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
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
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
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
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
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
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
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
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
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
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
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
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
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 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 NOTICE: This e-mail message and all attachments transmitted with it may contain legally privileged and confidential information intended solely for the
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
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
Re: [Asterisk-Users] Newbie Question: Building an Asterisk systemto replace an old PBX but using existing phone
Well, it applies to many phones, such as the Cisco and Polycoms, among others, but generally, there is a way to define a dialplan that changes the amount of time you have to wait for the phone to assume that you are done dialing. (ie: if it sees 10 digits, wait one second, and if it sees 11 digits, don't wait at all.) [EMAIL PROTECTED] has a great web interface for configuring these settings for Cisco phones, too. Tom On Aug 11, 2005, at 5:06 PM, Andrew M Stemen wrote: 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
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
RE: [Asterisk-Users] Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone
I think the Sipura's will do this too, I just didn't know itlearn something every day :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes Sent: Thursday, August 11, 2005 5:35 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
Re: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone
Tom Rymes wrote: On Aug 11, 2005, at 11:49 AM, Sean Rima wrote: Tom Rymes wrote: On Aug 11, 2005, at 10:35 AM, Sean Rima wrote: Andrew Kohlsmith wrote: On Thursday 11 August 2005 09:31, Sean Rima wrote: They are standard phones but I also want them to have all the features that Asterisk does provide, so I may build a bos for my house and show them that as well Sean, The client has a good idea in keeping the basic analog phones, since all of their guests know how to use them. If you put a SPA-841 or a Polycom IP301, you will likely intimidate the guests. This might sound silly to most of us techies, but picture a 74 year old guest just trying to call and let his children know he arrived safely. He might very well look at the Polycom and be confused, especially if he dials and has to press send, etc. Also, if they already have the phones, the per channel cost of using a T1 card and a channel bank is reasonably low, compared to a decent SIP hardphone. Not to mention that if you don't spend the extra money for a good hardphone, you are likely to have quality issues, as I have heard many quality complaints about many of the cheaper phones. Finally, sticking with the analog phones means you won't have to re- wire the whole place. I never thought of it this way, I will prive up a T1 card and bank, possibly on ebay as well as there are a a few items there for sale Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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). 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
RE: [Asterisk-Users] Newbie Question: Building an Asterisk system toreplace an old PBX but using existing phone
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. Regards, Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Sean Rima Sent: Thursday, August 11, 2005 10:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk system toreplace 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 ___ 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: Type of card
On Thu, 7 Jul 2005 10:49:32 -0700 Dan Adams [EMAIL PROTECTED] wrote: Hi, I am sorta a newbie to the asterisk community at least in the realm of hardware types. I was wondering, what type of card is used to allow asterisk, on a slackware installation to talk to a standard phone line so that asterisk can call out? Dan The link below gives you great information on the card you need. Look espicially close to the box with all the writing in it just below the URL to www.asterisk.org that starts with For interconnection with digital and analog telephony equipment http://www.voip-info.org/wiki-Asterisk ___ 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: Type of card
Take a look here: http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TDM400P MARK. Dan Adams wrote: Hi, I am sorta a newbie to the asterisk community at least in the realm of hardware types. I was wondering, what type of card is used to allow asterisk, on a slackware installation to talk to a standard phone line so that asterisk can call out? Dan ___ 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: Type of card
Here is what I use: http://www.digitnetworks.com/store/product_info.php?cPath=22products_id=28 I have used it with Slack, but now I am running it with FC4. -Original Message- From: Dan Adams [mailto:[EMAIL PROTECTED] Sent: Thursday, July 07, 2005 12:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Newbie Question: Type of card Hi, I am sorta a newbie to the asterisk community at least in the realm of hardware types. I was wondering, what type of card is used to allow asterisk, on a slackware installation to talk to a standard phone line so that asterisk can call out? Dan ___ 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 - A.G. Edwards Sons' outgoing and incoming e-mails are electronically archived and subject to review and/or disclosure to someone other than the recipient. - ___ 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 reg. Asterisk and Channel Access Bank I and TE110p
Recheck your zaptel.conf. That's not the correct setup for a T1 trunk. You need to know the signalling the channel bank uses, and specify the voice channels (bchannel=1-24), and the signalling channel (dchannel=25). Those numbers are bogus, as I've never worked with T1 ;) BTW, why are you using such setup (1 channel bank to connect to 24 analog lines) instead of asking your Telco to install a T1 trunk in your office? Julian. On 7/5/05, Mehran Mozaffari [EMAIL PROTECTED] wrote: Hi, I have some problem to get this setup working. I have a CAC Channel Banl I, with FXO and an Asterisk box ( I am using [EMAIL PROTECTED] 1.2) and I have a TE110p installed in this box. What I want to do is, just to be able to dial one of those lines that already are connected to the channel bank, and transfer that call through TE110p and Asterisk to a user agent somewhere through Internet. PSTN-CAC Channel Bank I-TE110p-Asterisk-SIP User Agent At this time the SIP UAs can communicate with each other and everything works properly, but I can't dial through channel bank. When I dial one of those numbers, I will get no answer ring, and I can't see anything coming to Asterisk through CLI. and when I tried to dial through SIP UA to the PSTN end, I will get all circuits are busy now from asterisk. Any Idea what should I do? at this time all lights are green and it looks like that everything is working properly, but I am not sure where is the problem, here are my settings: /etc/zaptel.conf: span=1,1,0,esf,b8zs fxols=1-24 ___ 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..
