RE: [Asterisk-Users] X100P dial in/out to sip phones
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Simon
Chappell
Sent: 07 March 2004 16:46
To: [EMAIL PROTECTED]
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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Re: [Asterisk-Users] X100P dial in/out to sip phones
Thanks for your help David
Your configs are a little to complicated for this complete asterisk
newbie though.
All i am actually after is how to get a sip phone to ring when the X100P
is dialed on out landline, and how to get a sipphone to dial out through
the X100P.
I have saved all your configs and had a trawl through them though.
I am a great believer in start simple then build it up and step by step
it seems simple in the end but I keep stumbling on this task. once i
have this i will look at call parking,conferencing (all the fun stuff)
etc.. but at the moment all i would like to acheive is bridging the gap
from sip to BT :-)IF you have any quick pointers to help me acheive
that I would be very pleased.
Thanks again for taking the time to reply (especially on a sunday
evening with the roast going cold)
Simon
David J Carter wrote:
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Simon
Chappell
Sent: 07 March 2004 16:46
To: [EMAIL PROTECTED]
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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--
Kind Regards
Simon Chappell
url : www.isnsuk.com
email : [EMAIL PROTECTED]
PH: 01403 268474
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Re: [Asterisk-Users] X100P dial in/out to sip phones
On Sunday 07 March 2004 20:08, Simon Chappell wrote: > Thanks for your help David > > Your configs are a little to complicated for this complete asterisk > newbie though. > All i am actually after is how to get a sip phone to ring when the X100P > is dialed on out landline, and how to get a sipphone to dial out through > the X100P. > I have saved all your configs and had a trawl through them though. > I am a great believer in start simple then build it up and step by step > it seems simple in the end but I keep stumbling on this task. once i > have this i will look at call parking,conferencing (all the fun stuff) > etc.. but at the moment all i would like to acheive is bridging the gap > from sip to BT :-)IF you have any quick pointers to help me acheive > that I would be very pleased. > Thanks again for taking the time to reply (especially on a sunday > evening with the roast going cold) > I've emailed you my configs off list :) Like you, I'm not yet looking to do anything complicated with my asterisk setup and I've just finished implementing exactly what you're trying to do. HTH Jon ___ 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
Re: [Asterisk-Users] X100P dial in/out to sip phones
On Sunday 07 March 2004 20:08, Simon Chappell wrote: > Thanks for your help David > > Your configs are a little to complicated for this complete asterisk > newbie though. > All i am actually after is how to get a sip phone to ring when the X100P > is dialed on out landline, and how to get a sipphone to dial out through > the X100P. > I have saved all your configs and had a trawl through them though. > I am a great believer in start simple then build it up and step by step > it seems simple in the end but I keep stumbling on this task. once i > have this i will look at call parking,conferencing (all the fun stuff) > etc.. but at the moment all i would like to acheive is bridging the gap > from sip to BT :-)IF you have any quick pointers to help me acheive > that I would be very pleased. > Thanks again for taking the time to reply (especially on a sunday > evening with the roast going cold) > > Simon Hopefully the attached configs will be of help to you. They're pretty basic :) The one's you'll be interested in are zapata.com, extensions.conf and possibly sip.conf. HTH Jon Lawrence asterisk_configs.tgz Description: application/tgz
Re: [Asterisk-Users] X100P dial in/out to sip phones
That is really appreciated :-) I look forward to receiving them and getting straight to work.. Many thanks Simon Jon Lawrence wrote: On Sunday 07 March 2004 20:08, Simon Chappell wrote: Thanks for your help David Your configs are a little to complicated for this complete asterisk newbie though. All i am actually after is how to get a sip phone to ring when the X100P is dialed on out landline, and how to get a sipphone to dial out through the X100P. I have saved all your configs and had a trawl through them though. I am a great believer in start simple then build it up and step by step it seems simple in the end but I keep stumbling on this task. once i have this i will look at call parking,conferencing (all the fun stuff) etc.. but at the moment all i would like to acheive is bridging the gap from sip to BT :-)IF you have any quick pointers to help me acheive that I would be very pleased. Thanks again for taking the time to reply (especially on a sunday evening with the roast going cold) I've emailed you my configs off list :) Like you, I'm not yet looking to do anything complicated with my asterisk setup and I've just finished implementing exactly what you're trying to do. HTH Jon ___ 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 -- Kind Regards Simon Chappell url : www.isnsuk.com email : [EMAIL PROTECTED] PH: 01403 268474 ___ 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
Re: [Asterisk-Users] X100P dial in/out to sip phones
Simon,
Try the following configs:
/etc/asterisk/zaptel.conf
fxsks=1
loadzone=uk
defaultzone=uk
/etc/asterisk/zapata.conf
languages=en
context=inbound-analog
signalling=fxs_ks
; I always create dial groups for making outbound calls, you can use the
specific channels as well
group = 1
channel => 1
/etc/asterisk/sip.conf
[general]
; I generally prefer gsm and ulaw, you can allow any codecs you like
disallow=all
allow=gsm
allow=ulaw
; use your IP address in the bind address or leave as 0.0.0.0 to bind to
all active interfaces
port=5060
bindaddr=0.0.0.0
; set your tos - see www.voip-info.org command reference for tos values
tos =0x10
;next create an entry for your SIP phones
; you can specify username and secret or you can set a very explicit permit.
