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Hello asterisk-users, Sean -- +---+ |VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie | |GPG Key http://thecivvie.fastmail.fm/thecivvie.asc | +---+ Strange things happen under the midnight sun when Men and Dogs go hunting for gold smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re[2]: [Asterisk-Users] Indications for Ireland
Hello Ronan,
Thursday, September 15, 2005, 10:13:13 AM, you wrote:
Hi Sean,
This is what I've got in my zaptel zonedata.c file for a small * box in
Dublin:
{ 18, ie, Ireland, { 400, 200, 400, 2000 },
{
/* Dialtone = 400//425//450 */
{ ZT_TONE_DIALTONE, 425 },
{ ZT_TONE_BUSY, 425/500,0/500 },
/* Ringtone = 400+450//425 */
{ ZT_TONE_RINGTONE,
400+450/400,0/200,400+450/400,0/2000 },
{ ZT_TONE_CALLWAIT, 425/180,0/200,425/200,0/4500 },
{ ZT_TONE_INFO, 950/330,1400/330,1800/330,0/1000 },
{ ZT_TONE_STUTTER, 350+440 },
{ ZT_TONE_CONGESTION, 400/400,0/350,400/225,0/525 },
{ ZT_TONE_DIALRECALL, 350+440 } },
},
Strange thing that my [EMAIL PROTECTED] box doesn't seem to recognize the ie
prefix when
I use it in the /etc/zaptel.conf file
Sean
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[Asterisk-Users] Indications for Ireland
Hello asterisk-users, Just curious if anyone has the indications for Ireland, tried googling for it to no avail. Sean -- +---+ |VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie | |GPG Key http://thecivvie.fastmail.fm/thecivvie.asc | +---+ Strange things happen under the midnight sun when Men and Dogs go hunting for gold smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Spped Dial setup from wiki
Hello Asterisk-Users,
I copied the speed-dial set at the wiki to my extensions_custom and
included it, the code is:
; Speed dial application. This will store 99 speed dials in the bins 01 - 99
; The database family is called speed and the varible is called spnum
;Storing 11 digit numbers
exten = _*#X,1, DBput(speed/${EXTEN:2:2}=${EXTEN:-11:11})
exten = _*#X,2, Playback(val_sp)
exten = _*#X,3, Hangup
;Reading the stored number back.
exten = _*1XX,1, DBget(spnum=speed/${EXTEN:2:2}) ;sets spnum to be the
required bin number
exten = _*1XX,2, Playback(currently)
exten = _*1XX,3, SayNumber(${EXTEN:2:2})
exten = _*1XX,4, Playback(is-set-to)
exten = _*1XX,5, SayDigits(${spnum})
exten = _*1XX,6, Hangup
;Retreiving numbers for alog dialing
exten = _*9XX,1, DBget(spnum=speed/${EXTEN:2:2}) ;sets spnum to be the
required bin number
exten = _*9XX,2, Dial(${TRUNK1}/${spnum})
exten = _*9XX,3, Congestion
;Retreiving numbers for iax dialing
exten = _*8XX,1, DBget(spnum=speed/${EXTEN:2:2}) ;sets spnum to be the
required bin number
exten = _*8XX,2, Dial(${TRUNKiax}/44${spnum:${TRUNKMSD}})
exten = _*8XX,3, Congestion
;Retreiving numbers for alog Phonecoop dialing
exten = _*7XX,1, DBget(spnum=speed/${EXTEN:2:2}) ;sets spnum to be the
required bin number
exten = _*7XX,2, Dial(${TRUNK1}/184088${spnum})
exten = _*7XX,3, Congestion
If I do *101 from the extension I get a recording saying the number
stored at 01 is . NP there is nothing there. But if I do *# and a
number I get a 484 on the phone, nothing shows in the debug log.
