Re: [asterisk-users] newbie questions
On Sat, 20 Jun 2009, C. Savinovich wrote: Let me see if I get you: you inserted the installation CD, then you restarted the computer, and now you want to know what to do next? How about: 1) Turn off the computer. 2) Read the installation guide for the CD. 3) Install the software. 4) Read ATOF to get a clue to the scope of what Asterisk can do. 5) Get frustrated trying to do really cool things within the GUI. 6) Format the drive. 7) Install CentOS. 8) Install Asterisk from source. 9) Learn to configure the configuration files by hand. But then, I gladly admit to being a command line weenie. 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
[asterisk-users] newbie questions
I have an Asterisknow.org CD. When I boot up, it seems ready for me to choose update, console, etc. I'm assuming I need to do something at the CLI prompt. Is there a tutorial that would take me from loading CD to making first test call? Computer is Dell Optiplex GX260 50GB free disk space 1.5GB RAM P4 processor external mic speakers Skype is on board, and would be good to use it, if possible. If I want to use Skype, do I need anything additional? Would it be better to install CD on my hard drive? Any help appreciated. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie questions: seting up extension for miSDN
Hello! Sorry, I'm sure it's stupid. but I've got a simple ISDN line and a simple ISDN-card, now finally running. :-) I'm using application Jack and asterisk (CLI) only to do my bidding. Now I can make calls. But how ca I setup my extensions.conf to receive a call? I've had an example like tis: [default] exten = 500,1,Answer() exten = 500,n,Jack() But it seems I can't do anything with me. Besides, the PortAudio (at least I assume) gives me some errors. Here's an example: CLI console dial [EMAIL PROTECTED] [Sep 8 19:02:40] --- () --- Call from Console has been Answered --- () --- [Sep 8 19:02:40] WARNING[7399]: chan_console.c:367 start_stream: Failed to open stream - (-9997) Invalid sample rate CLI P[ 1] MGMT: SSTATUS: L2_RELEASED Now if I can do it with application Jack, please anyone give me a pointer to the right doc or an example. But for the docs: Please give me an example for an extension that is my simple telephone. I have nothing fancy over here and I get confused. Sorry... Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Questions . . .
I wanted to say thanks to those who responded to my query. You all gave me some good ideas to explore that I had not considered before, which is what I was hoping for. :^) On Tuesday 14 November 2006 10:50, Henry.L.Coleman wrote: By the time you purchase PCI cards for you extensions (FSO ports)you would be better off purchasing SIP phones like Grandstream GXP 2000 this will give you a fully featured PBX IP phone for about the same cost or less than FSO ports. Asterisk will have no problem running 25 or more SIP phones Personally I would reduce the incoming analog lines to 4 (FXO) ports and add some DID lines. This way you will only have to buy one PCI board with 4 FXO ports Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada -- Jason Flatt Father of Six: http://www.flattfamily.com/ (Joseph, 13; Cramer, 11; Travis, 9; Angela; Harry, 5; and William, 12:04 am, 12-29-2005) Linux User: http://www.sourcemage.org/ Drupal Fanatic: http://drupal.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 Questions . . .
By the time you purchase PCI cards for you extensions (FSO ports)you would be better off purchasing SIP phones like Grandstream GXP 2000 this will give you a fully featured PBX IP phone for about the same cost or less than FSO ports. Asterisk will have no problem running 25 or more SIP phones Personally I would reduce the incoming analog lines to 4 (FXO) ports and add some DID lines. This way you will only have to buy one PCI board with 4 FXO ports Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Maybe you should try this http://www.digium.com/en/products/hardware/aadk.php . Is very heavy loaded if 9PCI cards at a server. But is possible but not encourge. Maybe you can consider to have digital extension with IP phone. THis is my opinion. :-) good luck On 11/14/06, Jason Flatt [EMAIL PROTECTED] wrote: Hello all. My company currently has an older Executone PBX system that we are outgrowing. Rather than wait until the last minute to make a hasty decision, I thought it would be a good idea to do some research and compare options first. My expertise is in computers and networking, and telephony systems are mostly foreign to me. What we currently have are 5 incoming POTS lines and 25 stations and are wanting to add 1 or 2 more stations. I think we might have added at least one more incoming line, except that the phones we have only support 5 lines (so I'm told). Our PBX system has room for 5 more stations, then it's time to buy a new one. I'm assuming I need to add some hardware in order to make Asterisk work with our existing setup, but I'm not entirely sure what. Based on the reading I've done so far and my limited understanding, if we wanted to use it in place of our existing PBX system, I would need to get an analog interface card (several, actually), like Digium's TDM400P, like so: 2 - Wildcard TDM04B cards for FXO and 7 - Wildcard TDM40B cards for FXS -or- 1 - Wildcard TDM04B card for FXO and 1 - Wildcard TDM22B card for FXO FXS and 7 - Wildcard TDM40B cards for FXS I might as well use the top configuration for future expansion. If I am correct, that is 9 PCI cards in a PC. I don't know of any motherboard that supports that many cards, so either I'm wrong, or I'll need different cards, or I'll need to utilize 2 or more PCs in conjunction with each other. I haven't yet found any mention on the last two options, so I'm assuming I'm wrong and I need a little enlightenment. Thank you for any information that will help me better understand this. -- Jason Flatt Father of Six: http://www.flattfamily.com/ (Joseph, 13; Cramer, 11; Travis, 9; Angela; Harry, 5; and William, 12:04 am, 12-29-2005) Linux User: http://www.sourcemage.org/ Drupal Fanatic: http://drupal.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 -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --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
[asterisk-users] Newbie Questions . . .
