[asterisk-users] Newbie Question...
Hello! Im new to Asterisk configuration and I have few questions regarding its configuration. I have 4 PSTN lines connected to TDM410. I can make exact 60 minutes of free calls from each of 4 pstn lines... Can I configure Asterisk to call thru pstn line that has free minutes? For example Outgoing calls are going through PSTN 1 for 60 minutes. When I use all of these free minutes - outgoing calls go thru PSTN 2. When I use all free minutes from PSTN 2 outgoing calls go via PSTN3. -- Piotrek Gorski -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Question...
On Mon, 31 Jan 2011, Piotr Górski wrote: I have 4 PSTN lines connected to TDM410. I can make exact 60 minutes of free calls from each of 4 pstn lines... Can I configure Asterisk to call thru pstn line that has free minutes? For example Outgoing calls are going through PSTN 1 for 60 minutes. When I use all of these free minutes - outgoing calls go thru PSTN 2. When I use all free minutes from PSTN 2 outgoing calls go via PSTN3. You will need to keep track of the call duration for each channel in a persistent store -- something like MySQL. You may also want to read up on setting the absolute timeout on a channel so a caller won't consume all of your 'prepaid' (nothing is free) minutes and drive you into unexpected charges. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie question on GSM adapter
Hi, I've recently installed Asterisk 1.6.2.13. I'd like to connect GSM Trunk to it. I purchased a few Mobigater ProOpen gateways. It states that I should use chan_celliax module to it. On the gsmopen site I see a comment in the documentation that I can install the module on Asterisk 1.2.x, 1.4.x, 1.6.0.x but not on the 1.6.1.x. Could somebody tell me if I can install it on my 1.6.2.13 version or I need to go back to 1.6.0.28 or the 1.4 line. If it works on my installation what should I change in the makefile. Thank you, Zoltán http://www.clamagent.org - Free Antivirus for Exchange http://www.it-pro.hu http://emaildetektiv.hu -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] newbie question
Hi All, When typing 'help' on the command line (* console) is there a way to keep it from just scrolling most of the information off the top of the screen? I can't hit ctrl-s fast enough so I miss most of the info. This makes 'help' be not much help. Thanks, Bill ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
You can tee your CLI screen (google for it) so your output is in a file that you can use more|less|vi or some other controlled viewing method on. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bill Shaw Sent: Tuesday, November 17, 2009 10:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] newbie question Hi All, When typing 'help' on the command line (* console) is there a way to keep it from just scrolling most of the information off the top of the screen? I can't hit ctrl-s fast enough so I miss most of the info. This makes 'help' be not much help. Thanks, Bill ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote: Hi All, When typing 'help' on the command line (* console) is there a way to keep it from just scrolling most of the information off the top of the screen? I can't hit ctrl-s fast enough so I miss most of the info. This makes 'help' be not much help. No. But you can either: 1. Use a terminal that has a long enough scroll-back buffer (or screen inside one that doesn't) 2. Run from the external shell prompt: asterisk -rx 'help whatever' | less -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote: Hi All, When typing 'help' on the command line (* console) is there a way to keep it from just scrolling most of the information off the top of the screen? I can't hit ctrl-s fast enough so I miss most of the info. This makes 'help' be not much help. On Tue, 17 Nov 2009, Tzafrir Cohen wrote: No. But you can either: 1. Use a terminal that has a long enough scroll-back buffer (or screen inside one that doesn't) 2. Run from the external shell prompt: asterisk -rx 'help whatever' | less Or, you can use the script command to capture the output to a file so you can refer to it as needed. script is also useful to capture the console log to a file when you are trying to debug a call and your console output looks like a broken fire hydrant. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
Option #2 is really the best option unless you need real time viewing of your help information (IMO). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Tuesday, November 17, 2009 10:59 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] newbie question On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote: Hi All, When typing 'help' on the command line (* console) is there a way to keep it from just scrolling most of the information off the top of the screen? I can't hit ctrl-s fast enough so I miss most of the info. This makes 'help' be not much help. No. But you can either: 1. Use a terminal that has a long enough scroll-back buffer (or screen inside one that doesn't) 2. Run from the external shell prompt: asterisk -rx 'help whatever' | less -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Bill Shaw Sent: Tuesday, November 17, 2009 11:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] newbie question When typing 'help' on the command line (* console) is there a way to keep it from just scrolling most of the information off the top of the screen? I can't hit ctrl-s fast enough so I miss most of the info. This makes 'help' be not much help. Another option is to use 'screen' and use the integrated scroll back buffer. I'm pretty lazy so most of my servers have established screen sessions with consoles, logs, mysql, etc. already running that I simply reconnect to. sl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
On Tue, Nov 17, 2009 at 09:09:39AM -0800, Steve Edwards wrote: On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote: Hi All, [snip] 2. Run from the external shell prompt: asterisk -rx 'help whatever' | less Or, you can use the script command to capture the output to a file so you can refer to it as needed. I find screen helpful here you can set the scroll back buffer to a large number and you can detach the running screen from the console to reattach some time later. my default scroll back buffer is set to around 1000 usually enough to capture what I need, plus you can cut paste between screens script is also useful to capture the console log to a file when you are trying to debug a call and your console output looks like a broken fire hydrant. -- It's amazing I won. I was running against peace, prosperity, and incumbency. - George W. Bush 06/14/2001 speaking to Swedish Prime Minister Goran Perrson, unaware that a live television camera was still rolling. signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
When typing 'help' on the command line (* console) is there a way to keep it from just scrolling most of the information off the top of the screen? I can't hit ctrl-s fast enough so I miss most of the info. This makes 'help' be not much help. my default scroll back buffer is set to around 1000 usually enough to capture what I need, plus you can cut paste between screens You could also make it much simpler and just set your verbosity very low or just turn it off, so there are very few messages coming across your screen. Unless you're on a really busy machine, you should be able to read most of the help screens. core set verbose 0 - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
On Tue, 17 Nov 2009, Noah Miller wrote: You could also make it much simpler and just set your verbosity very low or just turn it off, so there are very few messages coming across your screen. Unless you're on a really busy machine, you should be able to read most of the help screens. core set verbose 0 Unfortunately, when your boss comes in and says Why did this just* happen?, those logs are kind of handy. I like a lot of logging on production systems. I funnel everything from every server to a single loghost via syslog. First thing every morning, a cron job bzip2s the previous days syslog file and saves it as syslog.bz2-$(date +%d) so I always have 30 days logs on tap and don't have to worry about deleting old log files. *) Sometime in the last 30 days. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie, Question on making a PSTN call..
I understand the desire to try, but you are trying too hard. Getting a soft modem to work with Asterisk is. like trying to push a string up a 10 foot pipe. At the least, buy an inexpensive FXO device from someone like Grandstream and use it via Ethernet to work with Asterisk. If you have greater ambitions, buy any appropriate piece of hardware and start with that. Otherwise, You are going to have a lot of string in that pipe, before you see any come out the top. You won't get help on this because no one really knows how to do it or if it will work at all. I am trying to help, by getting you to try a better way. Good luck. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shiva Kumar Sent: Tuesday, June 16, 2009 12:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Newbie, Question on making a PSTN call.. Need help pls..Anyone? On Mon, Jun 15, 2009 at 10:58 PM, Shiva Kumar shi...@gmail.com wrote: Hello Asterisk-users, I am new to Asterisk. I got SIP Calls to work between two computers using a soft phone and asterisk in the middle. Since then, I have been trying to get my soft phone to make a PSTN call with terrible failure for about two days now. On Windows using asteriskwin32: I have a soft modem (Agere HDA Modem) in my laptop. Windows' default dialer is able to make a PSTN call by connecting the Phone's RJ line into my laptop's RJ 11. I am unsure what drivers to choose where and what parameters to change in tapi/fx configuration files etc. to get asterisk to use this modem to call out. Read plenty of articles about how asterisk cannot make a good phone call using a half duplex modem. But, This is for experimental purposes and I will be thrilled to just get my phone ringing before I go out to buy specific hardware. On my Ubuntu: Next up, I connected my phone(NOKIA N73) to my computer, ensured that I am able to connect to internet on my ubuntu. wvdial works good too. Again, I am unsure how to get asterisk to connect to this modem so that I can use my soft phones to make a call. Need help. Thanks in Advance. -- Shivku, http://blog.shivku.com -- Shivku, http://blog.shivku.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie, Question on making a PSTN call..
On Mon, 15 Jun 2009, Shiva Kumar wrote: Hello Asterisk-users, I am new to Asterisk. I got SIP Calls to work between two computers using a soft phone and asterisk in the middle. Since then, I have been trying to get my soft phone to make a PSTN call with terrible failure for about two days now. On Windows using asteriskwin32: I have a soft modem (Agere HDA Modem) in my laptop. Windows' default dialer is able to make a PSTN call by connecting the Phone's RJ line into my laptop's RJ 11. I am unsure what drivers to choose where and what parameters to change in tapi/fx configuration files etc. to get asterisk to use this modem to call out. Read plenty of articles about how asterisk cannot make a good phone call using a half duplex modem. But, This is for experimental purposes and I will be thrilled to just get my phone ringing before I go out to buy specific hardware. Go out and buy specific hardware. OpenVox are really cheap these days. Well under £100 for a card with an FXO interface now. Gordon___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie, Question on making a PSTN call..
Hello Asterisk-users, I am new to Asterisk. I got SIP Calls to work between two computers using a soft phone and asterisk in the middle. Since then, I have been trying to get my soft phone to make a PSTN call with terrible failure for about two days now. On Windows using asteriskwin32: I have a soft modem (Agere HDA Modem) in my laptop. Windows' default dialer is able to make a PSTN call by connecting the Phone's RJ line into my laptop's RJ 11. I am unsure what drivers to choose where and what parameters to change in tapi/fx configuration files etc. to get asterisk to use this modem to call out. Read plenty of articles about how asterisk cannot make a good phone call using a half duplex modem. But, This is for experimental purposes and I will be thrilled to just get my phone ringing before I go out to buy specific hardware. On my Ubuntu: Next up, I connected my phone(NOKIA N73) to my computer, ensured that I am able to connect to internet on my ubuntu. wvdial works good too. Again, I am unsure how to get asterisk to connect to this modem so that I can use my soft phones to make a call. Need help. Thanks in Advance. -- Shivku, http://blog.shivku.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie, Question on making a PSTN call..
Need help pls..Anyone? On Mon, Jun 15, 2009 at 10:58 PM, Shiva Kumar shi...@gmail.com wrote: Hello Asterisk-users, I am new to Asterisk. I got SIP Calls to work between two computers using a soft phone and asterisk in the middle. Since then, I have been trying to get my soft phone to make a PSTN call with terrible failure for about two days now. On Windows using asteriskwin32: I have a soft modem (Agere HDA Modem) in my laptop. Windows' default dialer is able to make a PSTN call by connecting the Phone's RJ line into my laptop's RJ 11. I am unsure what drivers to choose where and what parameters to change in tapi/fx configuration files etc. to get asterisk to use this modem to call out. Read plenty of articles about how asterisk cannot make a good phone call using a half duplex modem. But, This is for experimental purposes and I will be thrilled to just get my phone ringing before I go out to buy specific hardware. On my Ubuntu: Next up, I connected my phone(NOKIA N73) to my computer, ensured that I am able to connect to internet on my ubuntu. wvdial works good too. Again, I am unsure how to get asterisk to connect to this modem so that I can use my soft phones to make a call. Need help. Thanks in Advance. -- Shivku, http://blog.shivku.com -- Shivku, http://blog.shivku.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
Am Samstag, den 15.12.2007, 16:55 -0800 schrieb Philip Prindeville: I've got the following set up: Someone calls into my PBX on a single number (via SIP trunk from my carrier), and the get a voice menu of extensions. On one of the extensions, it rings a bunch of internal SIP hardphones, plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN gateway. The issue is that my cellphone shows my PBX's number, not the original calling number. This topic has been covered in length. In most cases it seems to end at the fact that providers correct caller-ids they get from the calling party: If you send any number which is assigned to the PRI (or SIP trunk), that is fine; if you send another number, it will be changed to the (first) number of the PRI/trunk. Few providers allow for foreign caller ids to be sent over their equipment - in some countries this is even illegal. For example, one of my providers (German) allows to set any CALLERID, but their documentation warns to not do stupid tricks, as calls can be tracked and using malicious information will be prosecuted. This feature is to be used only for sending _my_ cell phone number etc. BR Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
Anselm Martin Hoffmeister wrote: In most cases it seems to end at the fact that providers correct caller-ids they get from the calling party: If you send any number which is assigned to the PRI (or SIP trunk), that is fine; if you send another number, it will be changed to the (first) number of the PRI/trunk. Few providers allow for foreign caller ids to be sent over their equipment - in some countries this is even illegal. For example, one of my providers (German) allows to set any CALLERID, but their documentation warns to not do stupid tricks, as calls can be tracked and using malicious information will be prosecuted. This feature is to be used only for sending _my_ cell phone number etc. Do you know of any GSM providers/contracts where faking for a valid reason is possible? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
Philipp Kempgen wrote: Anselm Martin Hoffmeister wrote: In most cases it seems to end at the fact that providers correct caller-ids they get from the calling party: If you send any number which is assigned to the PRI (or SIP trunk), that is fine; if you send another number, it will be changed to the (first) number of the PRI/trunk. Few providers allow for foreign caller ids to be sent over their equipment - in some countries this is even illegal. For example, one of my providers (German) allows to set any CALLERID, but their documentation warns to not do stupid tricks, as calls can be tracked and using malicious information will be prosecuted. This feature is to be used only for sending _my_ cell phone number etc. Do you know of any GSM providers/contracts where faking for a valid reason is possible? Regards, Philipp Kempgen I can think of some... in rural Idaho, cell coverage is sparse. I might check my voice mail of my cell phone via a land line, and want to call back with a response originating from my cell number... Also the case if I have my cell set to forward-on-busy to my land line, or if I'm hosting an answering service, etc. -Philip ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
Philip Prindeville wrote: Philipp Kempgen wrote: Do you know of any GSM providers/contracts where faking for a valid reason is possible? I can think of some... in rural Idaho, cell coverage is sparse. I might check my voice mail of my cell phone via a land line, and want to call back with a response originating from my cell number... Also the case if I have my cell set to forward-on-busy to my land line, or if I'm hosting an answering service, etc. What I'm looking for is this scenario: I call someone's cell phone number via my GSM gateway (to save money). But I'd like to set my landline number as the callerid (instead of one of the numbers of the GSM gateway or no callerid). Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
I've got the following set up:
Someone calls into my PBX on a single number (via SIP trunk from my
carrier), and the get a voice menu of extensions.
On one of the extensions, it rings a bunch of internal SIP hardphones,
plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN
gateway.
The issue is that my cellphone shows my PBX's number, not the original
calling number.
My dialplan looks like:
[globals]
...