Can anyone see anything wrong here
Sean
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Re: [Asterisk-Users] Spped Dial setup from wiki
Hello Sean,
Sunday, August 28, 2005, 11:53:28 AM, you wrote:
Hello Asterisk-Users,
I copied the speed-dial set at the wiki to my extensions_custom and
included it, the code is:
; Speed dial application. This will store 99 speed dials in the bins 01 - 99
; The database family is called speed and the varible is called spnum
;Storing 11 digit numbers
exten = _*#X,1, DBput(speed/${EXTEN:2:2}=${EXTEN:-11:11})
exten = _*#X,2, Playback(val_sp)
exten = _*#X,3, Hangup
It seems to be 2 that is the problem. I discovered that the keys are
being saved okay but I cannot get the playback to work at storage time
Sean
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Re[2]: [Asterisk-Users] Spped Dial setup from wiki
Hello Sean,
Sunday, August 28, 2005, 1:38:42 PM, you wrote:
Hello Sean,
Sunday, August 28, 2005, 11:53:28 AM, you wrote:
Hello Asterisk-Users,
I copied the speed-dial set at the wiki to my extensions_custom and
included it, the code is:
; Speed dial application. This will store 99 speed dials in the bins 01 - 99
; The database family is called speed and the varible is called spnum
;Storing 11 digit numbers
exten = _*#X,1, DBput(speed/${EXTEN:2:2}=${EXTEN:-11:11})
exten = _*#X,2, Playback(val_sp)
exten = _*#X,3, Hangup
It seems to be 2 that is the problem. I discovered that the keys are
being saved okay but I cannot get the playback to work at storage time
I seem to have it work but two problems, the playback is not working
dunno why and I cannot store numbers less that 11 digits which
excludes a lot of local numbers for me
Sean
--
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Re: [Asterisk-Users] Satellite Broadband and VOIP
On Fri, 14 Nov 2003 04:38:59 -0800 cxpcman [EMAIL PROTECTED] wrote: You must have future vision :) ok then go ahead and try. but don't expect too much .the voices will be out of time . Sadly I am aware of this now but ISDN is slow slow :) Sean -- Sean Rima Gossamer Spider Web of Trust http://www.gswot.org Jabber/PSI [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Satellite Broadband and VOIP
On Sat, 27 Aug 2005 09:43:33 +0100 Sean Rima [EMAIL PROTECTED] wrote: On Fri, 14 Nov 2003 04:38:59 -0800 cxpcman [EMAIL PROTECTED] wrote: You must have future vision :) ok then go ahead and try. but don't expect too much .the voices will be out of time . Sadly I am aware of this now but ISDN is slow slow :) Thinking here and this can be bad :) At the moment, my asterisk box is connected to the net via a network, which is not the best thign for it anyway.Should I get Sat BB, I may transfer the EiCON Diva ISDN card over to it. But will I be able to control it going live, ie I would only want it live whilst I was making an outgoing VOIP call and then drop the inet connection. Sean -- Sean Rima Gossamer Spider Web of Trust http://www.gswot.org Jabber/PSI [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Satellite Broadband and VOIP
On Sat, 27 Aug 2005 12:14:55 -0700 (PDT) Julius Igugu [EMAIL PROTECTED] wrote: ok then go ahead and try. but don't expect too much .the voices will be out of time . Sadly I am aware of this now but ISDN is slow slow :) Thinking here and this can be bad :) At the moment, my asterisk box is connected to the net via a network, which is not the best thign for it anyway.Should I get Sat BB, I may transfer the EiCON Diva ISDN card over to it. But will I be able to control it going live, ie I would only want it live whilst I was making an outgoing VOIP call and then drop the inet connection. Sean I use a satellite connection and VOIP is ok! It depends,mostly, on what you expect! There's an inherent delay in the system usually about 700ms - 800ms! but this is bearable. I know there is a latency due to the distance. What type of sat modem do you use etc (maybe we shoud take this offlist) Sean -- Sean Rima Gossamer Spider Web of Trust http://www.gswot.org Jabber/PSI [EMAIL PROTECTED] smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re[2]: [Asterisk-Users] Satellite Broadband and VOIP
Hello Julius, Saturday, August 27, 2005, 10:38:18 PM, you wrote: RG384 - Gilat! That is the one I was thinking of going for :) Sean -- +---+ |VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie | |GPG Key http://thecivvie.fastmail.fm/thecivvie.asc | +---+ smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Satellite Broadband and VOIP
