Re: [asterisk-users] ChannelStateDesc: Ring ?

for the info. All the best, -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

Re: [asterisk-users] AMI 1.0 - 1.1 with originate.

Miguel Molina a écrit : Guillaume Yziquel escribió: So what is this permission issue? Where are the changes from 1.0 to 1.1 documented? When I was testing asterisk 1.6.0.X with the AMI Originate action, I fell into the same issue as you. I found that it was that the permissions now

[asterisk-users] AMI 1.0 - 1.1 with originate.

Message: Permission denied So what is this permission issue? Where are the changes from 1.0 to 1.1 documented? All the best, -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] ChannelStateDesc: Ring ?

gives me 'Ringing' but sometimes it only gives me 'Ring', which drives my application to 100% CPU (OK, not a very resilient app for now...) My question is: is 'Ring' a specific state for a channel, or is it a known bug in the AMI? All the best, -- Guillaume Yziquel http

[asterisk-users] 2 IPs for an Asterisk server.

in this setup. Is it possible to do such a thing with Asterisk? Does it need really special tweaking of Asterisk conf files? -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] 2 IPs for an Asterisk server.

work without any trouble Thank you for this valuable information. -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona

[asterisk-users] A in ACL of sip show peers.

the 'A' in the ACL field of the 'sip show peers' command might be. I've been unable to find this information on the net. Help would be appreciated. -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] Stop / Resume in Dialplan / AMI

asynchronous integration, and guidance would be appreciated. All the best, -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona

[asterisk-users] Incremented UniqueId

, -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] Incremented UniqueId

Steve Howes a écrit : On 4 Sep 2009, at 10:36, Guillaume Yziquel wrote: Uniqueid: asterisk-1252055630.26702 I'm really wondering how the Uniqueid works. Why is it incremented? What is the dot for in the Uniqueid? The bit before the dot is a unix timestamp (Fri, 04 Sep 2009 09:13:50 GMT

[asterisk-users] Originate calls with AMI.

Call from asterisk and not from 'yziquel'. Moreover when I pick up the phone it says You are now talking to asterisk, and then Zoiper closes the call immediately. There's surely something I do not get right here, and I'd appreciate some help. All the best, Guillaume YZiquel

Re: [asterisk-users] Originate calls with AMI.

priorities) in 1 context. =- Thanks a lot. -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net

[asterisk-users] Bottlenecks with my asterisk setup.

be the most common reasons behind this? Please feel free to ask for more relevant details. All the best, Guillaume Yziquel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona

Re: [asterisk-users] Bottlenecks with my asterisk setup.

Guillaume Yziquel a écrit : Hello. I've been seting up a small VoIP setup, with roughly 5 persons, doing essentially some Meetme conferences. People have been experiencing some quality problems with the sound. Essentially delay, and some tolerable echo. I'd appreciate advice on how

[asterisk-users] SIP server behind NAT

, or to documentation on this issue? Does this sound like a NAT issue? All the best, Guillaume Yziquel. Here's the beginning of the sip.conf file: [general] context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default

[asterisk-users] Outbound calls drop after 15 to 30 seconds.

this issue? All the best, Guillaume Yziquel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] Outbound calls drop after 15 to 30 seconds.

Steve Totaro a écrit : On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel guillaume.yziq...@citycable.ch wrote: Hello. I've set up and configured an Asterisk server to make SIP phone calls to external classic phones. However, it happens that after 15 or 30 seconds, the phone call drops