; canreinvite, no=asterisk remains in media path, yes=asterisk CAN step out
of media path
; if you have problems with authentication, try removing the username,
secret, and permit lines
; and setting host=a.b.c.d where a.b.c.d is the ip address of the SIP client
; the example permit will permit any clients with 10.0.0.0 255.255.255.0
address space
[2001]
type=friend
username=2001
secret=2001
host=dynamic
permit=10.0.0.0/8
canreinvite=no
context=intern
callerid=Test Caller
mailbox=2001
nat=yes
/etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=yes
[globals]
; used for global variables, which in this basic example, we'll completely
ignore
[outbound-analog]
exten => _X.,1,Dial(Zap/g1/${EXTEN},60)
exten => _X.,2,Hangup
[inbound-analog]
exten => s,1,Dial(SIP/2001,20)
exten => s,2,Voicemail(u2001)
exten => s,3,Hangup
exten => s,102,Voicemail(b2001)
exten => s,103,Hangup
[local]
; Note we don't send local callers to Voicemail in this example
exten => 2001,1,Dial(SIP/2001)
exten => 2001,2,Hangup
exten => 2001,102,Hangup
exten => 2999,1,Answer
exten => 2999,2,Wait(1)
exten => 2999,3,VoiceMailMain
exten => 2999,4,Hangup
[intern]
include => local
include => outbound-analog
/etc/asterisk/voicemail.conf
[EMAIL PROTECTED]
attach=yes
maxmessage=300
maxgreet=60
[default]
2001 => 1234,John Doe,[EMAIL PROTECTED]
This should give you a very basic system with a SIP phone client, one
outside line via X100P, and voicemail. the Sip client will be able to call
voicemail using 2999, and any other sip clients you configure by dialing
their extension. When someone calls the analog number from the outside
world, the sip client at 2001 will ring, if no one answers, the caller will
be sent to leave a voicemail message, if 2001 is busy, the caller will be
sent to voicemail with a prompt indicating the caller is busy.
Hope this helps.
-Chris
At 12:08 PM 3/7/2004, you wrote:
Thanks for your help David
Your configs are a little to complicated for this complete asterisk
newbie though.
All i am actually after is how to get a sip phone to ring when the X100P
is dialed on out landline, and how to get a sipphone to dial out through
the X100P.
I have saved all your configs and had a trawl through them though.
I am a great believer in start simple then build it up and step by step it
seems simple in the end but I keep stumbling on this task. once i have
this i will look at call parking,conferencing (all the fun stuff) etc..
but at the moment all i would like to acheive is bridging the gap from sip
to BT :-)IF you have any quick pointers to help me acheive that I
would be very pleased.
Thanks again for taking the time to reply (especially on a sunday evening
with the roast going cold)
Simon
David J Carter wrote:
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Simon
Chappell
Sent: 07 March 2004 16:46
To: [EMAIL PROTECTED]
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})
exte
Re: [Asterisk-Users] X100P dial in/out to sip phones
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.. Simon Jon Lawrence wrote: On Sunday 07 March 2004 20:08, Simon Chappell wrote: Thanks for your help David Your configs are a little to complicated for this complete asterisk newbie though. All i am actually after is how to get a sip phone to ring when the X100P is dialed on out landline, and how to get a sipphone to dial out through the X100P. I have saved all your configs and had a trawl through them though. I am a great believer in start simple then build it up and step by step it seems simple in the end but I keep stumbling on this task. once i have this i will look at call parking,conferencing (all the fun stuff) etc.. but at the moment all i would like to acheive is bridging the gap from sip to BT :-)IF you have any quick pointers to help me acheive that I would be very pleased. Thanks again for taking the time to reply (especially on a sunday evening with the roast going cold) Simon Hopefully the attached configs will be of help to you. They're pretty basic :) The one's you'll be interested in are zapata.com, extensions.conf and possibly sip.conf. HTH Jon Lawrence -- sig Kind Regards Simon Chappell url : www.isnsuk.com email : [EMAIL PROTECTED] PH: 01403 268474
Re: [Asterisk-Users] X100P dial in/out to sip phones
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 [EMAIL PROTECTED] 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] X100P dial in/out to sip phones
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 [EMAIL PROTECTED] 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 : [EMAIL PROTECTED] PH: 01403 268474 ___ 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
RE: [Asterisk-Users] X100P dial in/out to sip phones
> Caller ID does not work in the UK, well not on my BT or Telewest line's. What I didn't understand yet about * + X100P with caller id not working in some countries is, it's a hardware or software limitation? Isamar ___ 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
Re: [Asterisk-Users] X100P dial in/out to sip phones
On Monday 08 March 2004 00:27, Isamar Maia wrote: > > Caller ID does not work in the UK, well not on my BT or Telewest line's. > > What I didn't understand yet about * + X100P with caller id not working > in some countries is, it's a hardware or software limitation? > > > Isamar > I tihnk that it's primarily a hardware limitation. The callerID on a UK BT line is sent before the first ring tone. It is initiated by a line reversal - from what I can gather the x100p can't detect this. There are supposed to be some new modules coming out for the TDM400P which will work - but then they were supposed to be out before Christmas - does anyone know when/if they are going to get released. Jon ___ 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
