Re: [asterisk-users] Getting started with sample dial plans
Now I'm ready to begin playing with dial plans and am having a difficult time getting started. You may want to read the book : http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 That should help you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting started with sample dial plans
Okay, I have Asterisk up and running on Fedora Core 5 with a TDM400 board with one FXO and FXS module. Zap is up and running and * is functioning with the modules. Oh yeah, and I have some soft phones configured and have them working as well. Now I'm ready to begin playing with dial plans and am having a difficult time getting started. I'm looking for some simple samples that might demonstrate basic functionality such as running the inside phone extension when an incoming call is received. Or, a simple Dial 9 for an outside line plan where whatever number is dialed (after the 9) is simply dialed via the Zap Line. In other words, something that makes * nearly transparent to begin with. Then, I'd like to slowly add to the dial plan as I learn more of the commands. I have not found any good samples of dial plans (other than the defaults built with *) that demonstrate basics like this. Are there some online references I could work through? I've been to Asterisk.com and their respective documentation, and Asteriskguru.com and not found what I'm looking for (maybe I overlooked it?). Any guidance is sincerely appreciated. -- Mitch ___ --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] getting started
Wanted some advice for the docs that you'd recommend someone new to Asterisk to read. I have a good knowledge of Unix and networking, so that part shouldn't be a problem. Try... http://www.asteriskdocs.org/modules/news/ The authors of Asterisk: The Future of Telephony are pleased to announce their book in PDF form, available immediately, for free. The book can be downloaded from www.asteriskdocs.org. Thanks to O'Reilly Media for supporting us and allowing us to publish the book under the Creative Commons license. Or, purchase the book from O'Reilly. I'd recommend it as an excellent starting point. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] getting started
Hi Guys, Wanted some advice for the docs that you'd recommend someone new to Asterisk to read. I have a good knowledge of Unix and networking, so that part shouldn't be a problem. Cheers, Sukrit.D. -- \|||/ (o o) +ooO-(_)-Ooo---+ | SUKRIT D| www.liqwidkrystal.com| | Email: sukrit-at-liqwidkrystal.com | |--| | MSN:[EMAIL PROTECTED] YAHOO:sd_root | | SKYPE:sukritd | +--+ ( _ ) _| | | |_ (___| |___) ___ --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] getting started
Wanted some advice for the docs that you'd recommend someone new to Asterisk to read. I have a good knowledge of Unix and networking, so that part shouldn't be a problem. http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 Welcome 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] getting started
check http://www.asteriskguru.com/tutorials/ Diyanat From: sukrit [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] getting started Date: Fri, 16 Dec 2005 09:06:28 +0530 MIME-Version: 1.0 Hi Guys, Wanted some advice for the docs that you'd recommend someone new to Asterisk to read. I have a good knowledge of Unix and networking, so that part shouldn't be a problem. Cheers, Sukrit.D. -- \|||/ (o o) +ooO-(_)-Ooo---+ | SUKRIT D| www.liqwidkrystal.com| | Email: sukrit-at-liqwidkrystal.com | |--| | MSN:[EMAIL PROTECTED] YAHOO:sd_root | | SKYPE:sukritd | +--+ ( _ ) _| | | |_ (___| |___) ___ --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] getting started
Hi sjkrit If you are beginner of the Asterisk then i think Asterisk - Future of Telephoy is the best book to start with the asterisk. you can download a free copy from voip-info.org. http://www.voip-info.org/wiki/view/Asterisk:+The+Future+of+Telephony Thanks Chandan On 12/16/05, sukrit [EMAIL PROTECTED] wrote: Hi Guys,Wanted some advice for the docs that you'd recommend someone new toAsterisk to read. I have a good knowledge of Unix and networking, sothat part shouldn't be a problem.Cheers,Sukrit.D. --\|||/(o o)+ooO-(_)-Ooo---+| SUKRIT D| www.liqwidkrystal.com|| Email:sukrit-at-liqwidkrystal.com||--| |MSN:[EMAIL PROTECTED] YAHOO:sd_root ||SKYPE:sukritd |+--+ ( _ ) _| | | |_(___| |___)___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Chandan Kumar MishraSoftware Engg. Induslogic,Noida ___ --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] Getting started with Asterisk and Aastra 9133i
