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
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@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Simon
Chappell
Sent: 07 March 2004 16:46
To: asterisk-users@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
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Thanks again.. I was beginning to think I had a faulty card now i see it was just me.. :-) I am going to make a backup of my config and spend some time reading about all the extra nice features that i can implement now the basic stuff is working.. ##Thought## The last company i worked for I was in charge of getting a new phone system,, and I think it was just asterisk in a nice black box !! I have noticed that the voicemail is the same voice, the commands to access the voicemail are identical and the features are the same also.. We payed ?9600+vat for that !! and it looks like i have basically the same thing at home now for a bit of time and effort(and a little help) With that in mind I think Asterisk is here to stay !! Thanks very much for you help .. A very happy Simon :-) Jon Lawrence wrote:>On Sunday 07 March 2004 21:28, Simon Chappell wrote: > > >>thanks so much.. >> >>I have dialed from my mobile and nearly fell off my chair when the Sip >>phoone rang ,,!! then was sad enough to answer it and have a chat with >>myself!! >> >>Is there any provision for dialing out in these configs ? and if so is >>it dial 9 ? >> >>Thanks again as this has been a four day headache so far.. >> >> >> >To dial out, simply prefix by *2 > >Jon > >_______________________________________________ >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 > >-- Kind Regards Simon Chappell url : www.isnsuk.com email : s.chappell@isnsuk.com PH: 01403 268474