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Message: 6 Date: Fri, 03 Mar 2006 17:32:47 +0000 From: Conrad Wood <asterisk-users@conradwood.netasterisk-users@conradwood.netasterisk-users@conradwood.netasterisk-users@conradwood.net> Subject: Re: [Asterisk-Users] Problem with NAT!!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <1141407168.18488.10.camel@trivial.lem...

Does anyone have any recommendations on how to connect 160 analogue phones to an asterisk PBX? Background information: A client wishes to replace their current PBX with a new VoIP system. Currently they have 2 PRIs. I intent to set up 2 asterisk PBXs with Debian GNU/Linux on raided drives. These drives will be mounted only read-only to recover gracefully from power-cycles. I am considering 2

Hi everyone, I get these events sent like this: Event: ParkedCall Privilege: call,all Exten: 701 Channel: Zap/4-1 From: IAX2/cnw-4 Timeout: 120 CallerID: XXXXXXXXX CallerIDName: Conrad Wood Unqiueid: asterisk-1713-1139266402.909 ^^^^^^^^^ Please note the spelling of uniqueid. I find the spelling in res_features.c - but only once I patched it with bristuff patches. Does anyone know whether that

Hello What soundfile format, is the one that uses least transcoding during playback? As I can see, I can choose wav or gsm. What sucks least cpu power, during playback to example a Zap channel? I would guess wav, but is this correct? Kind Regards Jon Leren Sch?pzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.11/496 -

Hi I wonder whether anyone got the Sipura ata 3000 to decode British Telecoms callerid and pass it to asterisk? The userguide seems to suggest that this is not possible, is that right? Conrad

Hi List Is someone out there using one or more GSMgateway(s) from CyberTelecom ? Me and some friends are interested in buying some of them, but before we would like to ask, how the experiences are others have made. e.g. How easy to setup ? How reliable ? How's the voice quality ? etc. Any input/feedback is welcome. Greets Adibar --

Hi everyone, I'm trying to get early dial to work with our grandstream 100 phones. The phones use SIP, asterisk is 1-0-5 on debian GNU/Linux (sarge). Outside connections are via 2 ISDN BRI (British Telecom) lines using 2 billion isdn cards and bristuff. The phones are set up to be in context [internalphone]. I numbered all the internal extension with _6XX That works well with early dial.

Hi everyone. I see many posts complaining about Grandstream equipment, so I thought I tell the other side of the story as well. We use Grandstream Budgetone 101 phones in our office and they work extremely well. We have no echo, no crashes, no sudden resets, they "just work". We provision them using TFTP. Our asterisk box has 2 HFCPCI ISDN BRI cards. It's running asterisk 1.2.0 on

On my asterisk machines the following features.conf file crashes asterisk (core dump) This happens with 1.2.4, 1.2.10, 1.2.12, with or without bristuff. It's easy to work around, but broken nevertheless. Has anyone else experienced that or is it just me? ;) -------- /etc/asterisk/features.conf ---------------- [applicationmap] # THIS CRASHES asterisk: rateone => #1,caller,Macro,ratecall,1

A client used to use BT isdn30 and ported the numbers to telewest several years ago. Now, the client moved to adept telecom. I *think* adept resells BT products. We got new numbers from adept (bt?) and the old pbx on the telewest lines forwards the calls to the new numbers. On the adept line I got a digium card in an opteron supermicro server. ztcfg gives me over 99.99 pretty much all the time.