TRUNK=SIP/sip_proxy-out
CELL=${TRUNK}/208xxx
PHILIP=SIP/bedroom_1SIP/office_2SIP/kitchen_1${CELL}
[incoming]
exten = s,1,Answer()
; sometimes signaling and media get out of sync on cell networks...
exten = s,n,Wait(0.75)
exten = s,n,Playback(main-menu)
exten = s,n(exten),Background(vm-enter-number-to-call)
exten = s,n,WaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup
...
exten = 111,1,Macro(stdexten,111,${PHILIP})
exten = 111,n,Goto(s,exten)
exten = 112,1,Macro(stdexten,112,${REDFISH})
exten = 112,n,Goto(s,exten)
...
exten = i,1,Playback(pbx-invalid)
exten = i,n,Goto(s,exten)
exten = t,1,Goto(s,goodbye)
Ok, so far, so good.
The problem is that when we hit Macro(stdexten,111,${PHILIP}) and it
does the Dial(${PHILIP}) which includes the
SIP/sip_proxy-out/208xxx, 208xxx rings with my PBX's extension.
Oddly, the internal phones ring with outside caller's extension.
[sip_proxy-out]
type=peer
fromuser=208nnn
fromdomain=x.x.x.x
host=y.y.y.y
call-limit=5
nat=yes
So I'm not setting the callerid on the peer by default. What am I
missing? Do I need to modify the stdexten macro to dial with the 'o'
option? Or can I set this explicitly with a 'Set' before calling the
macro? Or do I need to be missing with the RDNIS?
Oh, I'm running Asterisk 1.2.25... (yes, I'll upgrade when AstLinux
upgrades).
-Philip
P.S. I tried adding |o to the end of the PHILIP variable, but this
didn't seem to make a difference.
___
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Re: [asterisk-users] Newbie Question about E1
Hi PaulH, thanks for your answer! Now, another question.. Every E1 card has support for pri_net/pri_cpe or only some of them has? Can you tell me at least one card that can do that? Thanks again. -- Luar Roji On Fri, Apr 20, 2007 at 03:32:45PM +1000, Paul Hales wrote: Your best bet is a dual port E1 card - set one side to pri_net and the other to pri_cpe. PaulH On Fri, 2007-04-20 at 01:52 -0300, Luar Roji wrote: Hi everybody. I'm about to ask a newbie question, be warned! I have a NEC 2000 IPS PBX connected to a E1. Now I want to set up an asterisk, with some digium card connected to that E1. (suggestions about the card? I'll have maybe another E1 more). The newbie question is.. How can I connect the asterisk PC with the PBX? I don't know so much about E1, so I don't know if a E1 card in the PC can do as a telephone company, or you can't do that.. I'm clear with the question? I'll make a little diagram: This is what I have now. +---++---+ --- phone 1 | ANTEL ||NEC PBX| --- phone 2 +---+ E1 +---+ --- ... What I want is: +---+++ +---+ --- phone 1 | ANTEL ||Asterisk box|---|NEC PBX| --- phone 2 +---+ E1 +| ? +---+ --- ... (ANTEL is my local phone company) Thanks! Another not very related question.. this NEC PBX says Internet Protocol Server. Is there a way to connect it to the asterisk? Thanks again! -- Luar Roji ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie Question about E1
Hi everybody. I'm about to ask a newbie question, be warned! I have a NEC 2000 IPS PBX connected to a E1. Now I want to set up an asterisk, with some digium card connected to that E1. (suggestions about the card? I'll have maybe another E1 more). The newbie question is.. How can I connect the asterisk PC with the PBX? I don't know so much about E1, so I don't know if a E1 card in the PC can do as a telephone company, or you can't do that.. I'm clear with the question? I'll make a little diagram: This is what I have now. +---++---+ --- phone 1 | ANTEL ||NEC PBX| --- phone 2 +---+ E1 +---+ --- ... What I want is: +---+++ +---+ --- phone 1 | ANTEL ||Asterisk box|---|NEC PBX| --- phone 2 +---+ E1 +| ? +---+ --- ... (ANTEL is my local phone company) Thanks! Another not very related question.. this NEC PBX says Internet Protocol Server. Is there a way to connect it to the asterisk? Thanks again! -- Luar Roji ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Question about E1
Your best bet is a dual port E1 card - set one side to pri_net and the other to pri_cpe. PaulH On Fri, 2007-04-20 at 01:52 -0300, Luar Roji wrote: Hi everybody. I'm about to ask a newbie question, be warned! I have a NEC 2000 IPS PBX connected to a E1. Now I want to set up an asterisk, with some digium card connected to that E1. (suggestions about the card? I'll have maybe another E1 more). The newbie question is.. How can I connect the asterisk PC with the PBX? I don't know so much about E1, so I don't know if a E1 card in the PC can do as a telephone company, or you can't do that.. I'm clear with the question? I'll make a little diagram: This is what I have now. +---++---+ --- phone 1 | ANTEL ||NEC PBX| --- phone 2 +---+ E1 +---+ --- ... What I want is: +---+++ +---+ --- phone 1 | ANTEL ||Asterisk box|---|NEC PBX| --- phone 2 +---+ E1 +| ? +---+ --- ... (ANTEL is my local phone company) Thanks! Another not very related question.. this NEC PBX says Internet Protocol Server. Is there a way to connect it to the asterisk? Thanks again! -- Luar Roji ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Question
On 3/8/07, Chris Nighswonger [EMAIL PROTECTED] wrote: Thanks for the responses. iptables on the * box has no rules and all tables default to 'accept.' I have not got to the point of placing calls out across the internet yet. The issue here is no audio back from the * box when running through the demo routine. I'll try to set it up to make a call outside tomorrow. Ok. I have not been able to setup the box to call outside, however, watching the packet traffic I see plenty of data flowing from the xlite client to the * server, but never any packets from the server to the client. (That is, during the course of the call.) The server and client talk just fine when establishing the connection, just no audio data from the server to the client. Any thoughts? From everything I've read, the initial setup should be much easier than mine has gone so far... :( Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Question
On 3/15/07, Chris Nighswonger [EMAIL PROTECTED] wrote: Ok. I have not been able to setup the box to call outside, however, watching the packet traffic I see plenty of data flowing from the xlite client to the * server, but never any packets from the server to the client. (That is, during the course of the call.) The server and client talk just fine when establishing the connection, just no audio data from the server to the client. Any thoughts? Setup the demo IVR on your Atrisk box and call that from your xlite softphone. The entire call will be on your local network so you'll be able to see if the problem is local or not. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Question
[test] disallow=all allow=gsm ;GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw Are you sure that the xlite phone can handle gsm?? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Question
On 3/9/07, mail-lists [EMAIL PROTECTED] wrote: [test] disallow=all allow=gsm ;GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw Are you sure that the xlite phone can handle gsm?? I use it on Linux and it does. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie Question
Hi all, I'm new to Astrisk so bear with me. I have just installed AsteriskNOW and am quite familiar with RH Linux. I have configured it and am using Xlite to connect and learn to move around the conf files. I have a problem, however. The client connects and dials ok, but there is no audio. In searching the archives I found discussion of this issue primarily centered on NAT issues. This is not my issue (I think). Here is some info: 1. * server and clients are all on the same subnet but are separated from the internet by a proxy/firewall which forces all port 80 traffic through the proxy. 2. The server has a single channel fxo card. 3. Snip of sip.conf: [test] type=friend secret=verysecret regexten=1234 ; When they register, create extension 1234 callerid=Test Unit 1234 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw [EMAIL PROTECTED]; Subscribe to status of multiple mailboxes context=internal Here is the problem: Xlite registers fine. When I dial 500 to access the demo, the * console shows the client connect and the demo audio plays. However, there is no sound on the client end. I have installed Xlite on an XP workstation and on a *nix workstation. Both installs behave the same. Any thoughts? Or do I need to post more details? Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Question
If both the asterisk server and the softphone are on the same LAN then I would look at your firewall settings on the box. Make sure you have 5060 and 10,000 - 20,000 UDP open. If the phone is connecting to the server over the internet and the server IS behind NAT then you need to forward ports 5060 and 10,000-20,000 UDP to the asterisk server. - Original Message - From: Chris Nighswonger [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, March 09, 2007 1:16 AM Subject: [asterisk-users] Newbie Question Hi all, I'm new to Astrisk so bear with me. I have just installed AsteriskNOW and am quite familiar with RH Linux. I have configured it and am using Xlite to connect and learn to move around the conf files. I have a problem, however. The client connects and dials ok, but there is no audio. In searching the archives I found discussion of this issue primarily centered on NAT issues. This is not my issue (I think). Here is some info: 1. * server and clients are all on the same subnet but are separated from the internet by a proxy/firewall which forces all port 80 traffic through the proxy. 2. The server has a single channel fxo card. 3. Snip of sip.conf: [test] type=friend secret=verysecret regexten=1234 ; When they register, create extension 1234 callerid=Test Unit 1234 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw [EMAIL PROTECTED]; Subscribe to status of multiple mailboxes context=internal Here is the problem: Xlite registers fine. When I dial 500 to access the demo, the * console shows the client connect and the demo audio plays. However, there is no sound on the client end. I have installed Xlite on an XP workstation and on a *nix workstation. Both installs behave the same. Any thoughts? Or do I need to post more details? Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Question
Don't forget about 4569 UDP port (IAX protocol) forwarded to your Asterisk box. Best Regards; Leonardo Kamache On 3/8/07, Dovid B [EMAIL PROTECTED] wrote: If both the asterisk server and the softphone are on the same LAN then I would look at your firewall settings on the box. Make sure you have 5060 and 10,000 - 20,000 UDP open. If the phone is connecting to the server over the internet and the server IS behind NAT then you need to forward ports 5060 and 10,000-20,000 UDP to the asterisk server. - Original Message - From: Chris Nighswonger [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, March 09, 2007 1:16 AM Subject: [asterisk-users] Newbie Question Hi all, I'm new to Astrisk so bear with me. I have just installed AsteriskNOW and am quite familiar with RH Linux. I have configured it and am using Xlite to connect and learn to move around the conf files. I have a problem, however. The client connects and dials ok, but there is no audio. In searching the archives I found discussion of this issue primarily centered on NAT issues. This is not my issue (I think). Here is some info: 1. * server and clients are all on the same subnet but are separated from the internet by a proxy/firewall which forces all port 80 traffic through the proxy. 2. The server has a single channel fxo card. 3. Snip of sip.conf: [test] type=friend secret=verysecret regexten=1234 ; When they register, create extension 1234 callerid=Test Unit 1234 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw [EMAIL PROTECTED]; Subscribe to status of multiple mailboxes context=internal Here is the problem: Xlite registers fine. When I dial 500 to access the demo, the * console shows the client connect and the demo audio plays. However, there is no sound on the client end. I have installed Xlite on an XP workstation and on a *nix workstation. Both installs behave the same. Any thoughts? Or do I need to post more details? Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Question
Thanks for the responses. iptables on the * box has no rules and all tables default to 'accept.' I have not got to the point of placing calls out across the internet yet. The issue here is no audio back from the * box when running through the demo routine. I'll try to set it up to make a call outside tomorrow. Chris On 3/8/07, Leonardo Kamache (Gmail) [EMAIL PROTECTED] wrote: Don't forget about 4569 UDP port (IAX protocol) forwarded to your Asterisk box. Best Regards; Leonardo Kamache On 3/8/07, Dovid B [EMAIL PROTECTED] wrote: If both the asterisk server and the softphone are on the same LAN then I would look at your firewall settings on the box. Make sure you have 5060 and 10,000 - 20,000 UDP open. If the phone is connecting to the server over the internet and the server IS behind NAT then you need to forward ports 5060 and 10,000-20,000 UDP to the asterisk server. - Original Message - From: Chris Nighswonger [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, March 09, 2007 1:16 AM Subject: [asterisk-users] Newbie Question Hi all, I'm new to Astrisk so bear with me. I have just installed AsteriskNOW and am quite familiar with RH Linux. I have configured it and am using Xlite to connect and learn to move around the conf files. I have a problem, however. The client connects and dials ok, but there is no audio. In searching the archives I found discussion of this issue primarily centered on NAT issues. This is not my issue (I think). Here is some info: 1. * server and clients are all on the same subnet but are separated from the internet by a proxy/firewall which forces all port 80 traffic through the proxy. 2. The server has a single channel fxo card. 3. Snip of sip.conf: [test] type=friend secret=verysecret regexten=1234 ; When they register, create extension 1234 callerid=Test Unit 1234 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw [EMAIL PROTECTED]; Subscribe to status of multiple mailboxes context=internal Here is the problem: Xlite registers fine. When I dial 500 to access the demo, the * console shows the client connect and the demo audio plays. However, there is no sound on the client end. I have installed Xlite on an XP workstation and on a *nix workstation. Both installs behave the same. Any thoughts? Or do I need to post more details? Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Chris Nighswonger Network Systems Director Foundations Bible College Seminary www.foundations.edu www.fbcradio.org [EMAIL PROTECTED] V:910-892-8761 C:919-820-5473 - NOTICE: The information contained in this electronic mail message is intended only for the use of the intended recipient, and may also be protected by the Electronic Communications Privacy Act, 18 USC Sections 2510-2521. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please reply to the sender, and delete the original message. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie question: How to config rtp packetization in 1.4?
Hi, any one test rtp packetization in 1.4?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] newbie question
Hello,I know little to nothing about asterisk. I am currently reading info online but was looking for other links as to good places to start. I basically just need to use asterisk to create a prototype. I need to demo a system that will allow the user to call a number, enter in some data, the backend then queries our db and returns results to the user via the phone. I've been told this can be done with asterisk so need to get started. This may never turn into an actual project so just need the minimal amount of work now to get it working. Any way to use a softphone or whatever to call, have the PBX prompt for info, receive it and then query the db and read the results to the user is what I want to do. The back end code will probably be PHP but can be in something else if needed. I am currently looking here;http://www.voip-info.org/tiki-index.php?page=Asterisk+AGIAre there other places to start? Is there a place to get an asterisk box/number set up for testing? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question about meetme
On 22:48, Sat 14 Oct 06, Rajeev Natarajan wrote: yes to ztdummy: but you may have trouble when you try and run multiple simultaneous meetme sessions. On 10/5/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: omar parihuana wrote: Is possible use meetme feature without Zaptel card? (ztdummy will be the solution? ) Yup. :P I switched from app_meetme.so to app_conference.so couple of weeks ago and was stunned with the quality of app_conference.so No ztdummy needed so finally my OpenBSD boxen can run meetme and iax2 trunks :) If you dont have zaptel hardware and are experiencing trouble with multiple meetme running the same time (like me) try the app_conference.so I'm in no way involved in this project, it just made my life easier. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question about meetme
yes to ztdummy: but you may have trouble when you try and run multiple simultaneous meetme sessions.On 10/5/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:omar parihuana wrote: Is possible use meetme feature without Zaptel card? (ztdummy will be the solution? )Yup. :P Thanks in advanced..--Mojo [EMAIL PROTECTED]Office Manager, Horan Company, LLC(907) 747- x112___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie question about meetme
Hi Folks, I'm reading about meetme feature, but in accordance to voip-info it say: A zaptel interface must be installed for conferencing to work. Unfortunately I don't have a Zaptel Card, I'm using Asterisk like SIP server then I would like to implement meetme function. What can I do? Is possible use meetme feature without Zaptel card? (ztdummy will be the solution? ) Thanks in advanced.. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question about meetme
yep, # modprobe ztdummy You need some special routines compiled in the kernel, google around a bit to find wich ones. Other solution may be use app_conference, is not included in asterisk sources, that app does not require zaptel timing. Regards On 10/4/06, omar parihuana [EMAIL PROTECTED] wrote: Hi Folks, I'm reading about meetme feature, but in accordance to voip-info it say: A zaptel interface must be installed for conferencing to work. Unfortunately I don't have a Zaptel Card, I'm using Asterisk like SIP server then I would like to implement meetme function. What can I do? Is possible use meetme feature without Zaptel card? (ztdummy will be the solution? ) Thanks in advanced.. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question about meetme
omar parihuana wrote: Is possible use meetme feature without Zaptel card? (ztdummy will be the solution? ) Yup. :P Thanks in advanced.. -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts
Thanks for the help!