Let me simplify my problem. I have a single Aastra 9133i SIP phone and latest Asterisk from SVN source running on Fedora Core 4. The phone currently says No Service I would like to be able to dial 1234 from the phone and get Asterisk to play back an audio message or say some digits. I can't get this to work with either SayDigits or Playback. Please help. == sip.conf == [general] port = 5060 bindaddr = 0.0.0.0 context=tutorial [3006] type=friend username=3006 secret=mypassword host=dynamic canreinvite=no permit=192.168.0.0/24 allow=all mailbox=3006 === extensions.conf === [tutorial] exten = 1234,1,Answer exten = 1234,2,SayDigits(123456789) ** TFTP directory ** = mymacaddress.cfg = sip line1 auth name: 3006 sip line1 password: mypassword sip line1 user name: 3006 sip line1 display name: myname sip line1 screen name: myname === aastra.cfg === dhcp: 1# DHCP enabled. sip silence suppression: 2 # 0 = off, 1 = on, 2 = default sip proxy port: 5060 # 5060 is set by default. sip registrar ip: 192.168.0.99# IP of registrar. --- THIS IS THE IP of my Asterisk and tftp server sip registrar port: 5060 # 5060 is set by default. sip digit time out: 6 time server disabled: 0 # Time server disabled. time server1: 192.168.0.99# Enable time server and enter at ___ --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] Getting started with Asterisk and Aastra 9133i
One more thing. I upgraded the firmware of the 9133i to 1.3. ___ --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] Getting started with Asterisk and Aastra 9133i
On Mon, 2005-12-05 at 11:15 -0500, Robert La Ferla wrote: Let me simplify my problem. I have a single Aastra 9133i SIP phone and latest Asterisk from SVN source running on Fedora Core 4. The phone currently says No Service I would like to be able to dial 1234 from the phone and get Asterisk to play back an audio message or say some digits. I can't get this to work with either SayDigits or Playback. Please help. == sip.conf == [general] port = 5060 bindaddr = 0.0.0.0 context=tutorial [3006] type=friend username=3006 secret=mypassword host=dynamic canreinvite=no permit=192.168.0.0/24 allow=all mailbox=3006 === extensions.conf === [tutorial] exten = 1234,1,Answer exten = 1234,2,SayDigits(123456789) ** TFTP directory ** = mymacaddress.cfg = sip line1 auth name: 3006 sip line1 password: mypassword sip line1 user name: 3006 sip line1 display name: myname sip line1 screen name: myname === aastra.cfg === dhcp: 1# DHCP enabled. sip silence suppression: 2 # 0 = off, 1 = on, 2 = default sip proxy port: 5060 # 5060 is set by default. sip registrar ip: 192.168.0.99# IP of registrar. --- THIS IS THE IP of my Asterisk and tftp server sip registrar port: 5060 # 5060 is set by default. sip digit time out: 6 time server disabled: 0 # Time server disabled. time server1: 192.168.0.99# Enable time server and enter at I wasted a lot of time getting 9112is to work with almost identical setup. The problem I eventually found was that the 9112is look for the config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas the documentation says they look for lower case, so they were ignoring my tftp settings. The 9133i may well be the same. The other thing I had to do was to provide the line next-server tftpserver ip; in dhcpd.conf to get them to pick everything up. (IIRC that last bit was only to do with timedate format though). Cheers Pete ___ --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] Getting started with Asterisk and Aastra 9133i
Pete Barnwell wrote: I wasted a lot of time getting 9112is to work with almost identical setup. The problem I eventually found was that the 9112is look for the config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas the documentation says they look for lower case, so they were ignoring my tftp settings. The 9133i may well be the same. The other thing I had to do was to provide the line next-server tftpserver ip; in dhcpd.conf to get them to pick everything up. (IIRC that last bit was only to do with timedate format though). I read about the mac address case sensitivity so I used an all uppercase filename which works fine. The downloading of the firmware works fine too. I also have the ntp time/date working. I just can't get Asterisk to respond to the phone! Help! ___ --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] Getting started with Asterisk and Aastra 9133i
On Mon, 2005-12-05 at 11:27 -0500, Robert La Ferla wrote: Pete Barnwell wrote: I wasted a lot of time getting 9112is to work with almost identical setup. The problem I eventually found was that the 9112is look for the config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas the documentation says they look for lower case, so they were ignoring my tftp settings. The 9133i may well be the same. The other thing I had to do was to provide the line next-server tftpserver ip; in dhcpd.conf to get them to pick everything up. (IIRC that last bit was only to do with timedate format though). I read about the mac address case sensitivity so I used an all uppercase filename which works fine. The downloading of the firmware works fine too. I also have the ntp time/date working. I just can't get Asterisk to respond to the phone! Help! One thing is to do a factory reset to reinit everything, I did that with my 9112i after upgrading the firmware. -- Dave Cotton [EMAIL PROTECTED] ___ --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] Getting started with Asterisk and Aastra 9133i
Dave Cotton wrote: One thing is to do a factory reset to reinit everything, I did that with my 9112i after upgrading the firmware. I just did that. Now Asterisk is giving me the follow error: (0.99 is my Asterisk server and 0.111 is the phone) Dec 5 12:04:10 NOTICE[14222]: chan_sip.c:10817 handle_request_register: Registration from 'No User sip:[EMAIL PROTECTED]:5060' failed for '192.168.0.111' - Username/auth name mismatch -- Registered SIP '3006' at 192.168.0.111 port 5060 expires 300 ___ --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
