similar to: How to slow down dialing

Hi, I'm new to telephony Software and Hardware, so please excuse my questioning. I plan to set up a little system, using Asterisk and VoDSL via Belcacom or Scarlet here in belgium. We are yust a little 2 man company and we are not always in our office. My idea is, to get VoDSL and set up a system that works as following: A customer sends SMS or phones to our office-numbers, if we are out,

I have a Sipura 2000 working fine, but whenever I dial any extension there is a delay of 5-10 seconds before it starts ringing. However, if I dial the extension and hit the pound key after the number, it goes through right away. Is there any way around this?

Hi. I have a SPA3102 updated with with Software Version: 5.1.7(GW). I have this settings on Voice/Regional: Interdigit Long Timer: 10 Interdigit Short Timer: 3 Anyway, when hooking up (without dialing anything), the timeout starts after 3 seconds. It's like the Long Timer is unused. After dialing, the Short Timer is also used to timeout. Is that normal? Am I missing something? Thanks. --

Possibly OT? I've hooked up a Linksys PAP2 to my Asterisk 1.6 fairly painlessly. The only issue I can't beat with it is the dial delay when calling internal or external numbers. No matter what it seems to take 10 -15 seconds to actually dial. I've altered the device removing all *xx combos and unnecessary waffle and cut the dialplan string to (x.S0) but the problem persists. Anyone

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Hello list, does anyone know how to change the "interdigit timeout" when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in tftp-config file etc. Thanks in advance, Roland

I am trying to integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only

Hi, Long story short - I have an ancient Britsh Telecom phone attached to my Asterisk PBX via Dahdi. It works beautifully, receiving calls, and the call quality is excellent. However, dialling out is impossible, as Asterisk consistently mis-reads the number of pulses the dial sends (it could be a squiffy dial, I'm not sure). Not to mention the fact that, in today's modern "want

Folks, I've got one of those GPO 1950's rotary dial phones that I'm trying to get working in the UK. I've got pretty much everything working with my TDM400, the phone rings and I can receive calls but I cannot dial with the rotary dialer. I have set pulsedial=true or whatever the exact setting is and I can dial from the phone by lifting the receiver and tapping out the number on

and nothing happens. i run mandriva spring2008.1 i have downloaded limewire and squeezebox duet and i cant get them to run. any suggestions? thanks randog

On Tue, 2005-01-18 at 09:49 -0600, asterisk-dev-request@lists.digium.com wrote: > > Hi, > > I am Arnaud F?vrier, I teach in a technical university in Marseille. > > I'd like to know if is is possible to connect a very old phone to > asterisk and dial pulses with it? > > Are digium cards pulse dial compatible? > > Is there any specific configuration

I am trying to configure the FXO port on a Sipura-3000 for use with Asterisk. When I connect to the Sipura to dial out on the PSTN line connected to the Sipura's FXO port, it gives me the dialtone of the PSTN line and then I can hear the DTMF for the number I dialled beforehand. It does work but the customer perceives this delayed second DTMF feedback as "unprofessional" and the

Hi, Which is the correct syntax to call using IAX? I have two Asterisk boxes behind a NAT and one of them use the default port 5036 for IAX, the second one use 5038. To call an extension of the first one, the line in extensions.conf is: exten => _9XXX,1,Dial(IAX/user:pass@195.3.32.191/${EXTEN:1}) and for the second one: exten => _8XXX,1,Dial(IAX/user:pass@195.3.32.191:5038/${EXTEN:1})

> On a related note, how do you get a Cisco 7940 to dial numbers with a > hash in them, instead of just using the hash as a dial key. For > example, I have *#21# to check diverts, but the phone will just dial > "*" as soon as you type the # after it. <DIALTEMPLATE> <TEMPLATE MATCH="#..." Timeout="5" User="Phone" /> <TEMPLATE

Hi all, I have a problem with the dialing tone in PAP2: When making a call, I can hear the calling tone 5 times and then it stops. The called party still hears the call but not the calling party. I've playing around with different parameters on the PAP2 web config with no success until now. Anyone has seen the same probelm? Thanks, Jose

Hi, I am using * to guide my callers throught my company's support menu. But I have problem when the caller has a "pulse dial" telephony. Could * detect digits dialed on pulse telephones? Daniel

We are installing a number of systems with 2 TDM04B cards. Have done all the isolation to unique IRQs, etc. All inbound calls seem to work fine. However, outbound calls are hit or miss. Sometimes they work fine and other times we get a "you must first dial a 1 or 0" message back from telco when dialing out standard POTS lines. We are running AAH 1.0 which is Asterisk 1.0.7. Six

I have the auto answer working fine on my 7960 phones on line 2, but when line 2 answers, it does not make any sound until the call begins speaking. I would like to have the phone beep when it answers so the person being called knows the speaker phone is in use. This would be much like a overhead paging systems that plays a tone before an announcement is made. How can I have asterisk answer the

Does anyone have any recommendations for overhead paging systems for use with Asterisk? Thanks, Randy Johnson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031113/82ae09ec/attachment.htm

I have server with a total of 6 Analog ports...using TDM04B and TDM02B. I have 3 Lines that are DIDs and 3 are Main/Roll Over lines and I have worked through getting the DIDs to work and route to the extensions...now what I need to do is when Extension 1111 picks up the phone to dial, I would like them to use their DID analog line first, unless someone has called in on it and they are trying to