Hello all. My company currently has an older Executone PBX system that we are outgrowing. Rather than wait until the last minute to make a hasty decision, I thought it would be a good idea to do some research and compare options first. My expertise is in computers and networking, and telephony systems are mostly foreign to me. What we currently have are 5 incoming POTS lines and 25 stations and are wanting to add 1 or 2 more stations. I think we might have added at least one more incoming line, except that the phones we have only support 5 lines (so I'm told). Our PBX system has room for 5 more stations, then it's time to buy a new one. I'm assuming I need to add some hardware in order to make Asterisk work with our existing setup, but I'm not entirely sure what. Based on the reading I've done so far and my limited understanding, if we wanted to use it in place of our existing PBX system, I would need to get an analog interface card (several, actually), like Digium's TDM400P, like so: 2 - Wildcard TDM04B cards for FXO and 7 - Wildcard TDM40B cards for FXS -or- 1 - Wildcard TDM04B card for FXO and 1 - Wildcard TDM22B card for FXO FXS and 7 - Wildcard TDM40B cards for FXS I might as well use the top configuration for future expansion. If I am correct, that is 9 PCI cards in a PC. I don't know of any motherboard that supports that many cards, so either I'm wrong, or I'll need different cards, or I'll need to utilize 2 or more PCs in conjunction with each other. I haven't yet found any mention on the last two options, so I'm assuming I'm wrong and I need a little enlightenment. Thank you for any information that will help me better understand this. -- Jason Flatt Father of Six: http://www.flattfamily.com/ (Joseph, 13; Cramer, 11; Travis, 9; Angela; Harry, 5; and William, 12:04 am, 12-29-2005) Linux User: http://www.sourcemage.org/ Drupal Fanatic: http://drupal.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 Questions . . .
Maybe you should try this http://www.digium.com/en/products/hardware/aadk.php . Is very heavy loaded if 9PCI cards at a server. But is possible but not encourge. Maybe you can consider to have digital extension with IP phone. THis is my opinion. :-) good luck On 11/14/06, Jason Flatt [EMAIL PROTECTED] wrote: Hello all.My company currently has an older Executone PBX system that we are outgrowing.Rather than wait until the last minute to make a hasty decision, I thought itwould be a good idea to do some research and compare options first.My expertise is in computers and networking, and telephony systems are mostlyforeign to me.What we currently have are 5 incoming POTS lines and 25 stations and arewanting to add 1 or 2 more stations.I think we might have added at least one more incoming line, except that the phones we have only support 5 lines(so I'm told).Our PBX system has room for 5 more stations, then it's timeto buy a new one.I'm assuming I need to add some hardware in order to make Asterisk work with our existing setup, but I'm not entirely sure what.Based on the readingI've done so far and my limited understanding, if we wanted to use it inplace of our existing PBX system, I would need to get an analog interface card (several, actually), like Digium's TDM400P, like so:2 - Wildcard TDM04B cards for FXO and7 - Wildcard TDM40B cards for FXS-or-1 - Wildcard TDM04B card for FXO and1 - Wildcard TDM22B card for FXO FXS and 7 - Wildcard TDM40B cards for FXSI might as well use the top configuration for future expansion.If I am correct, that is 9 PCI cards in a PC.I don't know of any motherboardthat supports that many cards, so either I'm wrong, or I'll need different cards, or I'll need to utilize 2 or more PCs in conjunction with each other.I haven't yet found any mention on the last two options, so I'm assuming I'mwrong and I need a little enlightenment.Thank you for any information that will help me better understand this. --Jason FlattFather of Six:http://www.flattfamily.com/ (Joseph, 13; Cramer, 11; Travis,9; Angela; Harry, 5; and William, 12:04 am, 12-29-2005)Linux User: http://www.sourcemage.org/Drupal Fanatic: http://drupal.org/___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --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 Questions . . .