What I have gathered mentally so far is that asterisk can't do
exactly what I am asking/expecting it to do.
Problem being that I am trying to get multiple inbound contexts
from multiple peers ( 3 of them in sip.conf) from one single provider.
What happens is that it matches the first peer (for my provider) and
never matches the next two that I also want to use.
Seems that it will only do a match based on IP Address/Host and not
on accountname or incoming phone number.
The help I have recieved here has not really addressed the origial question
of how to get the calls to come directly into a context from the sip peer
itself, however they have pointed out some work-arounds to what asterisk
seemingly does not support doing directly.
If I am wrong with this conclusion please help me out!
I have been able to accomplish what is needed by simply having an
initial context that everything comes into (possible security issue)
and then immediately issue a Goto() to get the call into the context where
it belongs.
This 'feels' very hokey and wrong, but it works for now!
Thanks for the help!
Take care!
Steve
What I do is the following and keep in mind I only use one register
statement with my provider:
exten = 18665551234,1,SetVar(FROM_DID=18665551234) ;
exten = 18665551234,2,Goto(from-pstn,s,1);
exten = 5185551234,1,SetVar(FROM_DID=5185551234) ;
exten = 5185551234,2,Goto(custom-callid,s,1) ;
On 4/2/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
I'm not an expert, but as far as i know, your incoming calls will
arrive with DID in ${EXTEN}
so the only thing you need is:
exten = 1234,1,GoTo(context1,1234,1) ; example for context extension
and priority
exten = 2345,1,GoTo(context2,2345,1)
exten = 3456,1,GoTo(context3,3456,1)
Be sure that you have created context1 context2 and context3 in your
extensions.conf
And in this context1 context2 and context3 you must have handler for
1234; 2345; and 3456;
example:
[context1]
exten = 1234,1,Answer()
exten = 1234,2,Playback(vm-goodbye)
exten = 1234,3,Hangup()
I didn't test this code, but this is my tip the main idea is that you
need to catch de DID and make a GoTo for the context you want.
Best regards,
Marco Mouta
On 4/2/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Gladden wrote:
What version of asterisk? (been lots of changes happening to the
sip
code over the last year)
SVN-branch-1.2-r9156
Have you looked at the sample configs in /usr/src/asterisk/configs?
Yes I have and my own configs are pretty much copies of them.
They do not detail, do or explain the simple concept that I am
trying to accomplish.
If they do I don't see it.
#1 I have more than one incoming SIP account
#2 I would like to have them come into the context of
my choice when a call comes in.
HOW do I do this?
currently I have 3 register lines
there is no way to specify in a register line
some way of making the call start in any other context
other than what is specified in the [general] section
of sip.conf
It seems that somehow maybe if there is a peer tat is somehow
matched to the register line (how???) it may work.
There may be some crazy way to do this within a peer
if so this is the information I am looking for...
The examples and descriptions are not at all clear to me
I have 3 accounts with the same provider
How do I get incoming calls to come into three different contexts
that I will create is the question.
From the example file I see:
Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register = user[:secret[:[EMAIL PROTECTED]:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension
needs to
; be defined in extensions.conf to be able to accept calls from this
SIP
proxy
I actually need to do 3 of these.
;register = 2345:[EMAIL PROTECTED]/1234
;
;Register 2345 at sip provider 'sip_proxy'. Calls from this
provider
;connect to local extension 1234 in extensions.conf, default
context,
;unless you configure a [sip_proxy] section below, and configure
a
;context.
Ok I have 3 accounts from the same provider
one [sip_proxy] section just puts me in the same problem boat I'm
already
in using a register line
the calls some into the context specified in [general] section of
sip.conf
I need to somehow differentiate the three SIP 'lines' and give
them different contexts to start in.
;Tip 1: Avoid assigning hostname to a sip.conf section like
[provider.com]
OK sure then how will this associate with my register line that
uses provider.com
This makes no sense to me...
I mean It really makes no sense...
Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts
Hi,
If you don't specify a host= statement in sip.conf and you have a
section that includes a username and secret plus type=peer, it will
match on username and secret. (That implies that if you have three
different numbers registered with your sip provider all under one
username, calls for all three will match the first section in
sip.conf
that contains that username and secret.)
Thank you for this tidbit as well.
It seems that I need the host= to actually be there for it to work though.
I've always used the same [peer] for incoming and outgoing calls,
If I get rid of the host= outgoing calls of course stop working.
This seems to be a strong hint that I need to explore using seperate peers
for incoming and outbound calls.
Put all the incoming peers first so they are not matched by host first and
then have the others at the bottom for outbound calls.
I will give this a try,
Thanks!
Steve
Thanks for the help!
What I have gathered mentally so far is that asterisk can't do
exactly what I am asking/expecting it to do.
Problem being that I am trying to get multiple inbound contexts
from multiple peers ( 3 of them in sip.conf) from one single provider.
What happens is that it matches the first peer (for my provider) and
never matches the next two that I also want to use.
Seems that it will only do a match based on IP Address/Host and not on
accountname or incoming phone number.
The help I have recieved here has not really addressed the origial question
of how to get the calls to come directly into a context from the sip
peer itself, however they have pointed out some work-arounds to what
asterisk seemingly does not support doing directly.
If I am wrong with this conclusion please help me out!
I have been able to accomplish what is needed by simply having an
initial context that everything comes into (possible security issue) and
then immediately issue a Goto() to get the call into the context where
it belongs.
This 'feels' very hokey and wrong, but it works for now!
Thanks for the help!
Take care!
Steve
What I do is the following and keep in mind I only use one register
statement with my provider:
exten = 18665551234,1,SetVar(FROM_DID=18665551234) ;
exten = 18665551234,2,Goto(from-pstn,s,1) ;
exten = 5185551234,1,SetVar(FROM_DID=5185551234);
exten = 5185551234,2,Goto(custom-callid,s,1);
On 4/2/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
I'm not an expert, but as far as i know, your incoming calls will
arrive with DID in ${EXTEN}
so the only thing you need is:
exten = 1234,1,GoTo(context1,1234,1) ; example for context extension
and priority
exten = 2345,1,GoTo(context2,2345,1)
exten = 3456,1,GoTo(context3,3456,1)
Be sure that you have created context1 context2 and context3 in your
extensions.conf
And in this context1 context2 and context3 you must have handler for
1234; 2345; and 3456;
example:
[context1]
exten = 1234,1,Answer()
exten = 1234,2,Playback(vm-goodbye)
exten = 1234,3,Hangup()
I didn't test this code, but this is my tip the main idea is that you
need to catch de DID and make a GoTo for the context you want.
Best regards,
Marco Mouta
On 4/2/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Gladden wrote:
What version of asterisk? (been lots of changes happening to the
sip
code over the last year)
SVN-branch-1.2-r9156
Have you looked at the sample configs in
/usr/src/asterisk/configs?
Yes I have and my own configs are pretty much copies of them. They
do not detail, do or explain the simple concept that I am trying
to accomplish.
If they do I don't see it.
#1 I have more than one incoming SIP account
#2 I would like to have them come into the context of
my choice when a call comes in.
HOW do I do this?
currently I have 3 register lines
there is no way to specify in a register line
some way of making the call start in any other context
other than what is specified in the [general] section
of sip.conf
It seems that somehow maybe if there is a peer tat is somehow
matched to the register line (how???) it may work.
There may be some crazy way to do this within a peer
if so this is the information I am looking for...
The examples and descriptions are not at all clear to me
I have 3 accounts with the same provider
How do I get incoming calls to come into three different contexts
that I will create is the question.
From the example file I see:
Asterisk can register as a SIP user agent to a SIP proxy
(provider)
; Format for the register statement is:
; register =
user[:secret[:[EMAIL PROTECTED]:port][/extension]
;
; If no extension is given, the 's' extension is used. The
extension
needs to
; be defined in extensions.conf to be able to accept calls from
this
SIP
proxy
I actually need
Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts
Steve Gladden wrote: What version of asterisk? (been lots of changes happening to the sip code over the last year) SVN-branch-1.2-r9156 Have you looked at the sample configs in /usr/src/asterisk/configs? Yes I have and my own configs are pretty much copies of them. They do not detail, do or explain the simple concept that I am trying to accomplish. If they do I don't see it. #1 I have more than one incoming SIP account #2 I would like to have them come into the context of my choice when a call comes in. HOW do I do this? currently I have 3 register lines there is no way to specify in a register line some way of making the call start in any other context other than what is specified in the [general] section of sip.conf It seems that somehow maybe if there is a peer tat is somehow matched to the register line (how???) it may work. There may be some crazy way to do this within a peer if so this is the information I am looking for... The examples and descriptions are not at all clear to me I have 3 accounts with the same provider How do I get incoming calls to come into three different contexts that I will create is the question. From the example file I see: Asterisk can register as a SIP user agent to a SIP proxy (provider) ; Format for the register statement is: ; register = user[:secret[:[EMAIL PROTECTED]:port][/extension] ; ; If no extension is given, the 's' extension is used. The extension needs to ; be defined in extensions.conf to be able to accept calls from this SIP proxy I actually need to do 3 of these. ;register = 2345:[EMAIL PROTECTED]/1234 ; ;Register 2345 at sip provider 'sip_proxy'. Calls from this provider ;connect to local extension 1234 in extensions.conf, default context, ;unless you configure a [sip_proxy] section below, and configure a ;context. Ok I have 3 accounts from the same provider one [sip_proxy] section just puts me in the same problem boat I'm already in using a register line the calls some into the context specified in [general] section of sip.conf I need to somehow differentiate the three SIP 'lines' and give them different contexts to start in. ;Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com] OK sure then how will this associate with my register line that uses provider.com This makes no sense to me... I mean It really makes no sense... Sorry for my confusion. Do I need the register line or do I not need the register line? Why even have a register line if you don't need it and can somehow do this in a peerf, riend or user section. and if you need the register line the instructions say not to use [provider.com] as the peer, then how the heck do you get that register line to work with an associated [peer]. I need to get a handle on how this works before I go posting my sporatic attempts to get a friend,peer or user to 'register' which is not working. The only way I've been able to get my system to take incoming calls from our sip provider so far is to use register lines and keep the system 'registered' with our provider. I don't use any sip providers, so be careful with what I say here. Based on the current sip.conf.sample comments (as of today), it would appear you need to do something like this: register = 2345:[EMAIL PROTECTED]/1234 register = 2346:[EMAIL PROTECTED]/2345 register = 2347:[EMAIL PROTECTED]/3456 The above register statements are used to inform your sip provider which IP address you are coming from, and calls for each of those three accounts (eg, 2345, 2346, and 2347) will be routed to your system. In your extensions.conf, you would need something like: exten = 1234,1,Dial(SIP/3000) exten = 2345,1,Dial(SIP/3001) exten = 3456,1,Dial(SIP/3002) Note the comments in the sample config relative to not using a host= statement in the type=peer section. Also note the above register statements assume the use of three different account names (eg, 2345, 2346, and 2347). As I mentioned above, I don't use any sip providers. But, if I read the sample file correctly, the key to the above working is having three different account names. Olle has made several changes to the sip implementation in asterisk over the last year or so, so there might be variations of how this is done that are asterisk version dependent. He has also posted (several times) comments relative to how incoming sip calls match the various definitions in sip.conf. Again, since I don't use sip providers, I'll go from memory to try and repeat at least a portion of his posts. Be careful as I don't have any recent practical experience on this. It goes something like this: If you specify a host= statement in sip.conf, incoming calls will match the first section in sip.conf that includes that statement (essentially disregarding username and secret, etc). If you don't specify a host= statement in sip.conf and you have a
Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts
Hi,
I'm not an expert, but as far as i know, your incoming calls will
arrive with DID in ${EXTEN}
so the only thing you need is:
exten = 1234,1,GoTo(context1,1234,1) ; example for context extension
and priority
exten = 2345,1,GoTo(context2,2345,1)
exten = 3456,1,GoTo(context3,3456,1)
Be sure that you have created context1 context2 and context3 in your
extensions.conf
And in this context1 context2 and context3 you must have handler for
1234; 2345; and 3456;
example:
[context1]
exten = 1234,1,Answer()
exten = 1234,2,Playback(vm-goodbye)
exten = 1234,3,Hangup()
I didn't test this code, but this is my tip the main idea is that you
need to catch de DID and make a GoTo for the context you want.
Best regards,
Marco Mouta
On 4/2/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Gladden wrote:
What version of asterisk? (been lots of changes happening to the sip
code over the last year)
SVN-branch-1.2-r9156
Have you looked at the sample configs in /usr/src/asterisk/configs?
Yes I have and my own configs are pretty much copies of them.
They do not detail, do or explain the simple concept that I am
trying to accomplish.
If they do I don't see it.
#1 I have more than one incoming SIP account
#2 I would like to have them come into the context of
my choice when a call comes in.
HOW do I do this?
currently I have 3 register lines
there is no way to specify in a register line
some way of making the call start in any other context
other than what is specified in the [general] section
of sip.conf
It seems that somehow maybe if there is a peer tat is somehow
matched to the register line (how???) it may work.
There may be some crazy way to do this within a peer
if so this is the information I am looking for...
The examples and descriptions are not at all clear to me
I have 3 accounts with the same provider
How do I get incoming calls to come into three different contexts
that I will create is the question.
From the example file I see:
Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register = user[:secret[:[EMAIL PROTECTED]:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension needs
to
; be defined in extensions.conf to be able to accept calls from this SIP
proxy
I actually need to do 3 of these.
;register = 2345:[EMAIL PROTECTED]/1234
;
;Register 2345 at sip provider 'sip_proxy'. Calls from this provider
;connect to local extension 1234 in extensions.conf, default context,
;unless you configure a [sip_proxy] section below, and configure a
;context.
