similar to: Re: Match a Numer - then continue with, dialplan

> -----Original Message----- > From: Tony Mountifield [mailto:tony@softins.clara.co.uk] > Sent: Wednesday, December 20, 2006 2:41 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > In article > <645FEC31A18FE54A8721500CDD55A7B6035D0C6C@mail.oneeighty.com>, > Douglas Garstang

> -----Original Message----- > From: David Gomillion [mailto:dgomillion@eyecarenow.com] > Sent: Wednesday, December 20, 2006 10:27 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > I think you're making it far too difficult. > > What I do is something like this: > > [outgoing]

Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Doug.

> -----Original Message----- > From: Richard Lyman [mailto:pchammer@dynx.net] > Sent: Wednesday, December 20, 2006 4:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > Douglas Garstang wrote: > >> -----Original Message----- > >> From: David

> -----Original Message----- > From: Richard Lyman [mailto:pchammer@dynx.net] > Sent: Wednesday, December 20, 2006 4:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > Douglas Garstang wrote: > >> -----Original Message----- > >> From: David

I think you're making it far too difficult. What I do is something like this: [outgoing] include => internal include => longdistance ;Always include internal first, as matches from the first include ;will be used first. This allows you to make sure your internal ;extensions don't go out your trunks. [longdistance] ignorepat => 9; include => default; already included from

On Fri, 2006-12-08 at 04:26 -0700, Douglas Garstang wrote: > Hi Steve. > > Thanks, but unfortunately, I can't be involved in that. We are > running Asterisk in a production environment and we're using > 1.2, not 1.4. I don't have the resources to work with 1.4. > Last time I filed a bug against 1.2 I got told off. >

Does anyone know if realtime extensions allows extensions in the format callerid/extension yet? ie the extensions.conf file allows you to do: 5551212/1000 => exten ... and it matches against extension 1000 when the caller id is 5551212. Last time I checked, realtime didn't support this yet. That's a show stopper for us. -------------- next part -------------- An HTML

All, Slightly off topic. Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk.

Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat' soft-key to work? When you change the status in this way, the phone does not send any communication to Asterisk, and it seems to have no effect in incoming calls. So... what's it for? Doug

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says "the context for the voicemail box that we're looking for in the dialplan for the jump to the

I've noticed that with FastAGI (and maybe AGI) that when you sequentially send a sequence of dial commands, if the call is picked up, that after the call ends, the Fast AGI script keeps executing the commands! Is there anyway to stop execution once a call is picked up? I think looking at the result codes after the Dial to determine if the call was picked up or not is not a good idea... if it

Can someone tell me why this is not valid... [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@Local) exten => 2000,1,Macro(DialProxy,115551212) exten => 3000,1,Queue(testq||||45) while this is: [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@start) exten =>

I have a configuration where RTP traffic is going out interface pub0, and coming back into through pub1. I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows: udp 0 788 0.0.0.0:5060 0.0.0.0:* which means that Asterisk is listening on all addresses (on all interfaces?). Anyway, when the RTP traffic comes back in on interface pub0, Asterisk does nothing with it. A

Can someone recommend the best way to view current calls in progress on the Asterisk console? Neither the 'show channels' or 'sip show channels' commands are easy to read. hestia*CLI> show channels Channel Location State Application(Data) SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2) SIP/2944079-e7f2

Does anyone know how I can get stderr from AGI to be sent to somewhere other than the console? It seems that this is the only place it can go. Changing logger.conf has no effect. If you want to see errors from AGI scripts, you have to run the Asterisk console, which isn't viable. Doug.

Hi everyone, I have a question concerning DNS SRV lookups. The situation is like this: - one central Asterisk server - many domains with SRV records, let's say we have bar.com and doe.com Now the question is: if the SRV lookup is done for foo@bar.com the call is mapped to foo@myasterisk.mydomain.net. Is that correct? If so, I have a problem: if somebody calls foo@bar.com, Asterisk

I recollect that there used to be a fixed, finite limit to the number of call groups that could exist. Does anyone know if that limitation still exists in 1.2.1, or maybe where I could look in the code to find out if it's a fixed length array or not? Thanks. Doug.

Off topic. Anyone know where I can get Polycom SIP software v1.6.6, unofficially? Not that Polycom is analy retentive, or anything like that... Doug