I had this, my problem turned out to be in zapata.conf on the receiving
end.
I'll do the KS, right now I am using LS. Any particular reason to use
KS? The LSCPD on the adit seems to work fairly decently.
Now I just need to work out some echo, although I have done milliwatt
tests to a local line, I still seem to get echo at the beginning of a
call regardless of how I set the training.
Thanks,
Greg
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F
Sent: Monday, October 17, 2005 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Callerid on t1 lines
How are you checking if CallerID is received?
You should do at least a Noop(${CALLERIDNUM}) or if running head:
Noop(${CALLERID(NUM)}) so that you can verify that.
How do you know that your telco is giving you CID?
If you live in the US then setup the Adit to do LSCPD and Asteisk as
ks_fxs. and not loop start.
On 10/17/05, gw@adcomcorp.com <gw@adcomcorp.com>
wrote:> Hello,
> That's what I really needed to know, that it was possible.
>
> Here is my setup:
>
> Telco Analog W/CID > FXO ADIT600 LoopStart > Loopstart Asterisk T1.
>
> Then LoopStart Asterisk T1 > Loopstart Panasonic DBS PBX T1.
>
> At this point, I do not see any CID coming in from the telco into
> asterisk. Even when I increase the wait time, and the zapata.conf has
> asreceived set.
>
> I tried E&M from the dbs to asterisk, but would get no dialtone from
> asterisk as it was not working properly with immediate mode.
>
> The main purpose of the setup is to do call recording on 3 analog and
> 2 bri lines, and pass them to the pbx transparently. Also to allow *
> transfers and queuing.
>
> Thanks,
> Greg
>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F
> Sent: Saturday, October 15, 2005 9:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Callerid on t1 lines
>
> What is the adit 600 doing? FXO? FXS? how you connected to the PSTN?
> I got an Adit 600 with both FXO and FXS as well as a PRI and I'm
> getting CallerID on all three.
>
> On 10/14/05, gw@adcomcorp.com <gw@adcomcorp.com> wrote:
> > Hello All,
> > Just a question, I have an adit600 and I am looking for a way to
> > pull the incoming cid into asterisk.
> >
> > Does anyone know if this is just not possible via t1? Or is it only
> > available on PRI?
> >
> > Thanks,
> > Greg
> > _______________________________________________
> > --Bandwidth and Colocation sponsored by Easynews.com --
> >
> > 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
> >
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> 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
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> 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
>
_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --
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