Jason,If you must stick with analog phones, you can find higher density channel banks that will host 8, 16 or up to 24 ports each. They communicate back to your asterisk server via your LAN. Or, as has been stated, you can purchase IP phones that also communicate back to your asterisk server via your LAN. This will leave you dealing only with the FXO ports. If you look at Sangoma gear, you can probably achieve what you're looking for and only occupy 1 PCI slot (even though the card needs the space of two PCI boards). On 11/13/06, Sharon Lim [EMAIL PROTECTED] wrote: Maybe you should try this http://www.digium.com/en/products/hardware/aadk.php . Is very heavy loaded if 9PCI cards at a server. But is possible but not encourge. Maybe you can consider to have digital extension with IP phone. THis is my opinion. :-) good luck On 11/14/06, Jason Flatt [EMAIL PROTECTED] wrote: Hello all.My company currently has an older Executone PBX system that we are outgrowing.Rather than wait until the last minute to make a hasty decision, I thought itwould be a good idea to do some research and compare options first.My expertise is in computers and networking, and telephony systems are mostlyforeign to me.What we currently have are 5 incoming POTS lines and 25 stations and arewanting to add 1 or 2 more stations.I think we might have added at least one more incoming line, except that the phones we have only support 5 lines(so I'm told).Our PBX system has room for 5 more stations, then it's timeto buy a new one.I'm assuming I need to add some hardware in order to make Asterisk work with our existing setup, but I'm not entirely sure what.Based on the readingI've done so far and my limited understanding, if we wanted to use it inplace of our existing PBX system, I would need to get an analog interface card (several, actually), like Digium's TDM400P, like so:2 - Wildcard TDM04B cards for FXO and7 - Wildcard TDM40B cards for FXS-or-1 - Wildcard TDM04B card for FXO and1 - Wildcard TDM22B card for FXO FXS and 7 - Wildcard TDM40B cards for FXSI might as well use the top configuration for future expansion.If I am correct, that is 9 PCI cards in a PC.I don't know of any motherboardthat supports that many cards, so either I'm wrong, or I'll need different cards, or I'll need to utilize 2 or more PCs in conjunction with each other.I haven't yet found any mention on the last two options, so I'm assuming I'mwrong and I need a little enlightenment.Thank you for any information that will help me better understand this. --Jason FlattFather of Six:http://www.flattfamily.com/ (Joseph, 13; Cramer, 11; Travis, 9; Angela; Harry, 5; and William, 12:04 am, 12-29-2005)Linux User: http://www.sourcemage.org/Drupal Fanatic: http://drupal.org/___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___--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 questions about Voice mail
Brian, I should concur with all that Dean raised. Given the experience level you describe and the clear business case for what you want to do, had you considered a commerical solution ? It would give you the peace of mind that all will work. It will also allow you to do many of the smaller features such as Outlook Integration in a click and drop manner as well as the group issues, setting up of voicemail delivery to email etc. See some other comments below. Steve (of course would be more than happy to promote our own but there are others you could do well to look at) - Original Message - From: [EMAIL PROTECTED] To: Dean Collins [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 05, 2006 9:01 PM Subject: RE: [asterisk-users] Newbie questions about Voice mail Dean Thanks for responding. I have added more info in your reply. Right now we do not operate our own PBX or voice mail system. All of the service is provided by the telco. As a start I was wondering if I could simply put in asterisk to do just voicemail. I am assuming the telco can configure all the phone to automatically call forward to asterisk on no answer. If asterisk can handle this I am assuming that a user would just call some number to retiev voice mail. They would lose the call waiting light on their phone so the email notification of a voice mail would be necessary. ..Brian On Sun, 5 Nov 2006, Dean Collins wrote: Date: Sun, 5 Nov 2006 00:04:36 -0500 From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Newbie questions about Voice mail Hi Brian, I'm sure some other people will give you better answers but quick answers are; 1/ Depends on volume of message leaving/collection, is it in a single location? Multiple locations with multiple time zones? Two locations, one time zone. Could be two different systems since they are in two different cites connected by a 1G connection. Estimate the number of voicemails left per hour and reply with this. There are about 3000 phones. Some are busier than others os lets say 2 messages per phone per day. An they are mostly in the peak work day so lets say 500 per hour and the average length is 30 seconds. This is less than 5 concurrent messages :) I think you will need to have at least a T1 system because you are going to face some fairly extreme variations in usage. 