hi Rich, thanks for ur help.. it works.. i have found another way, _9XXX,1,Dial(Zap/4/1800XX,5,D(${EXTEN})) D = will send dtmf thank a lot Rich.. best regard, shahdan --- Rich Adamson [EMAIL PROTECTED] wrote: the situation here is i want when user make outgoing call, asterisk will call 1800XX first then after 3 or 4 sec asterisk will insert the number that user want to call.. user don't know that the call is go to 1800XX first.. means user just insert the number that they want to call then asterisk will insert that number after 3 or 4 sec.. can i that in asterisk? i'll apreciate any help or advise.. Might try something like this: exten = _9XXX,1,Dial(Zap/4/1800XXw${EXTEN}) where each w adds a some delay. ___ 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 __ Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html ___ 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 about pressing a key to, be connected to the caller
You don't have to use queues to use agents. Do a show application dial and look at what he is showing you. You can have a macro run upon answer so put your menu there. Kevin ___ 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..
the situation here is i want when user make outgoing call, asterisk will call 1800XX first then after 3 or 4 sec asterisk will insert the number that user want to call.. user don't know that the call is go to 1800XX first.. means user just insert the number that they want to call then asterisk will insert that number after 3 or 4 sec.. can i that in asterisk? i'll apreciate any help or advise.. Might try something like this: exten = _9XXX,1,Dial(Zap/4/1800XXw${EXTEN}) where each w adds a some delay. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question about pressing a key to be connected to the caller
Check out ackcall=yes in agents.conf It allows them to press # to accept, or press * to not accept. then you can do something like: exten = 101,1,Dial(Agent/101,20,A(presspoundtoanswer)) or if you want to get more fancy, check out queues.conf where you can set ring orders and answer penalties. Hope this helps, Jon. On Tuesday 14 June 2005 09:18 pm, Jason wrote: I have a newbie question about the dialplan. I have a main menu that picks up on a certain DID number, and gives a list of options. When an option is selected, for instance 1 for sales, it rings a number of users in succession until one picks up and is connected to the caller, otherwise it goes to voicemail. This is all working well. However, I would like to have the system play a message to the user when they pick up, saying There is a call for sales, press 1 to accept the call or 2 to ignore. If the user pressed 1 they would then be connected to the person calling in, if 2 it would just go to the next person in the group. Any help would be appreciated, my current context is below for reference: [ext-sales] exten = 1,1,Answer(); exten = 1,2,SetCIDName(Sales); exten = 1,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?ext-cs-vm,1,1) exten = 1,4,Playback(custom/PleaseWait) exten = 1,5,Macro(dial,${RINGTIMER},tm,717-6197585949#) exten = 1,6,Macro(dial,${RINGTIMER},tm,707-6199208398#) exten = 1,7,Macro(dial,${RINGTIMER},tm,8323686410#) exten = 1,8,Macro(dial,${RINGTIMER},tm,717-6197585949#) exten = 1,9,Macro(dial,${RINGTIMER},tm,707-6199208398#) exten = 1,10,Macro(dial,${RINGTIMER},tm,8323686410#) exten = 1,11,Goto(ext-sales-vm,1,1) ___ 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
read in voip-info.org about Asterisk Call Manager API, and may be an easier soultion are the .call files that you can pleace in /var/spool/asterisk/outgoing/ these files have a description of the type of call you wanna make, in the very moment that you place the file there, a call will be Originated automagically. With Asterisk Call Manager API you can do something similar ( or equal ) using the Originate action. best regards On 6/8/05, Charles Austin [EMAIL PROTECTED] wrote: Greetings, I have my first asterisk installation up and running, thanks to a lot of reading. Could anyone point me in the direction of things to read on automated outbound dialing? NOT predictive dialing - I will not have agents handling the calls. These calls are reminders for appointments, etc. Thanks! Charles ___ 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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ 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: HOWTO make outgoing call on SIP account from internal extensions?