solved (Re: [Asterisk-Users] Getting started with Asterisk and Aastra 9133i)
I solved it by registering the phone in the sip.conf. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting started with Asterisk and Aastra 9133i
I have a Aastra 9133i phone and would like to do a simple test to make sure everything works. I already assigned an IP address to the phone (I'm able to ping it.) I have Asterisk running (installed Asterisk and Zaptel only) but not configured. I don't have a FXS/FXO card yet but I would like to test out the phone. Ideally, I'd like to be able to setup a mailbox, record a mailbox greeting, and play it back. How do I do this? I ran make samples to install the basic config files then: I added this to the sip.conf: [aastra] type=friend host=192.168.0.99 [EMAIL PROTECTED] I also added this to the extensions.conf file: Exten = 1234,1,Wait(2) Exten = 1234,2,Record(/tmp/asterisk-recording:gsm) Exten = 1234,3,Wait(2) Exten = 1234,4,Playback (/tmp/asterisk-recording) Exten = 1234,5,wait(2) Exten = 1234,6,Hangup When I dial 1234, nothing happens. I'm not sure if that's how it supposed to work or what. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting started, how to :D
Hello people, actually I'm new on this new technology (for me). Well I'll tell you what do I have got and what I like to do, so if I made the mistake to post this new thread here, just forgive me.- Let me intruduce myself and what I am working on.- Well, I'm from Argentina and, actually I'm a tinny ISP offering broadband service to office customers in 3 towns, we actually have 250 customer connected on our cloud, so what are we going to do is IP Telephony to those clients.- The main cuestion here is how to get started with Asterisk, I already read some kind of information but I don't know how to beging with this.- Could you lend me a hand?, thank you so much. Guillermo Nardoni Rosario - Argentina [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] getting started
Hi I'm just subscribed to this list because I'd like to try Asterisk I'd like some soggestions for getting started with it I tried to compile source tarball on my mandrake 10.1 but I got compilation error, I think because mdk 10.1 has a too newer gcc compiler I can use older mdk or red hat/fedora distributions, which ones are better and supported by asterisk? I have two IVR boards, a voicetronix openLine4 and a dialogic D41-JCT-PCI, I read somewhere that they are supported by asterisk, how can I use them? if someone has some useful link for starting with asterisk is welcome thanks Luca ___ 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] getting started
On Tue, 8 Mar 2005 16:55:30 +0100, Luca Bariani [EMAIL PROTECTED] wrote: Hi I'm just subscribed to this list because I'd like to try Asterisk I'd like some soggestions for getting started with it I tried to compile source tarball on my mandrake 10.1 but I got compilation error, I think because mdk 10.1 has a too newer gcc compiler I can use older mdk or red hat/fedora distributions, which ones are better and supported by asterisk? I have two IVR boards, a voicetronix openLine4 and a dialogic D41-JCT-PCI, I read somewhere that they are supported by asterisk, how can I use them? if someone has some useful link for starting with asterisk is welcome Heheee You could try installing [EMAIL PROTECTED] on a spare box. It seems to work just fine. Mike ___ 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] getting started
Luca Bariani wrote: I tried to compile source tarball on my mandrake 10.1 but I got compilation error, I think because mdk 10.1 has a too newer gcc compiler Luca, I run Mandrake 10.1 Official with current updates and compile without issue. Doug ___ 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] getting started
On Tue, 8 Mar 2005 16:55:30 +0100, Luca Bariani [EMAIL PROTECTED] wrote: I tried to compile source tarball on my mandrake 10.1 but I got compilation error, I think because mdk 10.1 has a too newer gcc compiler It compiles perfectly on MDK 10.0/10.1/Cooker, if you posted the compilation errors you could probably get some useful help. -- Dave Cotton [EMAIL PROTECTED] ___ 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] getting started
On Tue, Mar 08, 2005 at 04:55:30PM +0100, Luca Bariani wrote: Hi I'm just subscribed to this list because I'd like to try Asterisk I'd like some soggestions for getting started with it I tried to compile source tarball on my mandrake 10.1 but I got compilation error, I think because mdk 10.1 has a too newer gcc compiler Too new a compiler? Could you post the exact error? That said, if you want Asterisk up-and-running quickly, you could try Xorcom Rapid, http://www.xorcom.com/ (disclaimer: I work there). -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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] getting started
