[Asterisk-Users] X100P dial in/out to sip phones

David J Carter david.carter at codepipe.com
Sun Mar 7 10:18:26 MST 2004


Simon,

Caller ID does not work in the UK, well not on my BT or Telewest line's.

Have a look at my sample configs http://www.codepipe.com/id25.htm , I am
also in the UK and these work for me.

Give me a call if ya want to chat about it.

Regards


Dave

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Simon
Chappell
Sent: 07 March 2004 16:46
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] X100P dial in/out to sip phones


Hello all

I have recently stumbled accross voip and asterisk.
We have a small network of vpns running in the uk. I have managed to get
the sip phones dialing each other through asterisk and it is working
great. (we are having long free conversations and that is something to
get excited about)..
My problem is that I cannot get the X100P i recently bought to dial out
or do anything with incoming calls.
I did loads of googling and found this snippet that made the zaptel card
moan at me about callerid ask me to type a number then do nothing but
offer silence..
[inbound-analog]
exten => s,1,Zapateller(answer|nocallerid)
exten => s,2,NoOp
exten => s,2,Macro(record-on,${PHONE1},${CALLERIDNUM})
exten => s,3,PrivacyManager
exten => s,4,Dial(${PHONE1},15,Ttm)
exten => s,5,Answer
exten => s,6,Wait(1)
exten => s,7,Playback(new/hello)
exten => s,8,Playback(new/marisa-john-not-in-momnt)
exten => s,9,Playback(new/theyre-rattlesnake-wrstling)
exten => s,10,Voicemail(u${PHONE1VM})
exten => s,11,Hangup
exten => s,108,Wait(2)
exten => s,109,Voicemail(b${PHONE1VM})
exten => s,110,Hangup
If i rem out that and run asterisk with -vvg i get this when i dial in
to the x100p
Mar  7 16:43:41 NOTICE[245776]: chan_zap.c:4624 ss_thread: Got event 2
(Ring/Answered)...
Mar  7 16:43:44 NOTICE[245776]: chan_zap.c:4624 ss_thread: Got event 2
(Ring/Answered)...
Mar  7 16:43:44 NOTICE[245776]: chan_zap.c:4624 ss_thread: Got event 2
(Ring/Answered)...
Mar  7 16:43:49 WARNING[245776]: chan_zap.c:4695 ss_thread: CallerID
returned with error on channel 'Zap/1-1'
Mar  7 16:43:49 WARNING[245776]: pbx.c:1778 ast_pbx_run: Channel
'Zap/1-1' sent into invalid extension 's' in context 'default', but no
invalid handler

So i feel i am getting there..
I would like the extensions to dial out and ring when the line rings..
can anyone give me a clue or point me in the right direction

I am in the UK by the way if that makes a difference.

Many thanks in advance

Simon

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at 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




More information about the asterisk-users mailing list