I live in a very rural area, BB access will never happen and the only choice I have it Satellite. I seen from a post to this list that Gilat sat modems are not recommended. Is this still the case or is there another alternative? Sean -- Sean Rima Gossamer Spider Web of Trust http://www.gswot.org Jabber/PSI [EMAIL PROTECTED] smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Satellite Broadband and VOIP
On Fri, 14 Nov 2003 01:43:21 -0800 cxpcman [EMAIL PROTECTED] wrote: Sean Rima wrote: I live in a very rural area, BB access will never happen and the only choice I have it Satellite. I seen from a post to this list that Gilat sat modems are not recommended. Is this still the case or is there another alternative? Well is not recommended because of the seektime . the information you send and recive have a delay no matter how fast your conection is .. so you gonna hear the voice out of time . wire have a lot faster response times than air soo... ur choice Sadly I have no choice, my nearest BB is over 60 miles away, the only community schemes are all SAT based anyway :( Sean -- Sean Rima Gossamer Spider Web of Trust http://www.gswot.org Jabber/PSI [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] SPA-2100 Analog Telephone Adapter
Eric Wieling aka ManxPower wrote: Sean Rima wrote: Eric Wieling aka ManxPower wrote: Sean Rima wrote: Does anyone have any experience of these, I have been offered one and am thinking of adding sticking it onto the back of my Asterisk box and just ignore the WAN port if possible, It would be to stick my exisiting phones onto the asterisk box No, you would ignore the LAN port. When I am at home I use this setup: Phones - 2100 FXS ports - 2100 WAN port - Ethernet Switch - Asterisk If I were to get another 2100 would I use the LAN port to connect to it? You would only use the LAN port if you wanted the device to provide NAT translation/routing between the LAN port and the WAN port. Ahh ok, will get a bigger switch in time then Sean -- ++ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | ++ smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] SPA-2100 Analog Telephone Adapter
Eric Wieling aka ManxPower wrote: Sean Rima wrote: Does anyone have any experience of these, I have been offered one and am thinking of adding sticking it onto the back of my Asterisk box and just ignore the WAN port if possible, It would be to stick my exisiting phones onto the asterisk box No, you would ignore the LAN port. When I am at home I use this setup: Phones - 2100 FXS ports - 2100 WAN port - Ethernet Switch - Asterisk If I were to get another 2100 would I use the LAN port to connect to it? Sean -- ++ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | ++ smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] SPA-2100 Analog Telephone Adapter
Dennis Gilmore wrote: Sean Rima wrote: Does anyone have any experience of these, I have been offered one and am thinking of adding sticking it onto the back of my Asterisk box and just ignore the WAN port if possible, It would be to stick my exisiting phones onto the asterisk box Sean I just bought 12 of them to link 5 offices PBX systems together. so far in my testing they work extremmly well with asterisk. you will want to modify the dial plan on it otherwise you will get a delay when calling extentions. Excellent, I am still waiting on the bloke to get back to me or else it is ebay :) Sean -- ++ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | ++ smime.p7s Description: S/MIME Cryptographic Signature ___ 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
[Asterisk-Users] SPA-2100 Analog Telephone Adapter
Does anyone have any experience of these, I have been offered one and am thinking of adding sticking it onto the back of my Asterisk box and just ignore the WAN port if possible, It would be to stick my exisiting phones onto the asterisk box Sean -- ++ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | ++ smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] SPA-2100 Analog Telephone Adapter
Eric Wieling aka ManxPower wrote: Sean Rima wrote: Does anyone have any experience of these, I have been offered one and am thinking of adding sticking it onto the back of my Asterisk box and just ignore the WAN port if possible, It would be to stick my exisiting phones onto the asterisk box No, you would ignore the LAN port. When I am at home I use this setup: Phones - 2100 FXS ports - 2100 WAN port - Ethernet Switch - Asterisk Excellent, will get it then Sean -- ++ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | ++ smime.p7s Description: S/MIME Cryptographic Signature ___ 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