Better yet, ditch the Mandrake box and try [EMAIL PROTECTED] for you test machine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, March 08, 2005 10:00 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] getting started On Tue, Mar 08, 2005 at 04:55:30PM +0100, Luca Bariani wrote: Hi I'm just subscribed to this list because I'd like to try Asterisk I'd like some soggestions for getting started with it I tried to compile source tarball on my mandrake 10.1 but I got compilation error, I think because mdk 10.1 has a too newer gcc compiler Too new a compiler? Could you post the exact error? That said, if you want Asterisk up-and-running quickly, you could try Xorcom Rapid, http://www.xorcom.com/ (disclaimer: I work there). -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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] Getting started
I have installed asterisk - hurray! I want to configure, i.e., 800#---switch-asterisk-phone need to create members and callers. Realize I need to configure the dialplan in extensions.conf, however isn't there a checklist that helps coordinate the many extensions of this package so one can find an easy rhythem to its configuration? thanks, Chris ___ 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] Getting Started
Hello! I'm new to asterisk, too. * Bilal Ghayad [EMAIL PROTECTED] [2005-01-14 21:29]: 1) Where I can find the PC to Phone software to be used? Does it support G723 codec? If not, then the used codec will need how much bandwidth? What the client OS should be? I found something written about G723, but I haven't tested it yet. 2) Does Asterisk Contains PrePaid Billing software? Is there Calling Cards Software? I don't know about that, not even what it would be good for (private use) 3) Can Asterisk support H323? Is it stable? Yes! It is said to be stable, but I haven't tried it yet. I have only used the consoles and my old PTSN devices :-) 4) Is there SIP to H323 and H323 to SIP convertor? Yes! That's the great thing with asterisk. It should be able to connect everything together. Not only different VoIP protocols, but the PTSN and the different VoIP protocols, too. See you! Raoul ___ 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] Getting Started
Hello, I have been on the IRC Chat but will post here as well. We have 1 office currently with 4 land lines. We want to use asterisk as our call attendant with 6 lines. We were going to use vonage but it was recommended that we look over the voip providers in the wiki because other providers allow simultaneous incoming calls on the same line. It was said that other providers give you more features than vonage does. Can anyone recommend a good VOIP Provider that offers unlimited calling per line with the simultaneous incoming calls on 1 of those lines? I we be spending a couple days reading over the wiki, deciding on a piece of hardware and learning how to configure the asterisk. Thanks for your help! Randy ___ 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] Getting Started
Hi All; I am starting to know about Asterisk and trying to use it and knowing its functionalities, I hope that here I can find answers for my following questions: 1) Where I can find the PC to Phone software to be used? Does it support G723 codec? If not, then the used codec will need how much bandwidth? What the client OS should be? 2) Does Asterisk Contains PrePaid Billing software? Is there Calling Cards Software? 3) Can Asterisk support H323? Is it stable? 4) Is there SIP to H323 and H323 to SIP convertor? Regards Bilal ___ 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] Getting Started
I am starting to know about Asterisk and trying to use it and knowing its functionalities, I hope that here I can find answers for my following questions: Welcome to the community! Most of the answers are here http://voip-info.org/tiki-index.php?page=Asterisk http://asteriskdocs.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
[Asterisk-Users] Getting started with Asterisk
I am interested in learning Asterisk and have DSL (1 static IP) and a single POTS line at home. I have an Ethernet LAN running behind a Linksys router using NAT. My question is only about the hardware needed at this point. The software configuration I will read about and learn. So what hardware do I need to do the following? 1) Route incoming calls from the existing POTS line to 8 regular phones. (Work no differently than without Asterisk) 2) Sign up with a VOIP provider, get a few numbers and route each VOIP number to a subset of above phones. (e.g., One VoIP number rings 2 or 3 of the 8 phones, first phone to pick up gets the call, or even better, all 2 or 3 phones can hear the call. Is this possible? How well do fax machines work over VoIP?) 3) Outgoing calls will use VoIP provider if available, POTS line if not. 4) 911 will always use POTS line. 5) I want to use regular phones, not IP or SIP phones. I should ask first, is what I want to do possible? I've read the documents and manuals, but it is still all rather confusing. This is the hardware I think I would need in a Linux box running Asterisk, please correct if I am wrong: 1) One FXO module to connect Asterisk Linux box with POTS line 2) A FXS module for each regular phone in my house 3) Some module for connecting to the VOIP provider (Is this just the Ethernet NIC?) Can someone please provide me with the appropriate hardware product models for what I need? What is the lowest cost way to do this? If things cost too much, it may be too expensive a hobby to indulge in, at least until I win the lottery. =) And some general miscellaneous questions I have about telephony/Asterisk: 1) With the above setup, would I be able to just add SIP phones onto the Ethernet LAN in the future? 