Oh Dear..... You are moving into a world of hurt. ... systems ... would ... don't need ... FXO/FXS ... added ... which ... number maps ... want ... upgrading...
Yeah, but think of how much experience (read: he suffers and we get the benefit) he'll gain from this. Besides, telco/pabx is all about masochistic suffering -...
LOL Yes I can see what you mean..I look at it this way, on the job training paid for by my employer..well at least I get to play with new stuff and learn...
Hey all, I'm using asterisk@home v1.5 and wondered if it was possible to divert calls to a mobile say between the hours of 9-5. Now, I know how to divert to an...
Hi Dave. gotoiftime is your friend here. create two contexts and jump between them using gotoiftime. cheers, Mark ... -- regards, Mark P. Edwards FWD: 667917...
Even Easier.... You are using @home... setup a ring group with your mobile Number follwed by # IE Ring Group 200 0488256256# then in incomming calls, set you...
Hi list, Lately I've been through the SPA-300 hangup detection nightmare some other users experimented with it. I configured my disconnect tone to: ...
What value do you have to change to get more time to dial before you get an off hook signal. I am using asterisk@home with sipuras attached.Is this a value I...
... This will be in the ATAs config 1) Go to http://SPA IP/admin/advanced 2) Click on Regional 3) increase the value for: Interdigit Short Timer The value is...
Hello everyone, I want to use asterisk + sipura as PSTN line. When PSTN line is ringing sipura calls asterisk's extention which transfer it to sip extentions. ...
Hey, Just done it - easy as... The email from digium with your code takes a while... Look here for instructions. Only thing I can add (not being a linux pro...
Hi Guys, I have just signed up with Exetel for my DSL so I thought I'd give their VoIP a go. I have and AAH box and have config the SIP trunk, using any of the...
Incoming sip calls are given congenstion by default I think (you should be able to see this in sip.conf), I cant remember what settings I changed, but I think...
Yeah, This is what I have, I'm using asterisk@home too.. ; Note: If your SIP devices are behind a NAT and your Asterisk ; server isn't, try adding "nat=1" to...
Is there anyway with Asterisk to dial through numbers originating from an internal PBX extension. e.g. I dial 51XXX 51 is handled by Asterisk to go to the...
You can have something like this: [def-context] exten => 51,1,DISA(no-password|dialtone-context) [dialtone] exten => _XXXX.,1,Dial(SIP/isp/${EXTEN}) I'm...
Cool thanks for that has given me something to go and try. ... From: asterisk-anz@yahoogroups.com [mailto:asterisk-anz@yahoogroups.com] On Behalf Of Julio...
... How about: exten => _51XXX,1,Dial(ZAP/g1/{EXTEN:2}) Then you don't have to wait for the 2nd dialtone asterisk will pass the xxx for you out the FXO. Matt....
Hi all, I'm struggling with getting the volume levels correct on my two TDM400P cards with 4 FXO ports each... I found a howto calibrate them using the...
And you won't find a Milliwatt test number either .... Closely Guarded Telstra secret (and equipment hire charge) Quick and dirty is with the aid of ztmonitor ...
Hi, Is it possible to connect 2 asterisk servers if both of them are after firewall, but one firewall has port forwarded to asterisk? How can it be done? ...
using IAX it would be possible. - Dave strelnikov_michael wrote: Hi, Is it possible to connect 2 asterisk servers if both of them are after firewall, but one...
Has anyone had any experience with a 8+ port FXS gateway such as the soundwin s800 or one of the other similar products out there? I havn't been able to find...
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