Ok I have 3 accounts from the same provider
one [sip_proxy] section just puts me in the same problem boat I'm already
in using a register line
the calls some into the context specified in [general] section of sip.conf
I need to somehow differentiate the three SIP 'lines' and give
them different contexts to start in.
;Tip 1: Avoid assigning hostname to a sip.conf section like
[provider.com]
OK sure then how will this associate with my register line that
uses provider.com
This makes no sense to me...
I mean It really makes no sense...
Sorry for my confusion.
Do I need the register line or do I not need the register line?
Why even have a register line if you don't need it and can somehow
do this in a peerf, riend or user section.
and if you need the register line the instructions say
not to use [provider.com] as the peer, then how the heck do you
get that register line to work with an associated [peer].
I need to get a handle on how this works before I go posting my
sporatic attempts to get a friend,peer or user to 'register'
which is not working.
The only way I've been able to get my system to take incoming calls
from our sip provider so far is to use register lines and keep
the system 'registered' with our provider.
I don't use any sip providers, so be careful with what I say here.
Based on the current sip.conf.sample comments (as of today), it would
appear you need to do something like this:
register = 2345:[EMAIL PROTECTED]/1234
register = 2346:[EMAIL PROTECTED]/2345
register = 2347:[EMAIL PROTECTED]/3456
The above register statements are used to inform your sip provider which
IP address you are coming from, and calls for each of those three
accounts (eg, 2345, 2346, and 2347) will be routed to your system. In
your extensions.conf, you would need something like:
exten = 1234,1,Dial(SIP/3000)
exten = 2345,1,Dial(SIP/3001)
exten = 3456,1,Dial(SIP/3002)
Note the comments in the sample config relative to not using a host=
statement in the type=peer section. Also note the above register
statements assume the use of three different account names (eg, 2345,
2346, and
Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts
What I do is the following and keep in mind I only use one register
statement with my provider:
exten = 18665551234,1,SetVar(FROM_DID=18665551234) ;
exten = 18665551234,2,Goto(from-pstn,s,1) ;
exten = 5185551234,1,SetVar(FROM_DID=5185551234) ;
exten = 5185551234,2,Goto(custom-callid,s,1) ;
On 4/2/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
I'm not an expert, but as far as i know, your incoming calls will
arrive with DID in ${EXTEN}
so the only thing you need is:
exten = 1234,1,GoTo(context1,1234,1) ; example for context extension
and priority
exten = 2345,1,GoTo(context2,2345,1)
exten = 3456,1,GoTo(context3,3456,1)
Be sure that you have created context1 context2 and context3 in your
extensions.conf
And in this context1 context2 and context3 you must have handler for
1234; 2345; and 3456;
example:
[context1]
exten = 1234,1,Answer()
exten = 1234,2,Playback(vm-goodbye)
exten = 1234,3,Hangup()
I didn't test this code, but this is my tip the main idea is that you
need to catch de DID and make a GoTo for the context you want.
Best regards,
Marco Mouta
On 4/2/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Gladden wrote:
What version of asterisk? (been lots of changes happening to the sip
code over the last year)
SVN-branch-1.2-r9156
Have you looked at the sample configs in /usr/src/asterisk/configs?
Yes I have and my own configs are pretty much copies of them.
They do not detail, do or explain the simple concept that I am
trying to accomplish.
If they do I don't see it.
#1 I have more than one incoming SIP account
#2 I would like to have them come into the context of
my choice when a call comes in.
HOW do I do this?
currently I have 3 register lines
there is no way to specify in a register line
some way of making the call start in any other context
other than what is specified in the [general] section
of sip.conf
It seems that somehow maybe if there is a peer tat is somehow
matched to the register line (how???) it may work.
There may be some crazy way to do this within a peer
if so this is the information I am looking for...
The examples and descriptions are not at all clear to me
I have 3 accounts with the same provider
How do I get incoming calls to come into three different contexts
that I will create is the question.
From the example file I see:
Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register = user[:secret[:[EMAIL PROTECTED]:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension
needs to
; be defined in extensions.conf to be able to accept calls from this SIP
proxy
I actually need to do 3 of these.
;register = 2345:[EMAIL PROTECTED]/1234
;
;Register 2345 at sip provider 'sip_proxy'. Calls from this provider
;connect to local extension 1234 in extensions.conf, default context,
;unless you configure a [sip_proxy] section below, and configure a
;context.
Ok I have 3 accounts from the same provider
one [sip_proxy] section just puts me in the same problem boat I'm already
in using a register line
the calls some into the context specified in [general] section of sip.conf
I need to somehow differentiate the three SIP 'lines' and give
them different contexts to start in.
;Tip 1: Avoid assigning hostname to a sip.conf section like
[provider.com]
OK sure then how will this associate with my register line that
uses provider.com
This makes no sense to me...
I mean It really makes no sense...
Sorry for my confusion.
Do I need the register line or do I not need the register line?
Why even have a register line if you don't need it and can somehow
do this in a peerf, riend or user section.
and if you need the register line the instructions say
not to use [provider.com] as the peer, then how the heck do you
get that register line to work with an associated [peer].
I need to get a handle on how this works before I go posting my
sporatic attempts to get a friend,peer or user to 'register'
which is not working.
The only way I've been able to get my system to take incoming calls
from our sip provider so far is to use register lines and keep
the system 'registered' with our provider.
I don't use any sip providers, so be careful with what I say here.
Based on the current sip.conf.sample comments (as of today), it would
appear you need to do something like this:
register = 2345:[EMAIL PROTECTED]/1234
register = 2346:[EMAIL PROTECTED]/2345
register = 2347:[EMAIL PROTECTED]/3456
The above register statements are used to inform your sip provider which
IP address you
[Asterisk-Users] Newbie question - sip.conf incoming contexts
Hello! I've been struggling with the documentation for months on this simple subject... I still have not been able to get this concept down... I have 3 sip accounts (PSTN DID's) that come into my asterisk box and give me phone service from my itsp via SIP. I for the life of me have not been able to figure out how to get them to come in to 3 seperate contexts! This must be simple but I am missing the point. All 3 accounts need a register line (I think) in order to work. The register lines work great but I have not been able to figure out how to get the other two lines to come into another seperate inbound context that I have defined other than the one that is specified in the [general] section of sip.conf The /extension number does not do the trick for me I wuld like for these incoming lines (from the same itsp) to truly land in one of 3 seperate starting contexts in my dialplan based on what phone number (account) they are. Thank you very much for your help... this must be simple but I have not really figured it out in several months of playing around and reading I've figured out a TON of other complex things, but this simple incoming context thing has me a bit stumped. I've tried a few things in my sip peer like register=yes which was suggested on a web site but it does not work. I also tried maing the peer name match the account (phone number) of the sip account and that did not do it either. The peers work fine as outgoing but I've not figured out how to make them work for incoming as my sip itsp requires that I 'register' for inbound calls. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts
I've been struggling with the documentation for months on this simple subject... I still have not been able to get this concept down... I have 3 sip accounts (PSTN DID's) that come into my asterisk box and give me phone service from my itsp via SIP. I for the life of me have not been able to figure out how to get them to come in to 3 seperate contexts! This must be simple but I am missing the point. All 3 accounts need a register line (I think) in order to work. The register lines work great but I have not been able to figure out how to get the other two lines to come into another seperate inbound context that I have defined other than the one that is specified in the [general] section of sip.conf The /extension number does not do the trick for me I wuld like for these incoming lines (from the same itsp) to truly land in one of 3 seperate starting contexts in my dialplan based on what phone number (account) they are. Thank you very much for your help... this must be simple but I have not really figured it out in several months of playing around and reading I've figured out a TON of other complex things, but this simple incoming context thing has me a bit stumped. I've tried a few things in my sip peer like register=yes which was suggested on a web site but it does not work. I also tried maing the peer name match the account (phone number) of the sip account and that did not do it either. The peers work fine as outgoing but I've not figured out how to make them work for incoming as my sip itsp requires that I 'register' for inbound calls. What version of asterisk? (been lots of changes happening to the sip code over the last year) Have you looked at the sample configs in /usr/src/asterisk/configs? It would be far more helpful if you'd post your register statements and each of the sip contexts from sip.conf. Might also include the section of your dialplan that each of the sip.conf contexts refer to. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts
What version of asterisk? (been lots of changes happening to the sip code over the last year) SVN-branch-1.2-r9156 I think what I am trying to do is pretty basic and should not have changed much in the past year. I got started in July of 2005 and I upgrade about once per month. In all this time I have not gotten this simple concept down that I am asking about. Have you looked at the sample configs in /usr/src/asterisk/configs? Yes I have and my own configs are pretty much copies of them. They do not detail, do or explain the simple concept that I am trying to accomplish. If they do I don't see it. #1 I have more than one incoming SIP account #2 I would like to have them come into the context of my choice when a call comes in. HOW do I do this? currently I have 3 register lines there is no way to specify in a register line some way of making the call start in any other context other than what is specified in the [general] section of sip.conf It seems that somehow maybe if there is a peer tat is somehow matched to the register line (how???) it may work. There may be some crazy way to do this within a peer if so this is the information I am looking for... The examples and descriptions are not at all clear to me I have 3 accounts with the same provider How do I get incoming calls to come into three different contexts that I will create is the question. From the example file I see: Asterisk can register as a SIP user agent to a SIP proxy (provider) ; Format for the register statement is: ; register = user[:secret[:[EMAIL PROTECTED]:port][/extension] ; ; If no extension is given, the 's' extension is used. The extension needs to ; be defined in extensions.conf to be able to accept calls from this SIP proxy I actually need to do 3 of these. ; (provider). ; ; host is either a host name defined in DNS or the name of a section defined ; below. ; ; Examples: ; ;register = 1234:[EMAIL PROTECTED] ; ; This will pass incoming calls to the 's' extension ; ; ;register = 2345:[EMAIL PROTECTED]/1234 ; ;Register 2345 at sip provider 'sip_proxy'. Calls from this provider ;connect to local extension 1234 in extensions.conf, default context, ;unless you configure a [sip_proxy] section below, and configure a ;context. Ok I have 3 accounts from the same provider one [sip_proxy] section just puts me in the same problem boat I'm already in using a register line the calls some into the context specified in [general] section of sip.conf I need to somehow differentiate the three SIP 'lines' and give them different contexts to start in. ;Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com] OK sure then how will this associate with my register line that uses provider.com This makes no sense to me... I mean It really makes no sense... Sorry for my confusion. Do I need the register line or do I not need the register line? Why even have a register line if you don't need it and can somehow do this in a peerf, riend or user section. and if you need the register line the instructions say not to use [provider.com] as the peer, then how the heck do you get that register line to work with an associated [peer]. I need to get a handle on how this works before I go posting my sporatic attempts to get a friend,peer or user to 'register' which is not working. The only way I've been able to get my system to take incoming calls from our sip provider so far is to use register lines and keep the system 'registered' with our provider. ;Tip 2: Use separate type=peer and type=user sections for SIP providers ; (instead of type=friend) if you have calls in both directions It would be far more helpful if you'd post your register statements and each of the sip contexts from sip.conf. Might also include the section of your dialplan that each of the sip.conf contexts refer to. I can do this but only once I can try something that seemingly should work. Right now I'm pretty much using default configs, a single incoming context and register lines of which all of those calls come into this single context. I need to know 'what to try' in order to give this a shot! Thanks for your help and suggestions! Steve I've been struggling with the documentation for months on this simple subject... I still have not been able to get this concept down... I have 3 sip accounts (PSTN DID's) that come into my asterisk box and give me phone service from my itsp via SIP. I for the life of me have not been able to figure out how to get them to come in to 3 seperate contexts! This must be simple but I am missing the point. All 3 accounts need a register line (I think) in order to work. The register lines work great but I have not been able to figure out how to get the other two lines to come into another seperate inbound context that I have defined other than the one that
[Asterisk-Users] Newbie question
Hi there,I would like to connect an Aasterisk Server with a Panasonic PBX (has E1extension). I only need 4 Lines. So I thought I could use an Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1 card which is more expensive.I dont now which card to take.Please tell me what you think about. I appreciate all suggestions.Thanks in advanceHousi Mueller Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more on new and used cars.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question
On Wed, 15 Feb 2006 08:59:22 -0800 (PST) housi mueller [EMAIL PROTECTED] wrote: Hi there, I would like to connect an Aasterisk Server with a Panasonic PBX (has E1extension). I only need 4 Lines. So I thought I could use an Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1 card which is more expensive. I dont now which card to take. Please tell me what you think about. I appreciate all suggestions. Thanks in advance Housi Mueller My personal preference would be to go with the E1/T1 now. It would give you expansion opportunities in the future between the Asterisk and the Panasonic, allow you to be all digital between, and finally if you ever decided to ever get rid of the Panasonic, you could pull a T1 from the telco straight into the Asterisk box. Spend a little more now and save in the future. Just my $.02 Robert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question
That is a good argument. But I am not sureyet. Do you know if there are big voice quality differences between the Digital and the Analog card?HousiRobert Webb [EMAIL PROTECTED] wrote: On Wed, 15 Feb 2006 08:59:22 -0800 (PST)housi mueller <[EMAIL PROTECTED]>wrote: Hi there, I would like to connect an Aasterisk Server with a Panasonic PBX (has E1extension). I only need 4 Lines. So I thought I could use an Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1 card which is more expensive. I dont now which card to take. Please tell me what you think about. I appreciate all suggestions. Thanks in advance Housi Mueller My personal preference would be to go with the E1/T1 now. It would give you expansion opportunities in the future between the Asterisk and the Panasonic, allow you to be all digital between, and finally if you ever decided to ever get rid of the Panasonic, you could pull a T1 from the telco straight into the Asterisk box.Spend a little more now and save in the future.Just my $.02Robert Yahoo! Mail Use Photomail to share photos without annoying attachments.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie question about making outbound call
