search for: cottonpickinmind

...K reply. However when I dial into extension 2001 nothing happens to the led's on the 360. Inspection of the 'sip trace log' on the 360's web page reveals that it does not receive any NOTIFY from asterisk. I am at my wits end - anybody got any ideas ? Paul HE ~ -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 --

Hello I see it is possible to buy Flash Disks up to 4GB now. Has anyone any experience of building an Asterisk system with a flash disk as the only storage device? Any brands you recommend? Is 2 or 4GB enough for an Asterisk installation? Typically how many MB is required for voicemail recording files for say a 10 user system? What about voicemail - I suppose files could be emailed and

...n ssh session to this customer never recovers when this happens. However the ADSL link does eventually recover but * never recovers the IAX trunk. I have to restart every time. Is there any way of forcing * to restart the link without having to restart asterisk? Regards Paul -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 --

I would like to know if somebody else experienced that: sip show peers will always drop the first character of the 11th line. while sip show peers like [0-9,a-z] will not drop any character. Can anybody test this, please? bye Ronald Wiplinger

...The Userfield is not defined by default. You have to set a define during compilation of asterisk - I do not know A@H but look in the files cdr/cdr_csv.c cdr/cdr-sqlite.c for USERFIELD and in /etc/asterisk/cdr_mysql.conf for userfield=1 Paul -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.dig...

hi all, I have a problem with @ 1.2.4 on debian kernel 2.6.8-2-386.: -- Executing Dial("SIP/2003-bbae", "zap/2/03460816149|30|t") in new stack Feb 14 17:25:25 WARNING[4246]: channel.c:2535 ast_request: No channel type registered for 'zap' Feb 14 17:25:25 NOTICE[4246]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'zap' (cause 66 - Channel

I have A@H 2.1, running * 1.2.1. I am trying to put information into the userfield with SetCDRUserField and AppendCDRUserField. However, the field is never populated in the cdr - I've checked the csv files and the MySQL asteriskcdrdb table. The field is defined in the MySQL table, but is always empty. The csv files that get created don't have a userfield at all, that is, there

Lilantha, the tones are supposed to be switched using the loadzone and defaultzone lines in /etc/zaptel.conf , and, progzone in /etc/asterisk/zapata.conf. The information about countries and frequencies/times are at zonedata.c located in the sourcecode of zaptel. As you may know, changing zonedata.c information requires a re-compilation of the zaptel module. Hope it helps, Ricardo

This is a wierd one. Can't figure it out. I have an SPA-3000 at the house handling my incoming line. It's setup to direct the incoming call to asterisk. Works great 99% of the time. A few times a day, I'm getting calls that ring once internally and are then hungup. I managed to get a detailed log [1] of what's happening today and it looks to me that the SPA is acting wierd.

Hi, everyone: I'm in the processing of deciding what IP phones we should use with our Asterisk server, and I wanted to get feedback from the user community on the quality, reliability and ease of operation of Snom phones. What do you have to say about these phones? Are there other phones you'd suggest along with or instead of Snom? Thanks, -Stephen-