Dude,
You HAVE to join us on our pub crawl!
Jim
Ian Darwin wrote:
> Part N of my continuing saga.
>
> A few months back I wrote of my problems getting dialing
> working with an ATA. Two ATAs later, I have given up and
> replaced all the hardware. It is now a standard PC (Celeron
> 1.2GHz) running "Asterisk Now" 1.0.2 with a cloned TDM410 (1
> FXO) PCI card talking to my analog phone line, and various
> (SIP and Unistim) phones inside.
>
> Part of the complexity was and still is that we have an ALDS (Alt.
> L.D. service) that we activate by dialing 10 digits, waiting for
> dialtone, and dialing the actual 10-digit number.
>
> There is some progress. I can fairly reliably have Asterisk
> dial numbers both locally and over the ALDS. Incoming calls
> still work; this has never been the problem.
>
> But everything involving dialing from the phones while
> connected is unreliable. E.g., dealing with IVR-based
> voicemail, scheduling, etc.
>
> The setup, now, is
>
> Analog ---- TDM400 FXO -- Asterisk -- internal phones
>
> Most of the phones are on a full-duplex switch, because that
> is providing POE for several of the phones.
>
> Here is part of the sip.conf for one of the SIP phones:
>
> [general]
> realm=XXX
> disallow=all
> allow=ulaw
> localnet=192.168.1.0/24
>
> ; Grandstream BT100 phone
> [31]
> type=friend
> secret=XXX
> context=internal
> nat=no
> host=dynamic
> canreinvite=no
> qualify=yes
> mailbox=20
> dtmfmode=rfc2833
>
> The GS phone has these settings:
>
> Send DTMF: in-audio X via RTP (RFC2833) via SIP INFO
> DTMF Payload Type: 101 (that was the factory default)
>
> -------------------------
>
> Here is the zapata.conf:
>
> [trunkgroups]
>
> [channels]
>
> ; default
> ; usecallerid=yes
> ; hidecallerid=no
> ; callwaiting=no
> ; threewaycalling=yes
> ; transfer=yes
> ; echocancel=yes
> ; echotraining=yes
>
> ; increased, to let sending work.
> toneduration=300
>
> ; last resort
> relaxdtmf=yes
>
> ; define channels
> context=incoming
>
> ---------------------------------
>
> Unistim.conf isn't shown but it has no settings related to
> DTMF; these phones always communicate over a local LAN and
> always send DTMF out of band (according to chan_unistim anyway)
>
> ---------------------------------
>
> And here's the local and ALDS dialing macros:
>
> PSTN_DIALER=Zap/1
>
> [macro-dial-outbound-local]
> exten => s,1,SetCallerID("Darwin" <905-XXX-XXXX>) exten =>
> s,n,Dial(${PSTN_DIALER}/${ARG1},20)
> exten => s,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
> exten => s,n(unavail),Congestion() exten => s,n,Hangup()
> exten => s,n(busy),Congestion() exten => s,n,Hangup()
>
> [macro-dial-outbound-worldline]
> ;exten =>
> s,1,Dial(${PSTN_DIALER}/${WORLDLINE_ACCESS},10,D(wwwww${ARG1}))
> exten => s,1,Dial(${PSTN_DIALER}/${WORLDLINE_ACCESS}wwwwww${ARG1}))
> exten => s,n,GotoIf($["${DIALSTATUS}" = "BUSY"],busy) exten
> => s,n(busy),Congestion() exten => s,n,Hangup()
> ---------------------------------------
>
> Summary: dialing via either of these macros works, but
> anytime we send DTMF from any keypad, it's unreliable.
> Dialing "9" by itself, for example, dials Worldline's number,
> where sending 10 digits should work.
> I wonder if this is related to why the first attempt at
> dial-outbound-worldline didn't work?
>
> Using an analog phone on the same phone line works 100%, btw.
>
> Thoughts? Further information needed?
>
> Many thanks if you can help.
>
>
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