[Asterisk-Users] Help: TE100P connecting to non PRI, ISDN interfaces
Hello, i've googled and can't find a definite answer, so here goes: I have purchased the Digium TE100P, and am setting up the connection, however the telco i'm supposed to work with does not support PRI/ISDN E1 connections. They only support E1/R2 lines. Is there a way i can make the TE100P work with this? I've not seen any zaptel.conf that supports this. Any workarounds? Thanks for any help! Ian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cellsocket help needed
I have a cellsocket working with a Nokia 6150 right now. Funny the
model I bought was a cellsocket for a Nokia 5110, and for some reason,
it wont work with the 5110 unit i put in. The 6150 works like a
charm, though. For some reason, it ignores the first two characters
of the phone number you dial, so i had to do something like:
PAUSE=**
ZERO=0
SHARP=#
;
; Outbound to 5nxx- goes via: CellSocket
exten => _59X,1,Dial(Zap/1/${PAUSE}${ZERO}${EXTEN:1}${SHARP})
exten => _59X,2,Congestion
The ZERO is used to call local long distance, and the SHARP is needed
by cellsocket
to tell it that it ends the number stream.
On 5/9/05, Manny A. Wise <[EMAIL PROTECTED]> wrote:
>
>
>
> I need help from someone who has a working cellsocket, I have received
> couple email of people who wanted to help, but they just think they know how
> it supposed to work, but they don't have a working units, and they confused
> more…..I need someone with a working solution to get my cellsocket going….
>
> Thanks!!!
>
> Write offlits @ mawise (AT) mail.com
>
>
>
>
> ___
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Re: [Asterisk-Users] Is there any chance to bring Skype andAsteriskUser together?
I think the proper solution would be to use the proprietary Skype API for Linux and create an asterisk extension for it. There is a $1050 bounty on voip-info.org[1] but i don't think there are any takers for it yet. :( Another suggestion was "... to either get a Skype compatible ATA or FXS/FXO adapter, and just live with that, that's probably the closest you'll get to it any time soon" [1] http://www.voip-info.org/wiki-bounty+skype On 5/4/05, Dean Collins <[EMAIL PROTECTED]> wrote: > You could run an automated session out your speaker/mic to an incoming > fxs circuit but to answer your question - No. > > Never heard it happen before. > > Cheers, > Dean > > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili > > Sent: Tuesday, May 03, 2005 6:04 PM > > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Subject: RE: [Asterisk-Users] Is there any chance to bring Skype > > andAsteriskUser together? > > > > What do you mean? > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] On Behalf Of Kib > Eki > > > Sent: Tuesday, May 03, 2005 3:16 AM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: [Asterisk-Users] Is there any chance to bring Skype > > > and AsteriskUser together? > > > > > > Hi, > > > > > > is there any chance to bring Skype and Asterisk User together? > > > > > > Regards, > > > Kib > > > > > > ___ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * > Mobile Phone > Mobile Network
Another option is something like cellsocket http://www.cellsocket.com I haven't tried these, but some positive experiences posted in some sites ive been googling seem encouraging. There are models for motorola and nokia phones. On Mon, 21 Feb 2005 15:21:59 +1100, Mathew McKernan <[EMAIL PROTECTED]> wrote: > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of David > Uzzell > Sent: Monday, 21 February 2005 2:53 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] * > Mobile Phone > Mobile Network > > Ok I have a question. Seen it come and go around the mailling list for a > > while but never really seen an answer that seems to sort it out. > > What is needed is some interface from * > Mobile Phone > Mobile Network > Service. > > At this point all the providers in AUS that I have found are charging a > Premium Rate for Land Line > Mobile Network services. > > What I would like to do is be able to purchase a low rate Mobile SIM > that I can chuck into a Mobile Phone and have it setup so that I route > the Mobile calls through it. > > Rembering that most if not all mobile phones can be accessed via RS232 > interface. > > Anyone done this or seen it done or know how to do it using * and > whatever? > > Cheers > David > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > Hi David, > > Have a look at some second hand Ericson kits on Ebay. They had special > units, that basically had a normal GSM Ericson phone in them. But on the > side had a normal Australian 610 socket and rj11 socket. > > You could simply interface this into your digium cards as a normal pstn > line. > > They were originally designed for the exact purpose you want for > coupling with existing telephone systems. They are also used for > connection to fire signalling units and alarm systems. > > Thanks > > Mathew McKernan > Digital World Computers > Maribyrnong VIC > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie seeks help: Getting Asterisk to run on Mandrake 9.2
Hello, i was hoping someone from the list could point me in the correct direction. We recently purchased the Asterisk Developer's Kit (TDM) over at this link: http://www.digium.com/index.php?menu=developerskit_tdm And now, i'm trying to get this to work in Mandrake 9.2. I've gotten a fairly recent source RPM that takes the source from CVS, compilies and links, and provide three neat packages. I've gotten the kernel modules to load, the stock config files are in /etc/asterisk, and asterisk runs when invoked by asterisk -c My problem now is that i can't seem to get a dial tone from the extension phone. (i've connected a phone to the FXS, and an outside line to the FXO). Alhtough the LED next to the phone socket lights up, and the phone earpiece emits a tone when you press a key, there is no dialtone, and no matter what you do, nothing happens => both on the screen/cli of asterisk and the phone itself. Silence. When you call up the direct line, it just rings forever. Now, i'm thinking... config problem. However the stock config files of asterisk are a lot, and i haven't seen a config-set which is tailored to my exact setup. I've tried http://www.voip-info.org/wiki-Asterisk+quickstart But that gets you going with SIP, which i think i'll do when i've gotten the phone extension to "dial 9 for an outside line" successfully first. Any ideas or pointers to a cookbook recipe would be very much appreciated. Thanks in advance. - Ian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