Maybe you should review these: http://asteriskdocs.org http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html I've never seen the register line you have used, the ones I see are mostly like this: register = username:[EMAIL PROTECTED]/5551212 where 5551212 is an extension. That extension has a Dial() that rings whatever phones you like such as exten = 5551212,1,Dial(${OFFICE}${HOME},45) and OFFICE and HOME are definitions in the extensions.conf I have about 12 voip server providers and every single one, even the free ones usually provide a sample asterisk config. Oops, looks like stanaphone did not. Try a free FWD account using either SIP or IAX2. ___ 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: HOWTO make outgoing call on SIP account from internal extensions?
Steve wrote: I have read LOTS of docs and played quite a bit to get this far Good, keep playing!!! (a lot of your typing time deleted) OK here's what messes it all up (and I admit I'm clueless here) register = 2135551212:[EMAIL PROTECTED] [sipproviderexample.com] type=peer host=10.77.77.133 fromuser=2135551212 secret=2135551212 fromdomain=sipproviderexample.com adding this secttion breaks it and I really do not understand what it's even for... What does it mean for you, that this breaks it. Did it work before? What is your jump in point to the dial plan? You do not have a context= line. So you may jump into the context=default as usually mentioned in sip.conf [general] context=default; Default context for incoming calls Do you have something in the dialplan like: [from-sipproviderexample.com] ; if you would use this as your context= in sip.conf, otherwise the next lines in the default context of your extensions.conf exten = _X.,1,Dial(SIP/601,60,tr) ; if any number calling from this provider should ring extension 601 or exten = 2135551212 ,3,Dial(SIP/601,60,tr) ; if only a dialed in number 2135551212 of your provider should dial extension 601 Keep playing with: context=x and[x] exten = something with and without starting _ Hope it helps, ... bye Ronald does it work with the register line somehow? or is it totally seperate? what is it for? All the docs I have looked at seem to suggest adding this extra section but do not really seem to explain it or what exactly it does. I'm not sure what it's for or if it has anything to do with making outbound sip calls from the internal extensions. when I add it my sip provider account stops working and I get registration retries and timeouts without any successful registrations after that. I'm just looking for a good pointer in where to go for an example of how to use my provider account for outbound connections... I understand the dialplans themselves but do not know how to associate them with the actualy sip provider account for an outbound call. Thanks I'll keep reading until I figure these things out but any pointers to specific documentation that answers any of these questions would be very much appreciated I *think* I am familiar with just about all of the standard asterisk documentation I have been able to find... More than likely I am just missing some key points that I have read but have misinterpreted! Take care! Steve ___ 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 -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ 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: HOWTO make outgoing call on SIP account from internal extensions?
Ok I'm still playing and the way it's supposed to work is making much more sense now. However it is still 'not working' as soon as I add this: [sipproviderexample.com] type=peer host=sipprovider.com fromuser=2135551212 secret=2135551212 fromdomain=sipproviderexample.com to sip.conf I also learned that I needed to replace the internal IP host= with sipproviderexample.com for whatever reason the example I had been working off of was showing an internal address for that What is breaking is that the asterisk box stops accepting ANY inbound calls via sipprovider.com as soon as I add the extension noted above. I have also tried many variations of that... and using another context besides default. No change. It flat out 'stops working' no inbound calls (from PTSN via sip provider) as soon as I add that to sip.conf and associate it with any ANY context. even associating it with an empty context (no dialplan lines) in extensions.conf it behaves the same way. If I comment it out inbound calls work perfectly well all day long I'm baffled and it's very frusterating! if I attempt an inbound call in this non-working state from PSTN this is what I see with sip debug: Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKdf72.63b2d483.1 From: unknown sip:[EMAIL PROTECTED]:5060;user=phone;tag=e2d90014-1cffac42-8cd690d8 Call-ID: [EMAIL PROTECTED] To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as51e3a2ab CSeq: 1131072 ACK User-Agent: SIPProxy Content-Length: 0 - About 5 seconds later this comes back 8 headers, 0 lines Destroying call '[EMAIL PROTECTED]' remove those few lines, reload and incoming works just fine again. I also tried this with a different provider (stanaphone) and that behaves the same way. Anything else to try? Thanks for all the help and this is EXTREMELY cool! I am playing and having a blast wit hit Sure would like to be able to make some outbound calls! :-) Take care! Steve On Thu, 2 Jun 2005, Ronald Wiplinger wrote: Steve wrote: I have read LOTS of docs and played quite a bit to get this far Good, keep