re: [asterisk-users] make and receive call from dial-up modem

Hey Mahendra, We happen to still stock the same X100P cards in India. If you require one, do connect with me off the list. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India

[asterisk-users] SET SIP_CODEC and Video issues

Greetings List. I Have a small test server and i'm facing a small issue. i have setup two SIP PEERS and they are able to do Video calls. now I'm testing SET SIP_CODEC in a dial plan and when ever i'm setting the codec .. the inbound (=first) leg stops receiving or sending video and SIP SHOW

[asterisk-users] IAX2 passing back and forth variables

Hi all, I have two asterisk servers A and B. And I would like from A, dial to B passing some IAX variables. Then B handles the calls, setup some other variables that become available to A which can continue. So far, I have used IAXVAR function. It works when sending call from A to B But

Re: [asterisk-users] IAX2 passing back and forth variables

Uhm, if the dialplan is exactly as you pasted, you're not setting TESTVAR2 to anything. You would need some sort of Set(IAXVAR(TESTVAR2)=...) Noah From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi Sent: Saturday, May 19,

Re: [asterisk-users] IAX2 passing back and forth variables

Sorry, the dialplan is really on server B exten = s,n,Set(IAXVAR(TESTVAR2)=efgh) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah Engelberth Sent: 2012-05-19 14:45 To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] Configuring OpenVOX A400P issues

Thanks Vladimir for the response and apologies for my extremely late one! This week was quite crazy!! I have just upgraded my server to FreePBX to take advantage of an up-to-date system. I migrated the original config over with a few set changes. On 05/14/2012 04:42 AM, Vladimir Mikhelson

Re: [asterisk-users] SET SIP_CODEC and Video issues

Of course you are disabling the video maybe also include the video protocols in the sip_codec -Original Message- From: Tarek Sawah tareksa...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Sat, 19 May 2012 17:33:57 To: Asterisk Usersasterisk-users@lists.digium.com