[Asterisk-Users] Preventing an extension from dialing certain outbound codes
Is there anyway to prevent an extension from dialing certain codes. ie I want to prevent extension 203 from dialing number which start with 00 087 086 etc Sean -- ++ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | ++ smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] Preventing an extension from dialing certain outbound codes
Andrew Kohlsmith wrote: On Thursday 18 August 2005 15:35, Sean Rima wrote: Is there anyway to prevent an extension from dialing certain codes. ie I want to prevent extension 203 from dialing number which start with 00 087 086 etc You're thinking about it wrong. Devices can only dial #s that match the dialplan in their context. Simply don't include any extensions that match it. It is also painfully obvious that you haven't read much on Asterisk. I suggest you start by reading the Asterisk Handbook draft, and follow up by poking around the wiki and the mailing list, which is searchable with google. http://www.digium.com/handbook-draft.pdf http://voip-info.org/ and include site: lists.digium.com in your google terms to search the list. Ahh I never actually looked at it that way. I am atm reading the handbook and do browse the wiki site, it is usually my first port of call. I will read a bit more on the dial plans and contexts and work it out. -- ++ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | ++ smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] Preventing an extension from dialing certainoutbound codes
Benjamin Lawetz wrote: Just put 203 in his own context which reacts to those numbers, and then include your normal context [restrict] Exten = _00.,1,goto(unauthorised,1) Exten = _087.,1,goto(unauthorised,1) Exten = _086.,1,goto(unauthorised,1) Exten = unauthorised,1,Playback(invalid) Exten = unauthorised,2,wait(2) Exten = unauthorised,3,Hangup() Include = regular_context Thanks, I see I have a lot ore learning to do and will get back to reading my printout of the PDF Sean -- ++ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | ++ smime.p7s Description: S/MIME Cryptographic Signature ___ 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
[Asterisk-Users] Patchy audio to and from VOIPBUSTER
I was playing with using VOIPBUSTER and was testing their client, which I think is SIP. So I added the only setup I could find for asterisk which is iax2 but I found that the speech quality is poor compared to the client and there is a delay of almost 1 second whereas their client there is not a real noticeable delay. Should I try with SIP. I am using ISDN 64K dialup if that makes any difference Just tried SIP and the same problem exists. Currently the Asterisk box is behind a Firewall on a ISDN dialup connection which sadly the IP changes. Is there anything that I can try Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie FreeWorldDial 689482 smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] Patchy audio to and from VOIPBUSTER
Sean Rima wrote: I was playing with using VOIPBUSTER and was testing their client, which I think is SIP. So I added the only setup I could find for asterisk which is iax2 but I found that the speech quality is poor compared to the client and there is a delay of almost 1 second whereas their client there is not a real noticeable delay. Should I try with SIP. I am using ISDN 64K dialup if that makes any difference Just tried SIP and the same problem exists. Currently the Asterisk box is behind a Firewall on a ISDN dialup connection which sadly the IP changes. Is there anything that I can try To my sip.conf I added the outside_addr=tcob1.no-ip.com, which I use for other things, but it is still very patchy, and the delay is now about 2 to 3 seconds Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie FreeWorldDial 689482 smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] Patchy audio to and from VOIPBUSTER
Sean Rima wrote: Sean Rima wrote: I was playing with using VOIPBUSTER and was testing their client, which I think is SIP. So I added the only setup I could find for asterisk which is iax2 but I found that the speech quality is poor compared to the client and there is a delay of almost 1 second whereas their client there is not a real noticeable delay. Should I try with SIP. I am using ISDN 64K dialup if that makes any difference Just tried SIP and the same problem exists. Currently the Asterisk box is behind a Firewall on a ISDN dialup connection which sadly the IP changes. Is there anything that I can try To my sip.conf I added the outside_addr=tcob1.no-ip.com, which I use for other things, but it is still very patchy, and the delay is now about 2 to 3 seconds Done a bit of research and discovered that there is nothign I can do until I get my new ISDN card for the Asterisk PC, cannot use the one on the Windows PC as it is USB and soft at that :( Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie FreeWorldDial 689482 smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] Problem with FWD connection rejected
