Is it possible to make Asterisk jump into action and play a sound file as soon as a handset is lifted, instead of providing a Dial Tone and waiting for the user to dial an extension.

With analog phones (chan_dahdi) – you just have to set ‘immediate = yes’ in chan_dahdi.conf , with a SIP phone: that’s something to configure the handset for, as it only sends out a call once you “dialed”.

Thanks to: Tzafrir Cohen

Dear, when I dial *30 in order to get instructions to blacklist an
extension, Idon’t get the menu but I get a new dial tone.

What happen please ??? What can I do to solve this ???

Thanks a lot,

Alejandro

Hello
I am working on TDM2400p. I am having some problems like:
when i connect my analog phone with the card there is no dial tone, but i
can dial any extension… but after that i can’t hear any voice from my
receiver i have used different phone sets but still i cant communicate with
other extension.
Please help me out.

Thank you

Regards
Ali Raza

Hi Ott,

Have you made it work with Asterisk and Aastra IP Phone. I am also trying
the same thing, in Asterisk it shows registered OK but when I dial from
extension to extension, call is failed…

Please let me know have you made it work…:(

On Mon, Jul 13, 2009 at 11:46 PM, Ott Rose wrote:

>
> I did ” set sip debug on ” from the CLI
>
> It doesn’t scroll messages like it did on Fri
>
>
> i tried 99# and the screen on the phone changed to an ip of 10.0.0.99 which
> isn’t either one of the ips of the asterisk server. then it hung up
>
> i do have a dial tone
>
>
> i just figured something out after reading my post.
>
>
> if i dial 60# it shows the ip 10.0.0.60 of the other phone then switch to
> the extension and the other phone rings.
>
> still can’t get the 99 to call the asterisk server to work i put in the ips
> of the server but it hangs up right away
>
> ——————————
> From: href=”mailto:danny@debsinc.com”>danny@debsinc.com
> To: href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com
> Date: Mon, 13 Jul 2009 12:57:59 -0500
>
> Subject: Re: [asterisk-users] setting up phones
>
> I assume you get a dial tone when you pick up the handset? If you had
> a good phone-to-asterisk connection, debug would show a connection or
> rejection when you did 99#.
>
>
> ——————————
>
> *From:* href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [mailto:
> href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com] *On Behalf Of *Ott Rose
> *Sent:* Monday, July 13, 2009 12:49 PM
> *To:* href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com
> *Subject:* Re: [asterisk-users] setting up phones
>
>
>
> added that line to the extensions.conf file because i could find a way to
> add it in the GUI. I put it under the dial plan that i have selected. i just
> get a busy signal i tried #99 just 99, *99 nothing works. debugging isnt
> showing anything.
> ——————————
>
> From: href=”mailto:danny@debsinc.com”>danny@debsinc.com
> To: href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com
> Date: Mon, 13 Jul 2009 12:12:16 -0500
> Subject: Re: [asterisk-users] setting up phones
>
> Most folks (AFAIK) use TFTP to connect to the Asterisk server. I
> personally use HTTP, but that took a few days of research to figure out.
> You’re really only using that protocol for configuration and log transfers.
> The actual lifting is done on a TCP or UDP connection. Your posts Friday
> indicated that Asterisk was up and “functional” but that you couldn’t make
> your phones talk to it. I’m thinking that instead of trying to dial
> phone-to-phone, that you should first make one phone talk to asterisk using
> this little snippet.
>
>
>
> – exten => 99,1,Playback(tt-monkeys)
>
> – exten => 99,2,Playback(vm-goodbye)
>
> – exten => 99,3,hangup
>
>
>
> When you get your phone where it can dial 99 and get a message, you will be
> ready to proceed with P2P talking.
>
>
> ——————————
>
> *From:* href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [mailto:
> href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com] *On Behalf Of *Ott Rose
> *Sent:* Monday, July 13, 2009 12:02 PM
> *To:* href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com
> *Subject:* Re: [asterisk-users] setting up phones
>
>
>
> Ok here is what i did.
>
> reinstalled asterisk (i used the make samples option) and asterisk-gui
>
> in the gui i did the following
> created a dial plans using the defaults. no outgoing dial plans just local
> crated two users
> logged into the web interface with each phone and pointed them to our
> asterisk server. Just the Proxy server and Registrar server.
>
> Still doesn’t work. Should i be able to use the configuration server
> settings form the phones web gui. it has the options for tftp, ftp, http,
> https. I don’t know how this is supposed to be configured. I still don’t
> know what the problem is and sip set debug off does display any info like it
> was lastweek.
>
>
> I am just trying to use the gui like you suggestd
>
> > Date: Fri, 10 Jul 2009 14:22:25 -0700
> > From: href=”mailto:asterisk.org@sedwards.com”>asterisk.org@sedwards.com
> > To: href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com
> > Subject: Re: [asterisk-users] setting up phones
> >
> > On Fri, 10 Jul 2009, Ott Rose wrote:
> >
> > > I don’t think the GUI is editing the conf files correctly. I am not
> sure
> > > I have configure things right. At this point i think i am going to
> start
> > > from scratch.
> >
> > Yea!
> > –
> > Thanks in advance,
> > ————————————————————————-
> > Steve Edwards href=”mailto:sedwards@sedwards.com”>sedwards@sedwards.com Voice: +1-760-468-3867 PST
> > Newline Fax: +1-760-731-3000
> >
> > _______________________________________________
> > — Bandwidth and Colocation Provided by http://www.api-digital.com –
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> ——————————
>
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>

Okay;

So I can use a Digium FXO/FXS type card and use the dial tone to utilize
Vonage with Asterisk. Done it – simple enough.

However…..I am wondering if anyone is Cracker-Jack enough to come up
with a way to get SIP credentials? I went as far as asking Vonage
directly and the answer I got was a big fat “NO”.

I am thinking it is probably a violation of their acceptable use policy
to do it – honestly, I have been with Vonage for about 8 years and never
read it.

I know there are other services out there that will give you the SIP
info (Broadvoice).

Anyone been successful and willing to share the knowledge?

Cheers

Glen