2/ retrieve either via deliver to email or dial in to a number to collect voicemail via phone (or collect and play via a website) What does the conversion and how does one handle bulk updates? to users? How much control does the user have? How is a user informed that voicemail are waiting for them ? What is you existing PBX, how would the Asterisk based system interface with it ? does it use SIP ? or T1 interface ? How are the retrieving their voicemail now? Do you want to replicate this for ease of replacement as near as possible? Right now we are using the voice mail service provided by the teclo and are spending $0.06 per minute. The user connects to the voice mail by dialing *99 and entering a password on their office set or remoetely by dialing 123-MAIL on any phone (123 is the three digit prefix of their phone number) and then entering their password. They do not have any voice to email service today. If possible I would like to ease the transition if it can be done. Lots of stepswill follow discovery if it can be done. 3/ Not sure what you mean by tie in? How do you match a voice mail box to an email address? Can there be multiple email addresses for one voice mail box? You can program the Asterisk but with a good interface, click and drop. 4/ Sure, how do you have this configured at the moment? Why not replicate voicemail group delivery in the same format? Talkmail is a service provided by the telco where you group a bunch of numbers together so you can send the same message to all of them at the same time. Again it can be programmed but click and drop may be easier. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, 4 November 2006 11:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie questions about Voice mail I am totally ignorant about actually using asterisk for any purpose. I have read some of the docs but not all. I am currently doing a telephone audit for my company and one of the issues is voice mail. We are spending quit a bit of money with our telco for voice mail services and I was wondering about using asterisk as just a voice mail system. We are not quite ready to move to a full VOIP system yet but if I can get this system in place the VOIP will follow. Could I get
RE: [asterisk-users] Newbie questions about Voice mail
Dean Thanks for responding. I have added more info in your reply. Right now we do not operate our own PBX or voice mail system. All of the service is provided by the telco. As a start I was wondering if I could simply put in asterisk to do just voicemail. I am assuming the telco can configure all the phone to automatically call forward to asterisk on no answer. If asterisk can handle this I am assuming that a user would just call some number to retiev voice mail. They would lose the call waiting light on their phone so the email notification of a voice mail would be necessary. ..Brian On Sun, 5 Nov 2006, Dean Collins wrote: Date: Sun, 5 Nov 2006 00:04:36 -0500 From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Newbie questions about Voice mail Hi Brian, I'm sure some other people will give you better answers but quick answers are; 1/ Depends on volume of message leaving/collection, is it in a single location? Multiple locations with multiple time zones? Two locations, one time zone. Could be two different systems since they are in two different cites connected by a 1G connection. Estimate the number of voicemails left per hour and reply with this. There are about 3000 phones. Some are busier than others os lets say 2 messages per phone per day. An they are mostly in the peak work day so lets say 500 per hour and the average length is 30 seconds. 2/ retrieve either via deliver to email or dial in to a number to collect voicemail via phone (or collect and play via a website) What does the conversion and how does one handle bulk updates? to users? How much control does the user have? How are the retrieving their voicemail now? Do you want to replicate this for ease of replacement as near as possible? Right now we are using the voice mail service provided by the teclo and are spending $0.06 per minute. The user connects to the voice mail by dialing *99 and entering a password on their office set or remoetely by dialing 123-MAIL on any phone (123 is the three digit prefix of their phone number) and then entering their password. They do not have any voice to email service today. If possible I would like to ease the transition if it can be done. Lots of stepswill follow discovery if it can be done. 3/ Not sure what you mean by tie in? How do you match a voice mail box to an email address? Can there be multiple email addresses for one voice mail box? 4/ Sure, how do you have this configured at the moment? Why not replicate voicemail group delivery in the same format? Talkmail is a service provided by the telco where you group a bunch of numbers together so you can send the same message to all of them at the same time. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, 4 November 2006 11:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie questions about Voice mail I am totally ignorant about actually using asterisk for any purpose. I have read some of the docs but not all. I am currently doing a telephone audit for my company and one of the issues is voice mail. We are spending quit a bit of money with our telco for voice mail services and I was wondering about using asterisk as just a voice mail system. We are not quite ready to move to a full VOIP system yet but if I can get this system in place the VOIP will follow. Could I get all 3000 phones (on 2 sites) or a large subset set to have a call forward no-answer feature set to call a number that would be answered by asterisk's voice mail. If so: 1. what hardware do I need to handle 3000 phones? 2. how would users retrieve their voice mail? 3. how does one tie voice mail into an e-mail address? Are their ways to do bulk updates for several thousand new users every year? 4. is there a feature what we call talk mail where you set up a group of phone numbers and send the same message to all of them? Any help would be greatly appreciated. .TIA Brian Kaye ...UNB ___ --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 questions about Voice mail