John Fawcett wrote: Sean Rima wrote: Using the install instructions for [EMAIL PROTECTED], I setup a FWD account, this I tested using X-Lite and it works okay, Nowever I cannot make calls to fwd using Asterisk, my log showes: Aug 14 21:06:09 NOTICE[1324]: Registration of '689482' rejected: Registration Refused Aug 14 21:06:59 NOTICE[1324]: Registration of '689482' rejected: Registration Refused I'm new to this, but just a couple of thoughts: were you using SIP to connect to FWD from XTEN? Are you using SIP or IAX to connect to FWD from Asterisk? If you're using IAX, have you done the additional registration step needed by FWD to enable the IAX protocol? It might be worthwhile posting your configuration (sip.conf or iax.conf depending on what you're using). I have just scratched the setup and will be using the setup that is on the wiki to try it out, the only problem is that I am dialup and cannot use the externalip= setting, so have to work a way around that as yet Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] Problem with FWD connection rejected
Sean Rima wrote: Using the install instructions for [EMAIL PROTECTED], I setup a FWD account, this I tested using X-Lite and it works okay, Nowever I cannot make calls to fwd using Asterisk, my log showes: Aug 14 21:06:09 NOTICE[1324]: Registration of '689482' rejected: Registration Refused Well folks, I did it, sorted it out all on my ownsome :P) I did not enable the option at FWD to allow me to use IAX, once I enabled it, it worked :) Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie FreeWorldDial: 689482 smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] 8 FXS in Asterisk Server
Joseph wrote: Easy and cheap. Get two gateways AG-468 (each have 4 FXS ports) made by Atcom http://www.voip-info.org/tiki-index.php?page=Atcom one is about 88/ea I have two on the way and will let you know how it works. I would be interested in knowing how these work as well Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie FreeWorldDial 689482 smime.p7s Description: S/MIME Cryptographic Signature ___ 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
[Asterisk-Users] Problem with FWD connection rejected
Using the install instructions for [EMAIL PROTECTED], I setup a FWD account, this I tested using X-Lite and it works okay, Nowever I cannot make calls to fwd using Asterisk, my log showes: Aug 14 21:06:09 NOTICE[1324]: Registration of '689482' rejected: Registration Refused Aug 14 21:06:59 NOTICE[1324]: Registration of '689482' rejected: Registration Refused According the to info screen of [EMAIL PROTECTED] I have: IAX2 Sip Registry Host UsernamePerceived Refresh State 65.39.205.121:4569689482 60 Rejected -- Remote UNIX connection But the IAX2 Peers showes an OK connection. At the moment the asterisk box is behind a Windows XP box until I get it's own ISDN card and then it will have it's own connection, the FWD is for testing at the moment. I have tested dialing between extensions etc and that works. The full og showes: Aug 14 21:05:25 VERBOSE[1324]: -- Executing Dial(SIP/200-4acf, IAX2/fwd/612) in new stack Aug 14 21:05:25 VERBOSE[1324]: -- Called fwd/612 Aug 14 21:05:26 WARNING[1324]: Call rejected by 65.39.205.121: No authority found Aug 14 21:05:26 DEBUG[1324]: Immediately destroying 4, having received reject Aug 14 21:05:26 DEBUG[1324]: We're hanging up IAX2/fwd/4 now... Aug 14 21:05:26 DEBUG[1324]: Really destroying IAX2/fwd/4 now... Aug 14 21:05:26 VERBOSE[1324]: -- Hungup 'IAX2/fwd/4' Not sure what else to try Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ 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
[Asterisk-Users] One more newbie question
Ok, I am going for [EMAIL PROTECTED] with the CentOS iso disk. Installed and just checking a few things out. My other question is this, which I forgot to ask before. We have no Broadband here and more than likely will never have, so I am just looking at building Asterisk to handle inbound and outbound calls, at home via a ISDN card and for my hotel job via a PSTN line setup. Is this very complicated to setup or not? Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: ISDN Setup [was: Re: [Asterisk-Users] One more newbie question]
