search for: jklimek

...This flag trumps the 'H' flag and intercepts any dtmf while waiting for > the call to be answered and returns that value on the spot. This allows you > to dial a 1-digit exit extension while waiting for the call to be answered - > see also > > > On 6/22/06, John Klimek <jklimek@gmail.com> wrote: > > Anybody have any more information on this Dial() "d" option for incoming > calls? > > > > On 6/19/06, John Klimek < jklimek@gmail.com> wrote: > > > Thanks for the information... > > > > > > After doing some rea...

I'm trying to setup Asterisk on my Linksys WRT54G router and it appears to startup successfully (no errors) and it says it is listening on 0.0.0.0 port 5060, but I am unable to connect to it. I've tried "telnet localhost 5060" but it just says connection refused. I've also tried connecting from another machine on my network (eg. telnet 192.168.0.1 5060) but it also says

Ok, I've done some more research and I don't think I want an FXO box... What I'd like to do is use BroadVoice (with their BYOD plan) and then run Asterisk on my WRT54G router. I'd also like to use my regular home phones without having to use a special "SIP" phone... (eg. I like my Vtech normal cordless phones) What do I need to buy to get this working? It sounds like I

Asterisk seems to register just fine with BroadVoice (asterisk -r, and then sip show registry shows sip.broadvoice.com is "Registered") ...but when I try to call my broadvoice number (from a cell phone), it rings one single time and then says "The party you are trying to reach is not available to take your call." This doesn't seem to be an Asterisk message but seems to be

I have a very, very simple Asterisk setup in my house. I have a Sipura 3000 with a PSTN line connected and one analog phone connected. The [incoming] context looks like this: exten => s,1,Dial(SIP/50,23,r) exten => s,2,VoiceMail(u50@default) exten => s,3,Playback(vm-goodbye) exten => s,4,Hangup As you can see, when somebody calls in if I don't answer in 23 seconds then they are

Incoming calls from my Sipura 3000 don't seem to be correctly routing to Asterisk (or something?) Here is my Asterisk configuration for my incoming PSTN line: Code: [1000] type=friend host=dynamic context=incoming secret=6769 dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very Inside of extensions.conf, I have this: Code: [incoming] exten => s,1,Answer( ) exten =>

First off, I'm sorry for sending so many messages to the list-serv. Hopefully this will be my last for a while! I was going to use my WRT54G router as a small Asterisk box, but I forgot that I had a spare eMachines computer (Intel Celeron 633 MHz, 20GB HD, 64mb RAM). Will this machine work OK for a very simple dedicated home Asterisk box? Also, what is easiest linux distribution to use and

I'm trying to setup Asterisk on my Linksys WRT54G router but it only has 4MB of storage (and only about 250KB free with OpenWRT linux installed). I'd like to setup voicemail, but I'm afraid I don't have enough storage on my router. Can I somehow store this on my VOIP provider? (eg. BroadVoice) I was also thinking about mounting an NFS partition to store the voicemail, but then

I'm using a Linksys WRT54G router and it works great but it only has 4MB of storage. One of my only options is to modify the router to accept SD cards (eg. 512MB) but the access time is only around 200 KB/s. Will this be fast enough to store voicemails? ...or do I need another (faster) storage device?

First off, I'm a little confused but I've been trying to do some research and still have some questions... (please forgive me!) I'm trying to figure out the "best" (ie. fastest) way to connect from Ubuntu to Windows Server 2008. I can use SMB, SMB2, or NFS.