Hi Brian, Uhmmm as it appears you are using a centrex service from your telco (your comment about not having any pabx) I need to ask this question..are you sure that under your current commercial arrangements you are actually allowed to continue to use the telco as your centrex provider but not use them for your voicemail? Also if you decided to use a separate asterisk server for your voicemail service how would calls be transferred to this number? Would the carrier allow you to host and asterisk service off some of your existing centrex extensions? Would this incur a cost or similar. I think for your bosses 'discovery' report the answer would be Yes to can asterisk be used as just a voicemail server Yes to people can operate with the same methods of retrival they currently do Yes to people can also retrieve via additional methods such as web or email And finally yes this will save us money in the longer term at 6c per minute currently. The next step should be 1a/ You boss decides You or someone in your team skill up in asterisk Or Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, 5 November 2006 3:02 PM To: Dean Collins Cc: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Newbie questions about Voice mail Dean Thanks for responding. I have added more info in your reply. Right now we do not operate our own PBX or voice mail system. All of the service is provided by the telco. As a start I was wondering if I could simply put in asterisk to do just voicemail. I am assuming the telco can configure all the phone to automatically call forward to asterisk on no answer. If asterisk can handle this I am assuming that a user would just call some number to retiev voice mail. They would lose the call waiting light on their phone so the email notification of a voice mail would be necessary. ..Brian On Sun, 5 Nov 2006, Dean Collins wrote: Date: Sun, 5 Nov 2006 00:04:36 -0500 From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Newbie questions about Voice mail Hi Brian, I'm sure some other people will give you better answers but quick answers are; 1/ Depends on volume of message leaving/collection, is it in a single location? Multiple locations with multiple time zones? Two locations, one time zone. Could be two different systems since they are in two different cites connected by a 1G connection. Estimate the number of voicemails left per hour and reply with this. There are about 3000 phones. Some are busier than others os lets say 2 messages per phone per day. An they are mostly in the peak work day so lets say 500 per hour and the average length is 30 seconds. 2/ retrieve either via deliver to email or dial in to a number to collect voicemail via phone (or collect and play via a website) What does the conversion and how does one handle bulk updates? to users? How much control does the user have? How are the retrieving their voicemail now? Do you want to replicate this for ease of replacement as near as possible? Right now we are using the voice mail service provided by the teclo and are spending $0.06 per minute. The user connects to the voice mail by dialing *99 and entering a password on their office set or remoetely by dialing 123-MAIL on any phone (123 is the three digit prefix of their phone number) and then entering their password. They do not have any voice to email service today. If possible I would like to ease the transition if it can be done. Lots of stepswill follow discovery if it can be done. 3/ Not sure what you mean by tie in? How do you match a voice mail box to an email address? Can there be multiple email addresses for one voice mail box? 4/ Sure, how do you have this configured at the moment? Why not replicate voicemail group delivery in the same format? Talkmail is a service provided by the telco where you group a bunch of numbers together so you can send the same message to all of them at the same time. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, 4 November 2006 11:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie questions about Voice mail I am totally ignorant about actually using asterisk for any purpose. I have read some of the docs but not all. I am currently doing a telephone audit for my company and one of the issues is voice mail. We are spending quit a bit of money with our telco for voice mail services and I was wondering about using asterisk as just
RE: [asterisk-users] Newbie questions about Voice mail
On Sun, 5 Nov 2006, Dean Collins wrote: Date: Sun, 5 Nov 2006 15:21:19 -0500 From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Newbie questions about Voice mail Hi Brian, Uhmmm as it appears you are using a centrex service from your telco (your comment about not having any pabx) yes. I need to ask this question..are you sure that under your current commercial arrangements you are actually allowed to continue to use the telco as your centrex provider but not use them for your voicemail? Voice mail is a separately billed service that some lines have and some don't. We pay $0.06 per minute to use it. Its cash cow for the telco and a big bill for us. Also if you decided to use a separate asterisk server for your voicemail service how would calls be transferred to this number? I am assuming there is a feature to transfer a call when the phoen does not ring after a certain number of rings. But I don't thing I know to handle getting voice mail if the line is busy. Would the carrier allow you to host and asterisk service off some of your existing centrex extensions? Would this incur a cost or similar. I am sure there would be a cost whatever we had them do. I was hoping to go a little further if possible to install a server and a t1 circuit with enough capacity to handle the load. I think for your bosses 'discovery' report the answer would be Yes to can asterisk be used as just a voicemail server