2) With just one POTS line, I wouldn't be able to have multiple POTS phone numbers, is that correct? 3) How do phone calls come over a T1 line? I thought T1 is for data. Are those strictly for VoIP calls? Is this where a TE410P is used? Thanks for your help. ___ 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] Getting started with Asterisk
What is the minimum set of configuration files needed to operate a SIP only Asterisk setup? I wrote a howto that might help: http://iheavy.com/modules.php?op=modloadname=Newsfile=articlesid=35 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting started with Asterisk
Hello , Ill just started with asterisk and I would liket to to dial between your two phones with to cisco ATA 186 , but I have a problem The two cisco ATA and the server in the same networks and i have the ring in the phone but iam not able to place a call Between the twe phone . In attachement the sip.conf and a log file Any suggestement . Regards RAbii --sip.conf--- [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) allow=all ; Allow all codecs context = bogon-calls ; Send SIP callers that we don't know about here [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=9overthruster7 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=100 ; Activate the message waiting light if this dtmfmode=rfc2833 ; voicemailbox has messages in it [2001] ; Duplicate of 2000, except with different auth data type=friend username=2001 secret=11bbanzai9 host=dynamic context=from-sip mailbox=101 dtmfmode=rfc2833 - ~ ---Log --- Asterisk*CLI We're at 10.100.18.125 port 18294 Answering/Requesting with root capability 1 Answering with non-codec capability 0x1(G723) 12 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.100.18.125:5060;branch=z9hG4bK0b1527ad From: NafthaChimie sip:[EMAIL PROTECTED];tag=as49257c3d To: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 01 Dec 2004 14:35:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 218 v=0 o=root 11153 11153 IN IP4 10.100.18.125 s=session c=IN IP4 10.100.18.125 t=0 0 m=audio 18294 RTP/AVP 4 101 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.100.18.124:5060 Asterisk*CLI Sip read: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 10.100.18.125:5060;branch=z9hG4bK0b1527ad From: NafthaChimie sip:[EMAIL PROTECTED];tag=as49257c3d To: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;tag=556017164 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: Cisco ATA 186 v3.1.0 atasip (040211A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 Warning: Media type not available 10 headers, 0 lines Transmitting: ACK sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.100.18.125:5060;branch=z9hG4bK0b1527ad From: NafthaChimie sip:[EMAIL PROTECTED];tag=as49257c3d To: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;tag=556017164 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.100.18.124:5060 Destroying call '[EMAIL PROTECTED]' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting started with Asterisk
[EMAIL PROTECTED] is believed to have said: Hello , I'll just started with asterisk and I would liket to to dial between your two phones with to cisco ATA 186 , but I have a problem The two cisco ATA and the server in the same networks and i have the ring in the phone but i'am not able to place a call Between the twe phone . In attachement the sip.conf and a log file Any suggestement . Regards RAbii Rabii, I can't help as this is the same problem I have with the very same sample configurations. Here I am using the OSX install of Asterisk (with an older release). What I saw poking around is that on the server machine at port 5060, if I try peeking in with telnet, nobody is listening. So either the two configs (sip.conf and extensions.conf) are not enough or some configuration info in some other file is messing up. What is the minimum set of configuration files needed to operate a SIP only Asterisk setup? TIA Aldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting started with Asterisk
On Wednesday 01 December 2004 17:11, Aldo Bergamini wrote: [...] I can't help as this is the same problem I have with the very same sample configurations. Here I am using the OSX install of Asterisk (with an older release). What I saw poking around is that on the server machine at port 5060, if I try peeking in with telnet, nobody is listening. So either the two configs (sip.conf and extensions.conf) are not enough or some configuration info in some other file is messing up. What is the minimum set of configuration files needed to operate a SIP only Asterisk setup? Telnet uses TCP, SIP listens on UDP, use netstat instead. B ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting started with Asterisk
