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 ___ 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 ___ 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 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 ___ 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 ___ 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
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