Yes to people can operate with the same methods of retrieval they currently do Yes to people can also retrieve via additional methods such as web or email And finally yes this will save us money in the longer term at 6c per minute currently. The next step should be 1a/ You boss decides You or someone in your team skill up in asterisk Or Does the asterisk communitty have a presence at any of the IP telephony conference? ..Brian Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, 5 November 2006 3:02 PM To: Dean Collins Cc: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Newbie questions about Voice mail Dean Thanks for responding. I have added more info in your reply. Right now we do not operate our own PBX or voice mail system. All of the service is provided by the telco. As a start I was wondering if I could simply put in asterisk to do just voicemail. I am assuming the telco can configure all the phone to automatically call forward to asterisk on no answer. If asterisk can handle this I am assuming that a user would just call some number to retiev voice mail. They would lose the call waiting light on their phone so the email notification of a voice mail would be necessary. ..Brian On Sun, 5 Nov 2006, Dean Collins wrote: Date: Sun, 5 Nov 2006 00:04:36 -0500 From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Newbie questions about Voice mail Hi Brian, I'm sure some other people will give you better answers but quick answers are; 1/ Depends on volume of message leaving/collection, is it in a single location? Multiple locations with multiple time zones? Two locations, one time zone. Could be two different systems since they are in two different cites connected by a 1G connection. Estimate the number of voicemails left per hour and reply with this. There are about 3000 phones. Some are busier than others os lets say 2 messages per phone per day. An they are mostly in the peak work day so lets say 500 per hour and the average length is 30 seconds. 2/ retrieve either via deliver to email or dial in to a number to collect voicemail via phone (or collect and play via a website) What does the conversion and how does one handle bulk updates? to users? How much control does the user have? How are the retrieving their voicemail now? Do you want to replicate this for ease of replacement as near as possible? Right now we are using the voice mail service provided by the teclo and are spending $0.06 per minute. The user connects to the voice mail by dialing *99 and entering a password on their office set or remoetely by dialing 123-MAIL on any phone (123 is the three digit prefix of their phone number) and then entering their password. They do not have any voice to email service today. If possible I would like to ease the transition if it can be done. Lots of stepswill follow discovery if it can be done. 3/ Not sure what you mean by tie in? How do you match a voice mail box to an email address? Can there be multiple email addresses for one voice mail box? 4/ Sure, how do you have
RE: [asterisk-users] Newbie questions about Voice mail
The next step should be 1a/ You boss decides You or someone in your team skill up in asterisk Or Does the asterisk communitty have a presence at any of the IP telephony conference? ..Brian You just missed it check out www.astricon.net it was 2 weeks ago in Dallas. (but yes Digium were at VON and other events this year as well). Cheers, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie questions about Voice mail
I am totally ignorant about actually using asterisk for any purpose. I have read some of the docs but not all. I am currently doing a telephone audit for my company and one of the issues is voice mail. We are spending quit a bit of money with our telco for voice mail services and I was wondering about using asterisk as just a voice mail system. We are not quite ready to move to a full VOIP system yet but if I can get this system in place the VOIP will follow. Could I get all 3000 phones (on 2 sites) or a large subset set to have a call forward no-answer feature set to call a number that would be answered by asterisk's voice mail. If so: 1. what hardware do I need to handle 3000 phones? 2. how would users retrieve their voice mail? 3. how does one tie voice mail into an e-mail address? Are their ways to do bulk updates for several thousand new users every year? 4. is there a feature what we call talk mail where you set up a group of phone numbers and send the same message to all of them? Any help would be greatly appreciated. .TIA Brian Kaye ...UNB ___ --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 questions about Voice mail
Hi Brian, I'm sure some other people will give you better answers but quick answers are; 1/ Depends on volume of message leaving/collection, is it in a single location? Multiple locations with multiple time zones? Estimate the number of voicemails left per hour and reply with this. 2/ retrieve either via deliver to email or dial in to a number to collect voicemail via phone (or collect and play via a website) How are the retrieving their voicemail now? Do you want to replicate this for ease of replacement as near as possible? 3/ Not sure what you mean by tie in? 4/ Sure, how do you have this configured at the moment? Why not replicate voicemail group delivery in the same format? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, 4 November 2006 11:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie questions about Voice mail I am totally ignorant about actually using asterisk for any purpose. I have read some of the docs but not all. I am currently doing a telephone audit for my company and one of the issues is voice mail. We are spending quit a bit of money with our telco for voice mail services and I was wondering about using asterisk as just a voice mail system. We are not quite ready to move to a full VOIP system yet but if I can get this system in place the VOIP will follow. Could I get all 3000 phones (on 2 sites) or a large subset set to have a call forward no-answer feature set to call a number that would be answered by asterisk's voice mail. If so: 1. what hardware do I need to handle 3000 phones? 2. how would users retrieve their voice mail? 3. how does one tie voice mail into an e-mail address? Are their ways to do bulk updates for several thousand new users every year? 4. is there a feature what we call talk mail where you set up a group of phone numbers and send the same message to all of them? Any help would be greatly appreciated. .TIA Brian Kaye ...UNB ___ --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