Hi Jason. It seems your doing things right whatever that means. I think the problem is more hardware related. Sure you have line in the FXO?? have you tried dialing directly from some IP Phone?? I have several applications that relay on automatic call generation with Asterisk Manager and a PHP classes i have. But, as i said, i think the problem is related to the configuration of the card. what does ztcfg -vv says? what does zttool says?? best regardsOn 12/25/05, Jason D. Wolfe [EMAIL PROTECTED] wrote: Hello,Somehow I've missed something here, so hopefully I'll be able to provideenough of my setup to get some help.I feel I'm very close to gettingit, but missing something none the less...1. I have a digium TDM400 with (2) FXO modules on channel 3 and 4 hooked to two POTS lines.2. I have the following entry in zapata.conf file:usecallerid=yeshidecallerid=nocallwaiting=nothreewaycalling=yestransfer=yesechocancel=yesechotraining=yescallprogress=no context=incomingsignalling=fxs_kschannel=43. I have the following entry in extensions.conf[callAgent]exten=outbound,1,Dial(Zap/4/phonenumber) ;where phonenumber is a 10digit number exten=outbound,n,Playback(access-code) ; just for the sake of doingsomething!4. I am using Asterisk Java Manager AGI OriginateAction with thefollowing code in a jsp page running on atomcat server: //manageAGIManagerConnection managerConnection;ManagerConnectionFactory factory;OriginateAction originateAction;ManagerResponse originateResponse;factory = new ManagerConnectionFactory(); managerConnection = factory.getManagerConnection(192.168.1.4,jason,nosaj111);// connect to Asterisk and log inmanagerConnection.login ();originateAction = new OriginateAction();originateAction.setAsync(true);originateAction.setChannel(Zap/4);originateAction.setContext(callAgent); originateAction.setExten(outbound);originateAction.setPriority(new Integer(1));originateAction.setTimeout(3000);originateResponse =managerConnection.sendAction (originateAction, 3);6. when I execute the jsp page, I watch the console started with/usr/sbin/asterisk -cvvand I get the following message (I substituted phonenumber in for the 10digit number again) *CLI == Parsing '/etc/asterisk/manager.conf': Found== Manager 'jason' logged on from 192.168.1.3 Channel Zap/4-1 was answered.-- Executing Dial(Zap/4-1, Zap/4/phonenumber) in new stack Dec 25 10:55:40 NOTICE[3989]: app_dial.c:1010 dial_exec_full: Unable tocreate channel of type 'Zap' (cause 0 - Unknown)== Everyone is busy/congested at this time (1:0/0/1)-- Executing Playback(Zap/4-1, access-code) in new stack -- Playing 'access-code' (language 'en')== Manager 'jason' logged off from 192.168.1.3== Auto fallthrough, channel 'Zap/4-1' status is 'CHANUNAVAIL'-- Hungup 'Zap/4-1' exten = outbound,1,Hangup()What I eventually want to accomplish is the following:I want a web user (using a JSP page I think) to be able to click abutton and cause asterisk to dial outbound on both FXO ports, wait for an answer, play some files, accept some input, and bridge the two callstogether.am I on the wrong track?is there anything that is standing out that Iam just not understanding here?ANY comments will be much appreciated. Thank you,Jason___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie question about making outbound call
Hello, Somehow I've missed something here, so hopefully I'll be able to provide enough of my setup to get some help. I feel I'm very close to getting it, but missing something none the less... 1. I have a digium TDM400 with (2) FXO modules on channel 3 and 4 hooked to two POTS lines. 2. I have the following entry in zapata.conf file: usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes callprogress=no context=incoming signalling=fxs_ks channel=4 3. I have the following entry in extensions.conf [callAgent] exten=outbound,1,Dial(Zap/4/phonenumber) ;where phonenumber is a 10 digit number exten=outbound,n,Playback(access-code) ; just for the sake of doing something! 4. I am using Asterisk Java Manager AGI OriginateAction with the following code in a jsp page running on a tomcat server: //manageAGI ManagerConnection managerConnection; ManagerConnectionFactory factory; OriginateAction originateAction; ManagerResponse originateResponse; factory = new ManagerConnectionFactory(); managerConnection = factory.getManagerConnection(192.168.1.4,jason, nosaj111); // connect to Asterisk and log in managerConnection.login(); originateAction = new OriginateAction(); originateAction.setAsync(true); originateAction.setChannel(Zap/4); originateAction.setContext(callAgent); originateAction.setExten(outbound); originateAction.setPriority(new Integer(1)); originateAction.setTimeout(3000); originateResponse = managerConnection.sendAction(originateAction, 3); 6. when I execute the jsp page, I watch the console started with /usr/sbin/asterisk -cvv and I get the following message (I substituted phonenumber in for the 10 digit number again) *CLI == Parsing '/etc/asterisk/manager.conf': Found == Manager 'jason' logged on from 192.168.1.3 Channel Zap/4-1 was answered. -- Executing Dial(Zap/4-1, Zap/4/phonenumber) in new stack Dec 25 10:55:40 NOTICE[3989]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Playback(Zap/4-1, access-code) in new stack -- Playing 'access-code' (language 'en') == Manager 'jason' logged off from 192.168.1.3 == Auto fallthrough, channel 'Zap/4-1' status is 'CHANUNAVAIL' -- Hungup 'Zap/4-1' exten = outbound,1,Hangup() What I eventually want to accomplish is the following: I want a web user (using a JSP page I think) to be able to click a button and cause asterisk to dial outbound on both FXO ports, wait for an answer, play some files, accept some input, and bridge the two calls together. am I on the wrong track? is there anything that is standing out that I am just not understanding here? ANY comments will be much appreciated. Thank you, Jason ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question
I dont need to configure zaptel device, you dont use it :) 2005/11/30, [EMAIL PROTECTED] [EMAIL PROTECTED]: Hello friends, I am using asterisk 1.2 with ooh323. I am using sip and h323 phones. My question is I am using a Welltech FXO box and ip phones by Welltech. Do I still need to configure zapata.conf and zaptel.conf which I read in the documentation from asterisk pdf file downoladed from asterisk.org ? I think I dont because I dont use a digium card but do I have to still confugure for FXO and FXS ports? Kindly help me solving my doubt. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Truth springs from argument amongst friends. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question on 1.2 extension configs
try [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/ 2005/11/29, bram kortleven [EMAIL PROTECTED]: Are there any example configs? Or does anybody have a default config for this setup: 1 analog digium clone card for an analogue line (my home line) Several sip phones (a few of them on the outside of my lan (NAT fw between) and 2 insde my lan) Or a simple way of configging through a frontend/script/management utility... I installed astlinux But it does not allow to install and use AMP... Anyone having another script? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question on 1.2 extension configs
On 00:24, Tue 29 Nov 05, bram kortleven wrote: Are there any example configs? Or does anybody have a default config for this setup: 1 analog digium clone card for an analogue line (my home line) Several sip phones (a few of them on the outside of my lan (NAT fw between) and 2 insde my lan) Or a simple way of configging through a frontend/script/management utility... I installed astlinux But it does not allow to install and use AMP... Anyone having another script? Get the source of asterisk and type: make samples That will create a set of default config files. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie question on 1.2 extension configs
Are there any example configs? Or does anybody have a default config for this setup: 1 analog digium clone card for an analogue line (my home line) Several sip phones (a few of them on the outside of my lan (NAT fw between) and 2 insde my lan) Or a simple way of configging through a frontend/script/management utility... I installed astlinux But it does not allow to install and use AMP... Anyone having another script? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie question
Hello friends, I am using asterisk 1.2 with ooh323. I am using sip and h323 phones. My question is I am using a Welltech FXO box and ip phones by Welltech. Do I still need to configure zapata.conf and zaptel.conf which I read in the documentation from asterisk pdf file downoladed from asterisk.org ? I think I dont because I dont use a digium card but do I have to still confugure for FXO and FXS ports? Kindly help me solving my doubt. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Truth springs from argument amongst friends. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie question on 1.2 extension configs
Are there any example configs? Or does anybody have a default config for this setup: 1 analog digium clone card for an analogue line (my home line) Several sip phones (a few of them on the outside of my lan (NAT fw between) and 2 insde my lan) Or a simple way of configging through a frontend/script/management utility... I installed astlinux But it does not allow to install and use AMP... Anyone having another script? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie question. (Long)
Hi all :
My first posting to the group - please be gentle!
I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu),
downloaded the binary package.
Now I'm trying to put the working installation on my production server
along with HTTP etc.
( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com
2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386
GNU/Linux).
That box, until yesterday, was running Fedora core 3. I tried the
tarball download of 1.2.0.rc2, ran make OK, then make install, make samples.
When I tried to run Asterisk, I got (immediately) Illegal Instruction.
Tried on my FC4 laptop, worked just fine.
Concluded I needed FC4, so upgraded the server yesterday. Six hours later...
Reran make clean, make...
Same problem.
Then tried 1.2.0; same problem.
Then tried 1.0.9; same problem.
Finally removed everything to do with asterisk, pulled dowm 1.2.0 tar
ball again, and re-installed.
Same old problem, illegal instruction.
I did an strace, which follows. I don't know enough to decide what the
strace is telling me. (The missing /etc/ld.so.preload is also missing on
the FC4 laptop which works, so I concluded that that was not the problem.)
Any help much appreciated.
Regards
Roger
[EMAIL PROTECTED] sbin]$ sudo strace ./asterisk
execve(./asterisk, [./asterisk], [/* 27 vars */]) = 0
uname({sys=Linux, node=coach.hillconsult.com, ...}) = 0
brk(0) = 0x8773000
access(/etc/ld.so.preload, R_OK) = -1 ENOENT (No such file or
directory)
open(/etc/ld.so.cache, O_RDONLY) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=48722, ...}) = 0
old_mmap(NULL, 48722, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb7f85000
close(3)= 0
open(/lib/libdl.so.2, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\250\213...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=16760, ...}) = 0
old_mmap(0x8f8000, 12388, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8f8000
old_mmap(0x8fa000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x1000) = 0x8fa000
close(3)= 0
open(/lib/tls/i686/libpthread.so.0, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\334\306...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=103404, ...}) = 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS,
-1, 0) = 0xb7f84000
old_mmap(0x938000, 65980, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x938000
old_mmap(0x945000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xc000) = 0x945000
old_mmap(0x947000, 4540, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x947000
close(3)= 0
open(/usr/lib/libncurses.so.5, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\200\343...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=985952, ...}) = 0
old_mmap(0x5a0, 290348, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x5a0
old_mmap(0x5a3e000, 36864, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x3d000) = 0x5a3e000
close(3)= 0
open(/lib/tls/i686/libm.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\320\22...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=213872, ...}) = 0
old_mmap(0x8fe000, 139424, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8fe000
old_mmap(0x91f000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2) = 0x91f000
close(3)= 0
open(/lib/libresolv.so.2, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\350\323...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=72956, ...}) = 0
old_mmap(0x94b000, 71848, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x94b000
old_mmap(0x959000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xd000) = 0x959000
old_mmap(0x95b000, 6312, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x95b000
close(3)= 0
open(/lib/libssl.so.5, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\360\26...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=230056, ...}) = 0
old_mmap(0xaa8000, 228948, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0xaa8000
old_mmap(0xadd000, 12288, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x35000) = 0xadd000
close(3)= 0
open(/lib/tls/i686/libc.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\230n\177...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=1431008, ...}) = 0
old_mmap(0x7e2000, 1129660, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x7e2000
mprotect(0x8ef000, 27836,
Re: [Asterisk-Users] Newbie question. (Long)
Hi Roger,
Following this instructions:
http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3
I was able to install and run Asterisk several times without problems.
See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora
Regards,
Vassil Kolarov
www.ittconsult.com
Roger Hill wrote:
Hi all :
My first posting to the group - please be gentle!
I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu),
downloaded the binary package.
Now I'm trying to put the working installation on my production server
along with HTTP etc.
( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com
2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386
GNU/Linux).
That box, until yesterday, was running Fedora core 3. I tried the
tarball download of 1.2.0.rc2, ran make OK, then make install, make
samples.
When I tried to run Asterisk, I got (immediately) Illegal Instruction.
Tried on my FC4 laptop, worked just fine.
Concluded I needed FC4, so upgraded the server yesterday. Six hours
later...
Reran make clean, make...
Same problem.
Then tried 1.2.0; same problem.
Then tried 1.0.9; same problem.
Finally removed everything to do with asterisk, pulled dowm 1.2.0 tar
ball again, and re-installed.
Same old problem, illegal instruction.
I did an strace, which follows. I don't know enough to decide what the
strace is telling me. (The missing /etc/ld.so.preload is also missing
on the FC4 laptop which works, so I concluded that that was not the
problem.)
Any help much appreciated.
Regards
Roger
[EMAIL PROTECTED] sbin]$ sudo strace ./asterisk
execve(./asterisk, [./asterisk], [/* 27 vars */]) = 0
uname({sys=Linux, node=coach.hillconsult.com, ...}) = 0
brk(0) = 0x8773000
access(/etc/ld.so.preload, R_OK) = -1 ENOENT (No such file or
directory)
open(/etc/ld.so.cache, O_RDONLY) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=48722, ...}) = 0
old_mmap(NULL, 48722, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb7f85000
close(3)= 0
open(/lib/libdl.so.2, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\250\213...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=16760, ...}) = 0
old_mmap(0x8f8000, 12388, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8f8000
old_mmap(0x8fa000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x1000) = 0x8fa000
close(3)= 0
open(/lib/tls/i686/libpthread.so.0, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\334\306...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=103404, ...}) = 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS,
-1, 0) = 0xb7f84000
old_mmap(0x938000, 65980, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x938000
old_mmap(0x945000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xc000) = 0x945000
old_mmap(0x947000, 4540, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x947000
close(3)= 0
open(/usr/lib/libncurses.so.5, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\200\343...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=985952, ...}) = 0
old_mmap(0x5a0, 290348, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x5a0
old_mmap(0x5a3e000, 36864, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x3d000) = 0x5a3e000
close(3)= 0
open(/lib/tls/i686/libm.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\320\22...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=213872, ...}) = 0
old_mmap(0x8fe000, 139424, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8fe000
old_mmap(0x91f000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2) = 0x91f000
close(3)= 0
open(/lib/libresolv.so.2, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\350\323...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=72956, ...}) = 0
old_mmap(0x94b000, 71848, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x94b000
old_mmap(0x959000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xd000) = 0x959000
old_mmap(0x95b000, 6312, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x95b000
close(3)= 0
open(/lib/libssl.so.5, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\360\26...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=230056, ...}) = 0
old_mmap(0xaa8000, 228948, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0xaa8000
old_mmap(0xadd000, 12288, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x35000) = 0xadd000
close(3)
Re: [Asterisk-Users] Newbie question. (Long)
Thanks Vassil - I'll try those pointers and report back.
Roger
Vassil Kolarov wrote:
Hi Roger,
Following this instructions:
http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3
I was able to install and run Asterisk several times without problems.
See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora
Regards,
Vassil Kolarov
www.ittconsult.com
Roger Hill wrote:
Hi all :
My first posting to the group - please be gentle!
I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu),
downloaded the binary package.
Now I'm trying to put the working installation on my production
server along with HTTP etc.
( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com
2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386
GNU/Linux).
That box, until yesterday, was running Fedora core 3. I tried the
tarball download of 1.2.0.rc2, ran make OK, then make install, make
samples.
When I tried to run Asterisk, I got (immediately) Illegal Instruction.
Tried on my FC4 laptop, worked just fine.
Concluded I needed FC4, so upgraded the server yesterday. Six hours
later...
Reran make clean, make...
Same problem.
Then tried 1.2.0; same problem.
Then tried 1.0.9; same problem.
Finally removed everything to do with asterisk, pulled dowm 1.2.0 tar
ball again, and re-installed.
Same old problem, illegal instruction.
I did an strace, which follows. I don't know enough to decide what
the strace is telling me. (The missing /etc/ld.so.preload is also
missing on the FC4 laptop which works, so I concluded that that was
not the problem.)
