Re: [asterisk-users] asterisk.ctl file

Hi Shaun, thanks for the reply. This file is needed for debugging? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

[asterisk-users] RFC List

Hi There, Where can I get a complete set of RFCs and other specifications supported by Asterisk? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] RFC List

On 08/08/2012 06:30 AM, Kannan wrote: Where can I get a complete set of RFCs and other specifications supported by Asterisk? To my knowledge there is no such list. In addition, Asterisk (like many other pieces of software) does not claim 100% compliance with every RFC that is relevant, so

Re: [asterisk-users] Suggestion of Server Specifications for Asterisk

Let us know how does it performs... Leandro 2012/8/6 Shahid H shah...@gmail.com I have bought a new server today: i7-2600 CPU, 8GB and 2 x 256GB SSDs. 100Mbit Connection. I hope CPU is powerful enough for 200 concurrent calls. On Sun, Aug 5, 2012 at 1:57 AM, Michelle Dupuis

[asterisk-users] OT - Patton - FXO - Reduce incoming call delay

Hi, I'm benchmarking the performance of a Patton Smartnode 411X gateway. My setup is : GSM phone --PSTN-- SN411X --SIP-- Asterisk --SIP-- SIP phone My reference setup is: GSM phone --PSTN-- analog phone In the first case, it takes roughly 10s from the moment GSM user hits Send button to the

[asterisk-users] qualifysmoothing

Greetings list, I have a scenario where half a dozen phones at a site appear to be dropping offline for a few seconds every few hours, but the connection between them and the asterisk server remains up. It's been suggested to me that the problem might be to do with qualify - which is

[asterisk-users] alwaysauthreject=yes not working as expected

Asterisk 1.4.42 Set alwaysauthreject=yes in [general] section of sip.conf. Restarted asterisk However when I attempt to register I still get: [2012-08-08 21:11:34] NOTICE[15689] chan_sip.c: Registration from 'sip:000333082261...@domain.com' failed for '121.98.1.1' - Wrong password

Re: [asterisk-users] Block outbound calls based on IP address

Thanks for the reply however it is not possible to get the public IP address using the SIP_HEADER function (see my original post). We have many devices connecting from hundreds of dynamic external IPs. -- _ -- Bandwidth and

Re: [asterisk-users] alwaysauthreject=yes not working as expected

Asterisk 1.4.42 Set alwaysauthreject=yes in [general] section of sip.conf. Restarted asterisk However when I attempt to register I still get: [2012-08-08 21:11:34] NOTICE[15689] chan_sip.c: Registration from 'sip:000333082261...@domain.com' failed for '121.98.1.1' - Wrong password

[asterisk-users] tls is up but no audio

Hi All, I'm headbanging on this from a couple of days, begging here for some help :) I'm configuring tls on asterisk for the first time to experiment with an open (public) service idea about having asterisk accepting any sip user (with the sip.conf option 'autocreatepeer=yes') and call each

[asterisk-users] IAX with two asterisk boxes

Hello, I have two asterisk boxes running and both are using DAHDI PRI Card. I wish to know if IAX is the best method to connect both the boxes? Also, need some help with the following? 1. For incoming call on server2 I wish to run an IVR to the user for which all my prompt sound files resides