playing!!! (a lot of your typing time deleted) OK here's what messes it all up (and I admit I'm clueless here) register = 2135551212:[EMAIL PROTECTED] [sipproviderexample.com] type=peer host=10.77.77.133 fromuser=2135551212 secret=2135551212 fromdomain=sipproviderexample.com adding this secttion breaks it and I really do not understand what it's even for... What does it mean for you, that this breaks it. Did it work before? What is your jump in point to the dial plan? You do not have a context= line. So you may jump into the context=default as usually mentioned in sip.conf [general] context=default; Default context for incoming calls Do you have something in the dialplan like: [from-sipproviderexample.com] ; if you would use this as your context= in sip.conf, otherwise the next lines in the default context of your extensions.conf exten = _X.,1,Dial(SIP/601,60,tr) ; if any number calling from this provider should ring extension 601 or exten = 2135551212 ,3,Dial(SIP/601,60,tr) ; if only a dialed in number 2135551212 of your provider should dial extension 601 Keep playing with: context=x and[x] exten = something with and without starting _ Hope it helps, ... bye Ronald does it work with the register line somehow? or is it totally seperate? what is it for? All the docs I have looked at seem to suggest adding this extra section but do not really seem to explain it or what exactly it does. I'm not sure what it's for or if it has anything to do with making outbound sip calls from the internal extensions. when I add it my sip provider account stops working and I get registration retries and timeouts without any successful registrations after that. I'm just looking for a good pointer in where to go for an example of how to use my provider account for outbound connections... I understand the dialplans themselves but do not know how to associate them with the actualy sip provider account for an outbound call. Thanks I'll keep reading until I figure these things out but any pointers to specific documentation that answers any of these questions would be very much appreciated I *think* I am familiar with just about all of the standard asterisk documentation I have been able to find... More than likely I am just missing some key points that I have read but have misinterpreted! Take care! Steve ___ 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
Re: [Asterisk-Users] Newbie question
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hamish Whittal wrote: Hi folks, I have an asuscom ISDN BRI card in my server and was wanting to know whether this would be good enough to use with Asterisk. I am VERY new to this, so have no idea how to config the software, etc. But I am very eager to learn. Cheers H This should work fine with ISDN4Linux as it uses the Hisax chipset. Get it working first as a normal ISDN card, then add Asterisk. Look at /etc/asterisk/modem.conf, it will need to contain something like this: [interfaces] context=inbound driver=i4l language=en type=autodetect stripmsd=0 dialtype=tone mode=immediate msn=yourMSNhere group=9 dtmfmode=asterisk incomingmsn=* device = /dev/ttyI0 device = /dev/ttyI1 You _WILL_ need to set your MSN (change yourMSNhere to your full MSN, usually without the area code). HTH - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD:519961 Gossiptel:9309811 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBQpOLWEtP/KMNOfRbAQKyMwf5AWyv903Zk04g9g/lNGIErfDdQHlo7Vr7 lThVhfMScoW5PlFI3UIVjjzikwZYsqYUBL0aEevFa64hV40u84ZzQ0WbgLzTekgB qPOaFXwzI//D7GQUOcKpA9ydlWYv7y16BQ2lgj93DP/xttImu6qf+HwF2vzhYMmE TxC2CUkU3zeMnFYy3QGBHPyVJVH+jRD5C3jgnnWfGeUVYBjQhVA3PQHZ+nLw/Ip+ 3Iv5R4wcDrtM1F08ZLkH1NJ2D07NvmkasglEyaJlCDBh6LGyBexuiSjUAn8cxH3Z 3cfmDfKtWjAFZV060LyAnQQYAmhrnDc88/HRbIzHTVJJ6/+uPduHMQ== =9O75 -END PGP 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 on how to handle main office number
Check out AMP to see how call groups are used. http://www.voip-info.org/wiki-Asterisk+Management+Portal You group your phones, available handsets ring. You can roll from group to group however you want. Just a matter of writing the correct dialplan. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Teagle Sent: Wednesday, April 13, 2005 8:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Newbie Question on how to handle main office number Hi we are looking to swap out an old version of Cisco Call Manager for asterisk and are trying to work out the best way to handle the main office number. We will have about 35 phones and we have PRI from Colt, we are london based. At present when a call is made to the main number and is not answered or is engaged we forward the call to a shared number that every phone in the office is registered to and so can answer. But this limits us to only receiving two calls at once to our main number. We like the way this works apart from the limit of two calls at once and are trying to work out the best way to implement this in asterisk. Although we have looked at queue's we don't want everybody to having login every morning ?. Would a solution where we forward calls from the main number to an extension that calls all the phones, be a solution ? exten = 2000,1,Dail(SIP/101,SIP/102,SIP/103 etc) Or would this get engaged after the first call is answered on it, if could I have a number of these that roll over to each other ? Or are there other ways to handle the main number ? Thanks for any help Nick ___ 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