Any help much appreciated.
Regards
Roger
[EMAIL PROTECTED] sbin]$ sudo strace ./asterisk
execve(./asterisk, [./asterisk], [/* 27 vars */]) = 0
uname({sys=Linux, node=coach.hillconsult.com, ...}) = 0
brk(0) = 0x8773000
access(/etc/ld.so.preload, R_OK) = -1 ENOENT (No such file or
directory)
open(/etc/ld.so.cache, O_RDONLY) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=48722, ...}) = 0
old_mmap(NULL, 48722, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb7f85000
close(3)= 0
open(/lib/libdl.so.2, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\250\213...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=16760, ...}) = 0
old_mmap(0x8f8000, 12388, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8f8000
old_mmap(0x8fa000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x1000) = 0x8fa000
close(3)= 0
open(/lib/tls/i686/libpthread.so.0, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\334\306...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=103404, ...}) = 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS,
-1, 0) = 0xb7f84000
old_mmap(0x938000, 65980, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x938000
old_mmap(0x945000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xc000) = 0x945000
old_mmap(0x947000, 4540, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x947000
close(3)= 0
open(/usr/lib/libncurses.so.5, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\200\343...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=985952, ...}) = 0
old_mmap(0x5a0, 290348, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x5a0
old_mmap(0x5a3e000, 36864, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x3d000) = 0x5a3e000
close(3)= 0
open(/lib/tls/i686/libm.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\320\22...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=213872, ...}) = 0
old_mmap(0x8fe000, 139424, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8fe000
old_mmap(0x91f000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2) = 0x91f000
close(3)= 0
open(/lib/libresolv.so.2, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\350\323...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=72956, ...}) = 0
old_mmap(0x94b000, 71848, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x94b000
old_mmap(0x959000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xd000) = 0x959000
old_mmap(0x95b000, 6312, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x95b000
close(3)= 0
open(/lib/libssl.so.5, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\360\26...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=230056, ...}) = 0
old_mmap(0xaa8000, 228948, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0xaa8000
old_mmap(0xadd000, 12288, PROT_READ|PROT_WRITE,
Re: [Asterisk-Users] Newbie question. (Long)
Hi All:
I've been through the compile/install procedure pointed out by Vassil: I
still crash on startup. Can anyone else give me some pointers, please?
Roger
Roger Hill wrote:
Thanks Vassil - I'll try those pointers and report back.
Roger
Vassil Kolarov wrote:
Hi Roger,
Following this instructions:
http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3
I was able to install and run Asterisk several times without problems.
See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora
Regards,
Vassil Kolarov
www.ittconsult.com
Roger Hill wrote:
Hi all :
My first posting to the group - please be gentle!
I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu),
downloaded the binary package.
Now I'm trying to put the working installation on my production
server along with HTTP etc.
( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com
2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386
GNU/Linux).
That box, until yesterday, was running Fedora core 3. I tried the
tarball download of 1.2.0.rc2, ran make OK, then make install, make
samples.
When I tried to run Asterisk, I got (immediately) Illegal
Instruction.
Tried on my FC4 laptop, worked just fine.
Concluded I needed FC4, so upgraded the server yesterday. Six hours
later...
Reran make clean, make...
Same problem.
Then tried 1.2.0; same problem.
Then tried 1.0.9; same problem.
Finally removed everything to do with asterisk, pulled dowm 1.2.0
tar ball again, and re-installed.
Same old problem, illegal instruction.
I did an strace, which follows. I don't know enough to decide what
the strace is telling me. (The missing /etc/ld.so.preload is also
missing on the FC4 laptop which works, so I concluded that that was
not the problem.)
Any help much appreciated.
Regards
Roger
[EMAIL PROTECTED] sbin]$ sudo strace ./asterisk
execve(./asterisk, [./asterisk], [/* 27 vars */]) = 0
uname({sys=Linux, node=coach.hillconsult.com, ...}) = 0
brk(0) = 0x8773000
access(/etc/ld.so.preload, R_OK) = -1 ENOENT (No such file or
directory)
open(/etc/ld.so.cache, O_RDONLY) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=48722, ...}) = 0
old_mmap(NULL, 48722, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb7f85000
close(3)= 0
open(/lib/libdl.so.2, O_RDONLY) = 3
read(3,
\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\250\213..., 512) =
512
fstat64(3, {st_mode=S_IFREG|0755, st_size=16760, ...}) = 0
old_mmap(0x8f8000, 12388, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8f8000
old_mmap(0x8fa000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x1000) = 0x8fa000
close(3)= 0
open(/lib/tls/i686/libpthread.so.0, O_RDONLY) = 3
read(3,
\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\334\306..., 512) =
512
fstat64(3, {st_mode=S_IFREG|0755, st_size=103404, ...}) = 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0xb7f84000
old_mmap(0x938000, 65980, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x938000
old_mmap(0x945000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xc000) = 0x945000
old_mmap(0x947000, 4540, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x947000
close(3)= 0
open(/usr/lib/libncurses.so.5, O_RDONLY) = 3
read(3,
\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\200\343..., 512) =
512
fstat64(3, {st_mode=S_IFREG|0755, st_size=985952, ...}) = 0
old_mmap(0x5a0, 290348, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x5a0
old_mmap(0x5a3e000, 36864, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x3d000) = 0x5a3e000
close(3)= 0
open(/lib/tls/i686/libm.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\320\22...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=213872, ...}) = 0
old_mmap(0x8fe000, 139424, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8fe000
old_mmap(0x91f000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2) = 0x91f000
close(3)= 0
open(/lib/libresolv.so.2, O_RDONLY) = 3
read(3,
\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\350\323..., 512) =
512
fstat64(3, {st_mode=S_IFREG|0755, st_size=72956, ...}) = 0
old_mmap(0x94b000, 71848, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x94b000
old_mmap(0x959000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xd000) = 0x959000
old_mmap(0x95b000, 6312, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x95b000
close(3)= 0
open(/lib/libssl.so.5, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\360\26...,
512) = 512
fstat64(3,
Re: [Asterisk-Users] Newbie question. (Long)
Asterisk runs just fine on fc3. Best guess on your problem is that you've got come default config parameters in /etc/asterisk directory that it is not liking at all. You might try starting asterisk with 'asterisk -cvd' and watch the output for errors. Hi All: I've been through the compile/install procedure pointed out by Vassil: I still crash on startup. Can anyone else give me some pointers, please? Roger Roger Hill wrote: Thanks Vassil - I'll try those pointers and report back. Roger Vassil Kolarov wrote: Hi Roger, Following this instructions: http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3 I was able to install and run Asterisk several times without problems. See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora Regards, Vassil Kolarov www.ittconsult.com Roger Hill wrote: Hi all : My first posting to the group - please be gentle! I've been messing with Asterisk for a couple of weeks now. 1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), downloaded the binary package. Now I'm trying to put the working installation on my production server along with HTTP etc. ( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 GNU/Linux). That box, until yesterday, was running Fedora core 3. I tried the tarball download of 1.2.0.rc2, ran make OK, then make install, make samples. When I tried to run Asterisk, I got (immediately) Illegal Instruction. Tried on my FC4 laptop, worked just fine. ___ --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] Newbie question. (Long)
Rich: Thanks. I tried that, with and without any config files in /etc/asterisk. It still falls over instantly, no messages other than 'Illegal Instruction'. Asterisk is running on other machines for me quite happily, but just does not want to play nice on this box. I'm sure I'm doing something silly, but for the life of me cannot see what it is. It does not get as far as writing anything to any log files in /var/log/asterisk. Roger Rich Adamson wrote: Asterisk runs just fine on fc3. Best guess on your problem is that you've got come default config parameters in /etc/asterisk directory that it is not liking at all. You might try starting asterisk with 'asterisk -cvd' and watch the output for errors. Hi All: I've been through the compile/install procedure pointed out by Vassil: I still crash on startup. Can anyone else give me some pointers, please? Roger Roger Hill wrote: Thanks Vassil - I'll try those pointers and report back. Roger Vassil Kolarov wrote: Hi Roger, Following this instructions: http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3 I was able to install and run Asterisk several times without problems. See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora Regards, Vassil Kolarov www.ittconsult.com Roger Hill wrote: Hi all : My first posting to the group - please be gentle! I've been messing with Asterisk for a couple of weeks now. 1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), downloaded the binary package. Now I'm trying to put the working installation on my production server along with HTTP etc. ( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 GNU/Linux). That box, until yesterday, was running Fedora core 3. I tried the tarball download of 1.2.0.rc2, ran make OK, then make install, make samples. When I tried to run Asterisk, I got (immediately) Illegal Instruction. Tried on my FC4 laptop, worked just fine. ___ --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 -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --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] Newbie question. (Long)
Well... the next best guess is the binary package that you downloaded has some dependencies that are not on your system, or, the package simply wasn't intended for your distro (for one reason or another). Does the system have a developement environment that would allow you down download the cvs source and compile it? Rich: Thanks. I tried that, with and without any config files in /etc/asterisk. It still falls over instantly, no messages other than 'Illegal Instruction'. Asterisk is running on other machines for me quite happily, but just does not want to play nice on this box. I'm sure I'm doing something silly, but for the life of me cannot see what it is. It does not get as far as writing anything to any log files in /var/log/asterisk. Roger Rich Adamson wrote: Asterisk runs just fine on fc3. Best guess on your problem is that you've got come default config parameters in /etc/asterisk directory that it is not liking at all. You might try starting asterisk with 'asterisk -cvd' and watch the output for errors. Hi All: I've been through the compile/install procedure pointed out by Vassil: I still crash on startup. Can anyone else give me some pointers, please? Roger Roger Hill wrote: Thanks Vassil - I'll try those pointers and report back. Roger Vassil Kolarov wrote: Hi Roger, Following this instructions: http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3 I was able to install and run Asterisk several times without problems. See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora Regards, Vassil Kolarov www.ittconsult.com Roger Hill wrote: Hi all : My first posting to the group - please be gentle! I've been messing with Asterisk for a couple of weeks now. 1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), downloaded the binary package. Now I'm trying to put the working installation on my production server along with HTTP etc. ( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 GNU/Linux). That box, until yesterday, was running Fedora core 3. I tried the tarball download of 1.2.0.rc2, ran make OK, then make install, make samples. When I tried to run Asterisk, I got (immediately) Illegal Instruction. Tried on my FC4 laptop, worked just fine. ___ --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 -- Roger Hill07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --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 ---End of Original Message- ___ --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] Newbie question. (Long)
Rich: Sorry if I did not make myself clear. I was trying to give some history, which is where the downloaded package came from. On this box (FC4), I am currently downloading the 1.2.0 source from asterisk.org (but not the CVS), and trying to compile and build from scratch. The build seems fine - if it will help I can post the output from the makes - but the built executable just crashes. I have done the same thing on another FC4 box (my laptop) without any problems. Doees that help at all? (And many thanks for the help, BTW) Roger Rich Adamson wrote: Well... the next best guess is the binary package that you downloaded has some dependencies that are not on your system, or, the package simply wasn't intended for your distro (for one reason or another). Does the system have a developement environment that would allow you down download the cvs source and compile it? Rich: Thanks. I tried that, with and without any config files in /etc/asterisk. It still falls over instantly, no messages other than 'Illegal Instruction'. Asterisk is running on other machines for me quite happily, but just does not want to play nice on this box. I'm sure I'm doing something silly, but for the life of me cannot see what it is. It does not get as far as writing anything to any log files in /var/log/asterisk. Roger Rich Adamson wrote: Asterisk runs just fine on fc3. Best guess on your problem is that you've got come default config parameters in /etc/asterisk directory that it is not liking at all. You might try starting asterisk with 'asterisk -cvd' and watch the output for errors. Hi All: I've been through the compile/install procedure pointed out by Vassil: I still crash on startup. Can anyone else give me some pointers, please? Roger Roger Hill wrote: Thanks Vassil - I'll try those pointers and report back. Roger Vassil Kolarov wrote: Hi Roger, Following this instructions: http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3 I was able to install and run Asterisk several times without problems. See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora Regards, Vassil Kolarov www.ittconsult.com Roger Hill wrote: Hi all : My first posting to the group - please be gentle! I've been messing with Asterisk for a couple of weeks now. 1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), downloaded the binary package. Now I'm trying to put the working installation on my production server along with HTTP etc. ( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 GNU/Linux). That box, until yesterday, was running Fedora core 3. I tried the tarball download of 1.2.0.rc2, ran make OK, then make install, make samples. When I tried to run Asterisk, I got (immediately) Illegal Instruction. Tried on my FC4 laptop, worked just fine. ___ --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 -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --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 ---End of Original Message- ___ --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 -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --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
Illegal Instruction on new FC4 install [was: Re: [Asterisk-Users] Newbie question. (Long)]
On Fri, Nov 18, 2005 at 09:12:46AM +, Roger Hill wrote:
Hi all :
My first posting to the group - please be gentle!
Please use a more descriptive subject line.
I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu),
downloaded the binary package.
Now I'm trying to put the working installation on my production server
along with HTTP etc.
( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com
2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386
GNU/Linux).
That box, until yesterday, was running Fedora core 3. I tried the
tarball download of 1.2.0.rc2, ran make OK, then make install, make samples.
When I tried to run Asterisk, I got (immediately) Illegal Instruction.
Tried on my FC4 laptop, worked just fine.
Concluded I needed FC4, so upgraded the server yesterday. Six hours later...
Reran make clean, make...
Same problem.
Then tried 1.2.0; same problem.
Then tried 1.0.9; same problem.
Finally removed everything to do with asterisk, pulled dowm 1.2.0 tar
ball again, and re-installed.
Same old problem, illegal instruction.
What CPU is it? cat /proc/cpuinfo
I did an strace, which follows. I don't know enough to decide what the
strace is telling me. (The missing /etc/ld.so.preload is also missing on
the FC4 laptop which works, so I concluded that that was not the problem.)
Indeed it is not a problem.
Any help much appreciated.
Regards
Roger
[EMAIL PROTECTED] sbin]$ sudo strace ./asterisk
When you try to strace a process that may fork, try 'strace -f' instead.