[EMAIL PROTECTED] is believed to have said: Telnet uses TCP, SIP listens on UDP, use netstat instead. B Bob, thanks for the hint! I should have imagined that SIP could not use a tcp protocol... Aldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting started
Hi all Problem with gnophone: I can not make a call. (just hangs) Im am a novice to Asterisk but quite experienced Linux user. I am having some problems with the gnophone. I have tried to isert my user/password but nothing have changed. I have tested the michrophone and it is working. The sound also works. This is my configuration from preference-telephone Use Asterics. Server: iaxtel.com port 10004 (it is a NAT assigned tcp port on the router directed to my PC) Context: iaxtel prefix :empty username : my username provided by mail. password : my password peer(optional): my username provided by mail. secret : my password The user/pass-word is the same used to enter this mailing list. What I would like to know is how to fix my problems. Second I would like to know where there is some info for setting up a system. (I have tried to get these but only been able to find buy a book. Before buying a book I would like to get it work first.) With regards Anders Gnistrup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting started
I'm interested in getting started with * and have some questions that I could sure use some help. Appreciate any and all inputs 1) Regarding the softphone client - what are the differences between an SIP client such as available from www.xten.com or an IAX client (there are several available)? I want to be able to run from Windows 2000 / XP platforms and also from PDA's (PocketPC). Also, could you recommend softphone client where I can get access to source code for modifications,,,as I want to integrate the softphone into my web services client 2) I'm looking at a Digium Wildcard TE405P - with a T-1 PRI - anyone with experience here and are there any quirks or other issues i should consider? 3) regarding * database - what is the voicemail database (outgoing messages and incoming) and can I ODBC to my separate data servers using SQL 92 standards? 4) For a mid-size installation - regarding hardware configuration - i'm thinking RedHat 9, but unsure about the HDD config . Will i get good performance with IDE mirrored drives versus SCSI RAID?? again, thx in advance for your inputs Kaydon Stanzione [EMAIL PROTECTED]
[Asterisk-Users] Getting Started
What is the best way in getting started evaluating Asterisk? Are there recommendations on the types of card I should be using for an initial eval? Thanks Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting Started
download the code, complile the code, start bashing on configs. :) if you want to glue to the PSTN, i'd recommend getting a FXO (WC-X100P) and FXS (TDM-10B) card and some cheap SIP / VoIP phones (grandstream or snom) john brown, ceo chagres technologies, inc Providers of VoIP hardware http://www.chagres.net/products/voip/ On Sat, Nov 08, 2003 at 05:23:57PM -0500, Peter A. Solomon wrote: What is the best way in getting started evaluating Asterisk? Are there recommendations on the types of card I should be using for an initial eval? Thanks Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting started
Hi, I am a total newbie to asterisk and can't find any useful documentation for asterisk...how are people supposed to get started? I'd like to know, how I create User Accounts, so that a SIP UA can login into asterisk with a password, for example. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting started
Read the documentation, read the sip.conf file. And if it still doesn't make sense try one more time through the documentation and config file. At that point you should at least know enough to ask pointed questions at specific problems in your configs. On Mon, 2003-07-14 at 11:19, Johannes Herlitz wrote: Hi, I am a total newbie to asterisk and can't find any useful documentation for asterisk...how are people supposed to get started? I'd like to know, how I create User Accounts, so that a SIP UA can login into asterisk with a password, for example. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting Started with Digium T100/E100 cards
Just purchased a couple of T100 and E100 cards in order to interface from our company's proprietary system through a linux gateway. I am new to Asterisk and Digium. After installing the T100 card, I went looking for drivers for this card. Are the drivers built into the Asterisk application? If so, how do I enable the card through Asterisk? Are there any good getting started documents for a newbie like myself? I have downloaded and read the Asterisk handbook, while helpful, it doesn't answer all my questions. (of course I could just be daft!) Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting Started with Digium T100/E100 cards
To enable any of TxxxP or ExxxP card you must compile the zaptel package from cvs. Then to enable the T100P or E100P you should load the wct1xxp module. Then you can use the zttool (/sbin/zttool) to check the status of the cards and of the spans you've configured in /etc/zaptel.conf -- Stefano ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