Tzafrir Cohen wrote: [ Subject changed so people looking at the list index will actually have the minimal clue as to what this post is about ]. On Sat, Aug 13, 2005 at 01:50:16PM +0100, Sean Rima wrote: Ok, I am going for [EMAIL PROTECTED] with the CentOS iso disk. Installed and just checking a few things out. And obviously you want us to do all of your work for you? Nope, I admit that I should have done a better subject line You don't even give a meaningful subject to your messages, so in the future the discussions that followed will be useless to searchers. I didn't even bother reading previous newbie question threads. My other question is this, which I forgot to ask before. We have no Broadband here and more than likely will never have, so I am just looking at building Asterisk to handle inbound and outbound calls, at home via a ISDN card and for my hotel job via a PSTN line setup. Is this very complicated to setup or not? No. I read an online doc and it was a great help, shold ahve searched before I posted the message Does that answer your question? Now go and do the minimal search: * What type of ISDN services is availble at your country/area? * Did you read a bit about the availble ISDN support in Asterisk? - It is generally better to ask questions that indicate you did and point to parts you don't understand, than just ask general qustions. Yeah true and I will bear this in mind in the future Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] One more newbie question
Tom Rymes wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Rima Sent: Saturday, August 13, 2005 8:50 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] One more newbie question Ok, I am going for [EMAIL PROTECTED] with the CentOS iso disk. Installed and just checking a few things out. My other question is this, which I forgot to ask before. We have no Broadband here and more than likely will never have, so I am just looking at building Asterisk to handle inbound and outbound calls, at home via a ISDN card and for my hotel job via a PSTN line setup. Is this very complicated to setup or not? Sean This is not that difficult. The first question you need to answer is whether you want to use a dedicated circuit (E1/T1/PRI) or multiple copper lines. This mostly depends on the call volume that you will be handling. Depending on your choice, you can install either a Sangoma or a Digium card to handle a dedicated circuit or you can use a Digium TDM400B card or a PSTN gateway to connect to your POTS phone lines. For my own needswhich is the primary one, it will just be connected to one channel of the ISDN circuit, the other channel is taken by the PC connection to the net. I have seen a web site that deals with a lot of my questions so I will be reading it in some detail Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] Newbie Question: Building an Asterisk system toreplace an old PBX but using existing phone
Michael Boger Jr wrote: Sean, What kind of hotel do you have? Some PMS vendors require the call accounting and check-in interfaces to their system. I am not aware that asterisk supports these serial interfaces. No they have no call accounting etc as such everything is done manually. I will work out printing at a later stage Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ 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
[Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone
I have a brief from a local hotel to build a PBX using Asterisk but they want to use their exisiting telephones and wiring from an old PBX that no longer works. Basically, I can build the system but an looking for a card that will allow for upto 20 extensions to be wired into the back of the PC. Doeas anyone know of a solution to this Sean-- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone
Chad Osmond wrote: To use the old phones and existing wiring you'll need some E1/T1 FXS Channel banks and a T1/E1 Card. Each bank will handle 30/24 phones and pipe them into a single E1/T1 connection. You can connect up to 4 (or 8 soon from Sangoma) T1's per card. I really like the Sangoma cards, there are also Digium cards as well. The Wiki will have a lot more information regarding Channel Banks and FXS adapters, I would suggest starting there. Thanks for this info, I forgot to check the wiki, I am trying to get them to use IP phones and ditch the old wiring anyway Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone
Tom Hayden wrote: Well, it's unlikely you're going to find a PCI card that can handle twenty analog lines, however I suggest you look at purchasing a call bank such as the adit 600. You then can link up your * server with the call bank using a T1 card and control and route calls using that method. I told them it would be easier and cheaper to ditch the old phones and wiring to go for dedicated Asterisk phones, I may still go this method as I need a few for myself anyway Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone
Andrew Kohlsmith wrote: On Thursday 11 August 2005 08:34, Sean Rima wrote: I have a brief from a local hotel to build a PBX using Asterisk but they want to use their exisiting telephones and wiring from an old PBX that no longer works. Can you plug one of the phones into a REGULAR telephone line and get dialtone and take and place calls? If the answer is yes, you can use a te110p (T1 card) and an FXS channel bank to connect the phones to Asterisk. If not, you're SOL unless you can find some kind of proprietary-to-standard phone interface, and the chances of that are slim to none. They are standard phones but I also want them to have all the features that Asterisk does provide, so I may build a bos for my house and show them that as well Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone
Andrew Kohlsmith wrote: On Thursday 11 August 2005 09:31, Sean Rima wrote: They are standard phones but I also want them to have all the features that Asterisk does provide, so I may build a bos for my house and show them that as well Standard phones can still do MWI (if they have a light), call transfers, three-way calling... all the good stuff that any Zap channel can provide. If they have displays and conform to ADSI they can even have soft buttons and so on. I have that at my house. Nope nothing like that only basic telephones Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone
Tom Rymes wrote: On Aug 11, 2005, at 10:35 AM, Sean Rima wrote: Andrew Kohlsmith wrote: On Thursday 11 August 2005 09:31, Sean Rima wrote: They are standard phones but I also want them to have all the features that Asterisk does provide, so I may build a bos for my house and show them that as well Standard phones can still do MWI (if they have a light), call transfers, three-way calling... all the good stuff that any Zap channel can provide. If they have displays and conform to ADSI they can even have soft buttons and so on. I have that at my house. Nope nothing like that only basic telephones Sean This may be heresy for some, but I would look into [EMAIL PROTECTED] for a reasonably sized hotel. It has wakeup calls weather built-in, easy for the hotel to configure, etc, and despite the home in the name, it is solid and robust. Contrary to popular belief, you can also extend it as needed by using the extensions_custom.conf file. I will have a look at that and see if it helps, byt the sounds itmay Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone
Tom Rymes wrote: On Aug 11, 2005, at 11:49 AM, Sean Rima wrote: Tom Rymes wrote: On Aug 11, 2005, at 10:35 AM, Sean Rima wrote: Andrew Kohlsmith wrote: On Thursday 11 August 2005 09:31, Sean Rima wrote: They are standard phones but I also want them to have all the features that Asterisk does provide, so I may build a bos for my house and show them that as well Sean, The client has a good idea in keeping the basic analog phones, since all of their guests know how to use them. If you put a SPA-841 or a Polycom IP301, you will likely intimidate the guests. This might sound silly to most of us techies, but picture a 74 year old guest just trying to call and let his children know he arrived safely. He might very well look at the Polycom and be confused, especially if he dials and has to press send, etc. Also, if they already have the phones, the per channel cost of using a T1 card and a channel bank is reasonably low, compared to a decent SIP hardphone. Not to mention that if you don't spend the extra money for a good hardphone, you are likely to have quality issues, as I have heard many quality complaints about many of the cheaper phones. Finally, sticking with the analog phones means you won't have to re- wire the whole place. I never thought of it this way, I will prive up a T1 card and bank, possibly on ebay as well as there are a a few items there for sale Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ 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