Also: what does a simple 'asterisk -cddvv' give?
execve(./asterisk, [./asterisk], [/* 27 vars */]) = 0
uname({sys=Linux, node=coach.hillconsult.com, ...}) = 0
brk(0) = 0x8773000
access(/etc/ld.so.preload, R_OK) = -1 ENOENT (No such file or
directory)
open(/etc/ld.so.cache, O_RDONLY) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=48722, ...}) = 0
old_mmap(NULL, 48722, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb7f85000
close(3)= 0
open(/lib/libdl.so.2, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\250\213...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=16760, ...}) = 0
old_mmap(0x8f8000, 12388, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8f8000
old_mmap(0x8fa000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x1000) = 0x8fa000
close(3)= 0
open(/lib/tls/i686/libpthread.so.0, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\334\306...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=103404, ...}) = 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS,
-1, 0) = 0xb7f84000
old_mmap(0x938000, 65980, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x938000
old_mmap(0x945000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xc000) = 0x945000
old_mmap(0x947000, 4540, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x947000
close(3)= 0
open(/usr/lib/libncurses.so.5, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\200\343...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=985952, ...}) = 0
old_mmap(0x5a0, 290348, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x5a0
old_mmap(0x5a3e000, 36864, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x3d000) = 0x5a3e000
close(3)= 0
open(/lib/tls/i686/libm.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\320\22...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=213872, ...}) = 0
old_mmap(0x8fe000, 139424, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8fe000
old_mmap(0x91f000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2) = 0x91f000
close(3)= 0
open(/lib/libresolv.so.2, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\350\323...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=72956, ...}) = 0
old_mmap(0x94b000, 71848, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x94b000
old_mmap(0x959000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xd000) = 0x959000
old_mmap(0x95b000, 6312, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x95b000
close(3)= 0
open(/lib/libssl.so.5, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\360\26...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=230056, ...}) = 0
old_mmap(0xaa8000, 228948, PROT_READ|PROT_EXEC,
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0xaa8000
old_mmap(0xadd000, 12288, PROT_READ|PROT_WRITE,
Re: [Asterisk-Users] Newbie question. (Long)
Roger, Can you try with a fresh Fedora installation on this box? Vassil Roger Hill wrote: Rich: Sorry if I did not make myself clear. I was trying to give some history, which is where the downloaded package came from. On this box (FC4), I am currently downloading the 1.2.0 source from asterisk.org (but not the CVS), and trying to compile and build from scratch. The build seems fine - if it will help I can post the output from the makes - but the built executable just crashes. I have done the same thing on another FC4 box (my laptop) without any problems. Doees that help at all? (And many thanks for the help, BTW) Roger Rich Adamson wrote: Well... the next best guess is the binary package that you downloaded has some dependencies that are not on your system, or, the package simply wasn't intended for your distro (for one reason or another). Does the system have a developement environment that would allow you down download the cvs source and compile it? Rich: Thanks. I tried that, with and without any config files in /etc/asterisk. It still falls over instantly, no messages other than 'Illegal Instruction'. Asterisk is running on other machines for me quite happily, but just does not want to play nice on this box. I'm sure I'm doing something silly, but for the life of me cannot see what it is. It does not get as far as writing anything to any log files in /var/log/asterisk. Roger Rich Adamson wrote: Asterisk runs just fine on fc3. Best guess on your problem is that you've got come default config parameters in /etc/asterisk directory that it is not liking at all. You might try starting asterisk with 'asterisk -cvd' and watch the output for errors. Hi All: I've been through the compile/install procedure pointed out by Vassil: I still crash on startup. Can anyone else give me some pointers, please? Roger Roger Hill wrote: Thanks Vassil - I'll try those pointers and report back. Roger Vassil Kolarov wrote: Hi Roger, Following this instructions: http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3 I was able to install and run Asterisk several times without problems. See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora Regards, Vassil Kolarov www.ittconsult.com Roger Hill wrote: Hi all : My first posting to the group - please be gentle! I've been messing with Asterisk for a couple of weeks now. 1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), downloaded the binary package. Now I'm trying to put the working installation on my production server along with HTTP etc. ( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 GNU/Linux). That box, until yesterday, was running Fedora core 3. I tried the tarball download of 1.2.0.rc2, ran make OK, then make install, make samples. When I tried to run Asterisk, I got (immediately) Illegal Instruction. Tried on my FC4 laptop, worked just fine. ___ --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 -- Roger Hill07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --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 ---End of Original Message- ___ --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 ___ --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] Newbie question. (Long)
Roger Hill wrote: I've been messing with Asterisk for a couple of weeks now. 1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), downloaded the binary package. Now I'm trying to put the working installation on my production server along with HTTP etc. ( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 GNU/Linux). That box, until yesterday, was running Fedora core 3. I tried the tarball download of 1.2.0.rc2, ran make OK, then make install, make samples. When I tried to run Asterisk, I got (immediately) Illegal Instruction. Tried on my FC4 laptop, worked just fine. Concluded I needed FC4, so upgraded the server yesterday. Six hours later... Reran make clean, make... Same problem. Then tried 1.2.0; same problem. Then tried 1.0.9; same problem. Finally removed everything to do with asterisk, pulled dowm 1.2.0 tar ball again, and re-installed. Same old problem, illegal instruction. I suspect bad RAM. I'd memtest it. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] newbie question regarding asterisk
On Nov 14, 2005, at 6:21 AM, Markos Paraskevopulos wrote: Hello everyone, I’m new to VoIP and despite a lot of reading, I’m kind of more confused than before. I have following question – we currently have hardware Alcatel PBX and approx. 50 phones in the company. I was wondering if we would need to change the phone service provider, because they don’t provide VoIP services if we were about to switch to Asterisk instead of the Alcatel PBX? Or can Asterisk maintain current functionality plus adding VoIP by simply switching the alcatel pbx for Asterisk server? I hope I’m making at least a bit of sense. Thanks in advance for help Confused Markos Markos, The answer to your question is Maybe. It depends on how you connect your existing PBX to the PSTN, and it depends on what you want from your system. Asterisk is completely capable of connecting to standard analog and digital (T1/E1/PRI) phone circuits. You do not need to use VOIP to connect Asterisk to the phone network. However, how you will go about doing this depends on your call volume and budget. How many incoming/ outgoing phone lines you have, how much long distance you dial, and local telco rates all play a part here. The easiest way is to figure out how you connect the existing PBX, and then you can research to see if Asterisk will support that technology. (Chances are that it does). For example, if your Alacatel connects to the PSTN via a T1/E1 Circuit, then you could buy an T1/E1 interface card from Digium or Sangoma and plug the T1/E1 right into your Asterisk server. If you have multiple analog POTS lines, then it's more complicated, but there are solutions for that, too (digium X100P, TDM400p, TDM2400p, various SIP gateways, multiple Sipura SPA-3000, etc...) Then you might want to research your other options and make sure that you are using the most cost effective solution for your needs (This all depends on how you use the PSTN and what the local rates and availability are). The most basic knowledge you will need is the difference between a T1/E1 style connection and a regular analog POTS line. For example, if you have multiple analog lines, you might be able to save money by getting a full or fractional T1/E1. If you're still completely confused and you don't have a lot of telecom knowledge, you might want to consider hiring a consultant to help you out. Tom -- Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 Technology solutions for small and medium sized businesses. ___ --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] newbie question regarding asterisk
Hello everyone, Im new to VoIP and despite a lot of reading, Im kind of more confused than before. I have following question we currently have hardware Alcatel PBX and approx. 50 phones in the company. I was wondering if we would need to change the phone service provider, because they dont provide VoIP services if we were about to switch to Asterisk instead of the Alcatel PBX? Or can Asterisk maintain current functionality plus adding VoIP by simply switching the alcatel pbx for Asterisk server? I hope Im making at least a bit of sense. Thanks in advance for help Confused Markos ___ --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] newbie question regarding asterisk
On Mon, 2005-11-14 at 12:21 +0100, Markos Paraskevopulos wrote: Hello everyone, I’m new to VoIP and despite a lot of reading, I’m kind of more confused than before. I had an asterisk system up and running then read some dox and becuase what I read at that time wasnt well written it has that effect :) asteriskdocs.org is pretty good, and the oreilly book asterisk and the future of telephony (pdf is available at asteriskdocs.org) is a good read, and only takes about 1 night to read everything except the appendices. I have following question – we currently have hardware Alcatel PBX and approx. 50 phones in the company. I was wondering if we would need to change the phone service provider, because they don’t provide VoIP services if we were about to switch to Asterisk instead of the Alcatel PBX? Asterisk does more than VoIP. It can speak analog (both fxs and fxo - or act like a phone company (fxs) or act like a phone (fxo)), it can also do digital trunks (t1/e1/j1 - ds3 soon alledgly). While it can replace a pbx it can also provide a T1 to a pbx. It can talk to the phone company via VoIP or whatever circuits you already have. You dont *have* to switch phone companies if you dont want to, and it doesnt always make sense to switch. Or can Asterisk maintain current functionality plus adding VoIP by simply switching the alcatel pbx for Asterisk server? If you want to add VoIP you can do this more gradually if you dont have the budget to totally replace 100%. Asterisk can feed your current pbx with phone service, where it interconnects to can be either VoIP or PSTN or both. Eventually you can migrate off what you already have. If however you have the budget to replace every phone on the desktop (or get appropriate interface equipment so the phones can speak to asterisk) then asterisk should be able to maintain current functionality plus adding anything that it does that you dont have (ie VoIP). I hope I’m making at least a bit of sense. I hope my answer makes sense.. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning. Since all inbound calls come through my T1, I would like to setup a dial plan that handles the incoming call and tells the caller to enter the extension they wish to reach. All of my real extensions are in the [default] context, and the Zaptel is configured to go to the [default-incoming] context. It is the [default-incoming] context that I am unsure of how to set. If anyone could provide some insight, it would be much appreciated.get me started at least. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[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] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan This is how I do it. [default-incoming]exten = 2691,1,Goto(extensions,3212,1)exten = 2692,1,Goto(extensions,3204,1)exten = 2693,1,Goto(extensions,3207,1)exten = 2694,1,Goto(extensions,3212,1)exten = 2695,1,Goto(extensions,3205,1)exten = 2696,1,Goto(extensions,3208,1)exten = 2697,1,Goto(extensions,1105,1)exten = 3211,1,Goto(extensions,,1)exten = 3223,1,Goto(extensions,3207,1) You will have to know how many digits are being sent, in my case it is four. So for example, someone dials the DID xxx-xxx-2691 the dialplan matches on the first line and then sends the cal to the context "extensions" (you would replace with default) and the extension "3212" in the first priority. If more or less digits are being sent by the telco, you will have to adjust the exten = to match. Sometimes they send three. Thanks, Steve Totaro - Original Message - From: Dave Morrow To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, October 18, 2005 11:26 AM Subject: [Asterisk-Users] Newbie Question: Help with incoming dial plan Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning. Since all inbound calls come through my T1, I would like to setup a dial plan that handles the incoming call and tells the caller to enter the extension they wish to reach. All of my real extensions are in the [default] context, and the Zaptel is configured to go to the [default-incoming] context. It is the [default-incoming] context that I am unsure of how to set. If anyone could provide some insight, it would be much appreciated .get me started at least. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.12.1/136 - Release Date: 10/15/2005 ___ --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] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan I do not use any DID, all calls come in on the same number 111222 so what I would like to do is simply prompt the caller to enter the extension they wish to reach, then redirect to that extension in the [default] context. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: Wednesday, October 19, 2005 11:34 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Newbie Question: Help with incoming dial plan This is how I do it. [default-incoming]exten = 2691,1,Goto(extensions,3212,1)exten = 2692,1,Goto(extensions,3204,1)exten = 2693,1,Goto(extensions,3207,1)exten = 2694,1,Goto(extensions,3212,1)exten = 2695,1,Goto(extensions,3205,1)exten = 2696,1,Goto(extensions,3208,1)exten = 2697,1,Goto(extensions,1105,1)exten = 3211,1,Goto(extensions,,1)exten = 3223,1,Goto(extensions,3207,1) You will have to know how many digits are being sent, in my case it is four. So for example, someone dials the DID xxx-xxx-2691 the dialplan matches on the first line and then sends the cal to the context "extensions" (you would replace with default) and the extension "3212" in the first priority. If more or less digits are being sent by the telco, you will have to adjust the exten = to match. Sometimes they send three. Thanks, Steve Totaro - Original Message - From: Dave Morrow To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, October 18, 2005 11:26 AM Subject: [Asterisk-Users] Newbie Question: Help with incoming dial plan Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning. Since all inbound calls come through my T1, I would like to setup a dial plan that handles the incoming call and tells the caller to enter the extension they wish to reach. All of my real extensions are in the [default] context, and the Zaptel is configured to go to the [default-incoming] context. It is the [default-incoming] context that I am unsure of how to set. If anyone could provide some insight, it would be much appreciated.get me started at least. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.12.1/136 - Release Date: 10/15/2005 ___ --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] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan
exten =
s,1,Answerexten = s,2,Wait,2exten =
s,3,Background(enter-ext-of-person)exten = s,4,DigitTimeout,5exten
= s,5,ResponseTimeout,10
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
MorrowSent: 18 October 2005 16:41To: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] Newbie Question: Help with incoming dial
plan
I do not use any DID, all calls come in on the same
number 111222 so what I would like to do is simply prompt the caller to
enter the extension they wish to reach, then redirect to that extension in the
[default] context.
David A. Morrow
Technical Systems
Lead Autodata
Solutions Company [EMAIL PROTECTED] http://www.autodata.net
Tel: (519) 951-6079
Fax: (519) 451-6615
Poor planning
on your part does not necessarily constitute an emergency on my part!
This message has originated from
Autodata Solutions. The attached material is the Confidential and Proprietary
Information of Autodata Solutions. This email and any files transmitted with
it are confidential and intended solely for the use of the individual or
entity to whom they are addressed. If you have received this email in error
please delete this message and notify the Autodata system administrator at
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
asteriskSent: Wednesday, October 19, 2005 11:34
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] Newbie Question: Help with
incoming dial plan
This is how I do it.
[default-incoming]exten =
2691,1,Goto(extensions,3212,1)exten =
2692,1,Goto(extensions,3204,1)exten =
2693,1,Goto(extensions,3207,1)exten =
2694,1,Goto(extensions,3212,1)exten =
2695,1,Goto(extensions,3205,1)exten =
2696,1,Goto(extensions,3208,1)exten =
2697,1,Goto(extensions,1105,1)exten =
3211,1,Goto(extensions,,1)exten =
3223,1,Goto(extensions,3207,1)
You will have to know how many digits are being
sent, in my case it is four. So for example, someone dials the DID
xxx-xxx-2691 the dialplan matches on the first line and then sends the cal to
the context "extensions" (you would replace with default) and the extension
"3212" in the first priority.
If more or less digits are being sent by the
telco, you will have to adjust the exten = to match. Sometimes
they send three.
Thanks,
Steve Totaro
- Original Message -
From:
Dave Morrow
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, October 18, 2005 11:26
AM
Subject: [Asterisk-Users] Newbie
Question: Help with incoming dial plan
Hi all. I just got Asterisk installed with
a Digium TE110P T1 card. Have it working for outbound calls so I know
that all the hardware is functioning.
Since all inbound calls come through my T1, I
would like to setup a dial plan that handles the incoming call and tells the
caller to enter the extension they wish to reach. All of my real
extensions are in the [default] context, and the Zaptel is configured to go
to the [default-incoming] context. It is the [default-incoming]
context that I am unsure of how to set.
If anyone could provide some insight, it would be
much appreciated.get me started at least.
David A. Morrow Technical Systems Lead Autodata
Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519)
451-6615
Poor planning on your part
does not necessarily constitute an emergency on my part!