[asterisk-users] Newbie Questions
I've been doing a lot of reading over the last few weeks on Asterisk, and will be implementing a test system this week to play with. I've got two questions in regards to the ideal implementation for our company. First, has anyone written any drivers to interface with proprietary phones? Specifically we have a comdial system and if we could use our existing 35 phones instead of having to buy all new there'd be huge savings there. I can't find anywhere that anyone has written any type of interface for proprietary (no reverse hacks or anything anywhere from what I can find), so I figure this is a no. Now for the more complicated question, that I have my doubts on the ability to perform. Would it be possible to throw an Asterisk PBX system between our Comdial system the Internet, and then throw another Asterisk PBX system at a remote location with Comdial phones to tie in to our system that way? I'm imagining using a TDM400 or the likes, connecting to the Comdial via FXO and connecting the to Asterisk PBX's via FXS. Rereading on the FXO FXS I think I'm misunderstanding how FXS works and this won't work at all. Any suggestions for what I'd like to do aside from scrap everything and start over with IP phones? ___ --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 Questions
- Original Message - From: Ken Williams [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 01, 2006 2:10 AM Subject: [asterisk-users] Newbie Questions I've been doing a lot of reading over the last few weeks on Asterisk, and will be implementing a test system this week to play with. I've got two questions in regards to the ideal implementation for our company. First, has anyone written any drivers to interface with proprietary phones? Specifically we have a comdial system and if we could use our existing 35 phones instead of having to buy all new there'd be huge savings there. I can't find anywhere that anyone has written any type of interface for proprietary (no reverse hacks or anything anywhere from what I can find), so I figure this is a no. If they are SIP phones and they support SIP then most likely yes. If they are POTS phones then you can use them with a voice card or a channel bank. If they are proprietary phones from a different PBX then most likely not. To cut down costs you may want to look at selling your current systems and your phones on eBay. Now for the more complicated question, that I have my doubts on the ability to perform. Would it be possible to throw an Asterisk PBX system between our Comdial system the Internet, and then throw another Asterisk PBX system at a remote location with Comdial phones to tie in to our system that way? I'm imagining using a TDM400 or the likes, connecting to the Comdial via FXO and connecting the to Asterisk PBX's via FXS. I have never used this system so I cant comment on it. However if you can connect to it with POTS lines it shouldnt be too hard. Also if the system can handle a T1 card you may want to connect it to Asterisk that way. Rereading on the FXO FXS I think I'm misunderstanding how FXS works and this won't work at all. Basicluy an FXO port connects to a phone line (i.e. the line coming in from the telco) and the FXS connects to a device (such as a POTS phone or fax machine). Any suggestions for what I'd like to do aside from scrap everything and start over with IP phones? ___ --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 Questions
You can put the Asterisk system in front (i.e., betweenthe PSTNand your Comdial system). This will let Asterisk choose whether the call should go out over the PSTN or the Internet using VoIP. You would use the same for the second location, provided that is a complete Comdial system. You could not, however, just put Comdial phones over there and expect it to work. You also would not be on the same phone system. But, if you are looking at tying two offices together using VoIP (and not paying long distance), then yes, this would work. With the right dial plan, you could possibly dial direct if the Comdial has an autoattendant. In this case, Asterisk would dial into the remote Comdial, wait, then dial the extension number and complete the call. On the local COmdial, you would most problably have to dial a 9 to get to the Asterisk system. I imagine, you may be able to use speeddials for the remote extensions which would automatically dial the 9. The possibilities are endless. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie Questions - Any help appreciated
Sorry for the long email but I am having all sorts of
probs
I basically have a number od sip phones in the house
I have 3 incoming numbers (sipgate) and one outbound service (sipdiscount)
I want all extensions to be able to call out using the outbound lines (one
at a time obviousley) and I want various extensions to ring depending on
which inbound number is called.