This message has originated from Autodata
Solutions. The attached material is the Confidential and Proprietary
Information of Autodata Solutions. This email and any files transmitted with
it are confidential and intended solely for the use of the individual or
entity to whom they are addressed. If you have received this email in error
please delete this message and notify the Autodata system administrator
at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
___--Bandwidth and
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Re: [Asterisk-Users] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan
add this context
[default-incoming]exten =
111222,1,Goto(default-incoming,s,1)
exten = s,1,Answerexten =
s,2,DigitTimeout(10)exten = s,3,ResponseTimeout(20)exten =
s,4,Background(swelcome)exten = t,1,Hangupinclude =
extensions
add this to your extensions context
;directory appexten =
9,1,Directory(default-extensions)
; exten for recording greetings/menusexten
= 12,1,Authenticate(1234|)exten = 12,2,Wait(2)exten =
12,3,Record(/var/lib/asterisk/sounds/welcome:gsm)exten =
12,4,Wait(2)exten =
12,5,Playback(/var/lib/asterisk/sounds/welcome)exten =
12,6,Wait(2)exten = 12,7,Hangup
Reload and dial 12 with the password of 1234 and
record your greeting and then hangup. If you mess up just do it
over.
Thanks,
Steve
- Original Message -
From:
Dave Morrow
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, October 18, 2005 11:41
AM
Subject: RE: [Asterisk-Users] Newbie
Question: Help with incoming dial plan
I do not use any DID, all calls come in on the same
number 111222 so what I would like to do is simply prompt the caller to
enter the extension they wish to reach, then redirect to that extension in the
[default] context.
David A. Morrow
Technical Systems
Lead Autodata
Solutions Company [EMAIL PROTECTED] http://www.autodata.net
Tel: (519) 951-6079
Fax: (519) 451-6615
Poor planning
on your part does not necessarily constitute an emergency on my part!
This message has originated from
Autodata Solutions. The attached material is the Confidential and Proprietary
Information of Autodata Solutions. This email and any files transmitted with
it are confidential and intended solely for the use of the individual or
entity to whom they are addressed. If you have received this email in error
please delete this message and notify the Autodata system administrator at
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
asteriskSent: Wednesday, October 19, 2005 11:34
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] Newbie Question: Help with
incoming dial plan
This is how I do it.
[default-incoming]exten =
2691,1,Goto(extensions,3212,1)exten =
2692,1,Goto(extensions,3204,1)exten =
2693,1,Goto(extensions,3207,1)exten =
2694,1,Goto(extensions,3212,1)exten =
2695,1,Goto(extensions,3205,1)exten =
2696,1,Goto(extensions,3208,1)exten =
2697,1,Goto(extensions,1105,1)exten =
3211,1,Goto(extensions,,1)exten =
3223,1,Goto(extensions,3207,1)
You will have to know how many digits are being
sent, in my case it is four. So for example, someone dials the DID
xxx-xxx-2691 the dialplan matches on the first line and then sends the cal to
the context "extensions" (you would replace with default) and the extension
"3212" in the first priority.
If more or less digits are being sent by the
telco, you will have to adjust the exten = to match. Sometimes
they send three.
Thanks,
Steve Totaro
- Original Message -
From:
Dave Morrow
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, October 18, 2005 11:26
AM
Subject: [Asterisk-Users] Newbie
Question: Help with incoming dial plan
Hi all. I just got Asterisk installed with
a Digium TE110P T1 card. Have it working for outbound calls so I know
that all the hardware is functioning.
Since all inbound calls come through my T1, I
would like to setup a dial plan that handles the incoming call and tells the
caller to enter the extension they wish to reach. All of my real
extensions are in the [default] context, and the Zaptel is configured to go
to the [default-incoming] context. It is the [default-incoming]
context that I am unsure of how to set.
If anyone could provide some insight, it would be
much appreciated
.get me started at least.
David A. Morrow Technical Systems Lead Autodata
Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519)
451-6615
Poor planning on your part
does not necessarily constitute an emergency on my part!
This message has originated from Autodata
Solutions. The attached material is the Confidential and Proprietary
Information of Autodata Solutions. This email and any files transmitted with
it are confidential and intended solely for the use of the individual or
entity to whom they are addressed. If you have received this email in error
please delete this message and notify the Autodata system administrator
at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
___--Bandwidth and
RE: [Asterisk-Users] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan
Thanks Steve, this works like a charm!
Might I ask how I setup that Directory?
David A. Morrow
Technical Systems
Lead Autodata
Solutions Company [EMAIL PROTECTED] http://www.autodata.net
Tel: (519) 951-6079
Fax: (519) 451-6615
Poor planning on
your part does not necessarily constitute an emergency on my part!
This message has originated from
Autodata Solutions. The attached material is the Confidential and Proprietary
Information of Autodata Solutions. This email and any files transmitted with it
are confidential and intended solely for the use of the individual or entity to
whom they are addressed. If you have received this email in error please delete
this message and notify the Autodata system administrator at
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
asteriskSent: Wednesday, October 19, 2005 11:51 AMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Newbie Question: Help with incoming dial
plan
add this context
[default-incoming]exten =
111222,1,Goto(default-incoming,s,1)
exten = s,1,Answerexten =
s,2,DigitTimeout(10)exten = s,3,ResponseTimeout(20)exten =
s,4,Background(swelcome)exten = t,1,Hangupinclude =
extensions
add this to your extensions context
;directory appexten =
9,1,Directory(default-extensions)
; exten for recording greetings/menusexten
= 12,1,Authenticate(1234|)exten = 12,2,Wait(2)exten =
12,3,Record(/var/lib/asterisk/sounds/welcome:gsm)exten =
12,4,Wait(2)exten =
12,5,Playback(/var/lib/asterisk/sounds/welcome)exten =
12,6,Wait(2)exten = 12,7,Hangup
Reload and dial 12 with the password of 1234 and
record your greeting and then hangup. If you mess up just do it
over.
Thanks,
Steve
- Original Message -
From:
Dave Morrow
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, October 18, 2005 11:41
AM
Subject: RE: [Asterisk-Users] Newbie
Question: Help with incoming dial plan
I do not use any DID, all calls come in on the same
number 111222 so what I would like to do is simply prompt the caller to
enter the extension they wish to reach, then redirect to that extension in the
[default] context.
David A. Morrow
Technical Systems
Lead Autodata
Solutions Company [EMAIL PROTECTED] http://www.autodata.net
Tel: (519) 951-6079
Fax: (519) 451-6615
Poor planning
on your part does not necessarily constitute an emergency on my part!
This message has originated from
Autodata Solutions. The attached material is the Confidential and Proprietary
Information of Autodata Solutions. This email and any files transmitted with
it are confidential and intended solely for the use of the individual or
entity to whom they are addressed. If you have received this email in error
please delete this message and notify the Autodata system administrator at
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
asteriskSent: Wednesday, October 19, 2005 11:34
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] Newbie Question: Help with
incoming dial plan
This is how I do it.
[default-incoming]exten =
2691,1,Goto(extensions,3212,1)exten =
2692,1,Goto(extensions,3204,1)exten =
2693,1,Goto(extensions,3207,1)exten =
2694,1,Goto(extensions,3212,1)exten =
2695,1,Goto(extensions,3205,1)exten =
2696,1,Goto(extensions,3208,1)exten =
2697,1,Goto(extensions,1105,1)exten =
3211,1,Goto(extensions,,1)exten =
3223,1,Goto(extensions,3207,1)
You will have to know how many digits are being
sent, in my case it is four. So for example, someone dials the DID
xxx-xxx-2691 the dialplan matches on the first line and then sends the cal to
the context "extensions" (you would replace with default) and the extension
"3212" in the first priority.
If more or less digits are being sent by the
telco, you will have to adjust the exten = to match. Sometimes
they send three.
Thanks,
Steve Totaro
- Original Message -
From:
Dave Morrow
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, October 18, 2005 11:26
AM
Subject: [Asterisk-Users] Newbie
Question: Help with incoming dial plan
Hi all. I just got Asterisk installed with
a Digium TE110P T1 card. Have it working for outbound calls so I know
that all the hardware is functioning.
Since all inbound calls come through my T1, I
would like to setup a dial plan that handles the incoming call and tells the
caller to enter the extension they wish to reach. All of my real
extensions are in the [default] context, and the Zaptel is configured to go
to t
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
RE: [Asterisk-Users] Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone
It would be nice if there was a dialplan for each registration or line, which would allow me to never press send for any of the systems I register to On my Sipura there isI haven't checked one of the polycoms but I suspect they are no different. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: Thursday, August 11, 2005 8:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone Jonathan k. Creasy wrote: YeahI think that every install I have done the first thing that happens is why is there a delay before the call connects? and the answer is you have to hit dial or wait 10 seconds. What all phones does that apply to? I'm fairly certain it applies to the Polycom phones I've read about, but I'm not sure about others. I'm obviously a newbie to the field as well (well, at least to the physical phones). Actually, I never need to press send when dialling numbers from my polycom phone... well, actually I do, but 99% of people wouldn't... I'll explain: My sip phone registers to different SIP servers (all asterisk), and each server has it's own dialplan, ie, one is purely PSTN calls, and so there is no leading digit for 'external calls', you just dial the number. One is a PBX where you need to dial 9 to get an outside line, with 3 digit extensions. Another is a PBX where you need to dial 0 to get an outside line, with 4 digit extensions. Also, each PBX system has different voicemail numbers to get to voicemailmain etc... So, I don't press send when dialling through my own PBX, but I do if I am dialling through one of the other systems, since their dialplans are not catered for in my polycom. BTW, it would be nice if there was a dialplan for each registration or line, which would allow me to never press send for any of the systems I register to:) Regards, Adam ___ 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 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 anAsterisk systemtoreplace an old PBX but using existing phone
Which is where the dial 9 for an outside line came from. Just make sure no internal extensions start with 9. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tom Rymes Sent: Thursday, August 11, 2005 3:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone The difficulty is making the phone dial quickly when you dial a three or four digit extension number, yet not having it dial so quickly that it screws up a user who dials the first four digits of a 10 digit number and has to look down at a piece of paper to read the last 6 digits. It's pretty annoying when that happens and the phone has already initiated the call. (I would make the phone wait either one or two seconds to match an internal extension.) Tom On Aug 11, 2005, at 5:11 PM, Tarpo, Louie wrote: You write out a dialplan, then when you match a pattern in the dial plan, the Polycom will initiate the call immediately. This way you can have 4 digit internal extensions dial immediately, or have it wait for a long distance or international number. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew M Stemen Sent: Thursday, August 11, 2005 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk systemtoreplace an old PBX but using existing phone Jonathan k. Creasy wrote: YeahI think that every install I have done the first thing that happens is why is there a delay before the call connects? and the answer is you have to hit dial or wait 10 seconds. What all phones does that apply to? I'm fairly certain it applies to the Polycom phones I've read about, but I'm not sure about others. I'm obviously a newbie to the field as well (well, at least to the physical phones). ___ 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 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 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 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
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. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Rima Sent: August 11, 2005 8:34 AM To: Asterisk-Users@lists.digium.com Subject: [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 ___ 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
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. -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 8/11/05, Sean Rima [EMAIL PROTECTED] 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. 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 ___ 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 -- Tom ___ 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
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. -A. ___ 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
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. -A. ___ 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
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. Tom ___ 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
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. Tom Cascade Link Systems ___ 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 systemto replace an old PBX but using existing phone
YeahI think that every install I have done the first thing that happens is why is there a delay before the call connects? and the answer is you have to hit dial or wait 10 seconds. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes Sent: Thursday, August 11, 2005 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk systemto replace an old PBX but using existing phone 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. Tom Cascade Link Systems ___ 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 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 systemto replace an old PBX but using existing phone
My thoughts exactly, stick with whats there and use a channel bank. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tom Rymes Sent: Thursday, 11 August 2005 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk systemto replace an old PBX but using existing phone 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. Tom Cascade Link Systems ___ 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 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 systemto replace an old PBX but using existing phone
Jonathan k. Creasy wrote: YeahI think that every install I have done the first thing that happens is why is there a delay before the call connects? and the answer is you have to hit dial or wait 10 seconds. What all phones does that apply to? I'm fairly certain it applies to the Polycom phones I've read about, but I'm not sure about others. I'm obviously a newbie to the field as well (well, at least to the physical phones). ___ 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 systemtoreplace an old PBX but using existing phone
The only phones I have much experience with are the sipura spa-841's, the netweb 301/302 phones (which I really don't like) and the polycom 300/301's. It applies to the sipuras and the polycom's for sure. I can't remember about the netweb, we quit using them sometime last year. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew M Stemen Sent: Thursday, August 11, 2005 5:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk systemtoreplace an old PBX but using existing phone Jonathan k. Creasy wrote: YeahI think that every install I have done the first thing that happens is why is there a delay before the call connects? and the answer is you have to hit dial or wait 10 seconds. What all phones does that apply to? I'm fairly certain it applies to the Polycom phones I've read about, but I'm not sure about others. I'm obviously a newbie to the field as well (well, at least to the physical phones). ___ 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 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 systemtoreplace an old PBX but using existing phone
You write out a dialplan, then when you match a pattern in the dial plan, the Polycom will initiate the call immediately. This way you can have 4 digit internal extensions dial immediately, or have it wait for a long distance or international number. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew M Stemen Sent: Thursday, August 11, 2005 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk systemtoreplace an old PBX but using existing phone Jonathan k. Creasy wrote: YeahI think that every install I have done the first thing that happens is why is there a delay before the call connects? and the answer is you have to hit dial or wait 10 seconds. What all phones does that apply to? I'm fairly certain it applies to the Polycom phones I've read about, but I'm not sure about others. I'm obviously a newbie to the field as well (well, at least to the physical phones). ___ 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 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 anAsterisk systemtoreplace an old PBX but using existing phone
You are right. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tarpo, Louie Sent: Thursday, August 11, 2005 5:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone You write out a dialplan, then when you match a pattern in the dial plan, the Polycom will initiate the call immediately. This way you can have 4 digit internal extensions dial immediately, or have it wait for a long distance or international number. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew M Stemen Sent: Thursday, August 11, 2005 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk systemtoreplace an old PBX but using existing phone Jonathan k. Creasy wrote: YeahI think that every install I have done the first thing that happens is why is there a delay before the call connects? and the answer is you have to hit dial or wait 10 seconds. What all phones does that apply to? I'm fairly certain it applies to the Polycom phones I've read about, but I'm not sure about others. I'm obviously a newbie to the field as well (well, at least to the physical phones). ___ 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 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 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 anAsterisk systemtoreplace an old PBX but using existing phone
You write out a dialplan, then when you match a pattern in the dial plan, the Polycom will initiate the call immediately. This way you can have 4 digit internal extensions dial immediately, or have it wait for a long distance or international number. Ah... OK. Sounds like it's similiar to the 3Com VOIP Phones I've worked with then... they just relied on a dialplan to tell them when to complete the call. ___ 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