Problems
1) When I boot Asterisk it no longer connects to sipgate to register the
inbound lines, it did earlier on today but isn't anymore, does it look like
I did something with my config?
2) When I select the extension and try and dial out, I immediately get the
engaged tone on the phone. It hasn't had time to dial out so I know its at
the asterisk end.
3) When I dial from ext to ext the voicemail doesn't work.
Ho hum...
Here are my sip and extensions conf. Any help appreciated
__
extensions.conf
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
;
; The General category is for certain variables.
;
[general]
static=yes
writeprotect=no
[globals]
PHONES1=SIP/220
PHONES1VM=220
PHONES2=SIP/221
PHONES2VM=221
PHONES3=SIP/222
PHONES3VM=222
PHONES4=SIP/223
PHONES4VM=223
PHONES5=SIP/224
PHONES5VM=224
PHONES5=SIP/225
PHONES5VM=225
[sipdiscount-outbound]
exten = 220,1,Dial([EMAIL PROTECTED])
exten = 221,1,Dial([EMAIL PROTECTED])
exten = 222,1,Dial([EMAIL PROTECTED])
exten = 223,1,Dial([EMAIL PROTECTED])
exten = 224,1,Dial([EMAIL PROTECTED])
exten = 225,1,Dial([EMAIL PROTECTED])
[sipgate-inbound]
exten = 3858313,1,Dial(SIP/220SIP/221SIP/223)
exten = 3858294,1,Dial(SIP/220)
exten = 3858817,1,Dial(SIP/221SIP/220))
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat = 9
include = default
include = parkedcalls
include = trunklocal
; include = iaxtel700
include = trunktollfree
include = iaxprovider
include = sipdiscount-outbound
;This will create a macro we will use in the dialling plan
[macro-vmessage]
exten = s,1,VoiceMail2(u${ARG1})
exten = s,2,Playback(groovy)
exten = s,3,Playback(goodbye)
exten = s,4,Hangup
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1); Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy
announce
exten = s-BUSY,2,Goto(default,s,1); If they press #, return to start
exten = _s-.,1,Goto(s-NOANSWER,1); Treat anything else as no answer
exten = a,1,VoicemailMain(${ARG1}); If they press *, send the user into
VoicemailMain
; --
; DEFINE EXTENSIONS
; --
[home]
exten = 220,1,Dial(${PHONES1},20,Ttm)
exten = 220,2,Macro(vmessage,${PHONES1VM})
exten = 220,3,Hangup
; Line 2
exten = 221,1,Dial(${PHONES2},20,Ttm)
exten = 221,2,Macro(vmessage,${PHONES2VM})
exten = 221,3,Hangup
; Line 3
exten = 222,1,Dial(${PHONES3},20,Ttm)
exten = 222,2,Macro(vmessage,${PHONES3VM})
exten = 222,3,Hangup
; Line 4
exten = 223,1,Dial(${PHONES4},20,Ttm)
exten = 223,2,Macro(vmessage,${PHONES4VM})
exten = 223,3,Hangup
; Line 5
exten = 224,1,Dial(${PHONES5},20,Ttm)
exten = 224,2,Macro(vmessage,${PHONES5VM})
exten = 224,3,Hangup
; Line 6
exten = 225,1,Dial(${PHONES6},20,Ttm)include = sipdiscount-outbound
exten = 225,2,Macro(vmessage,${PHONES6VM})
exten = 225,3,Hangup
; --
; END DEFINE EXTENSIONS
; --
___
sip.conf
;
; SIP Configuration example for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/[EMAIL PROTECTED] to call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
;
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED]
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; sip debug Show all SIP messages
;
; reload chan_sip.so Reload configuration
[Asterisk-Users] newbie questions
Hi all I am new to this whole field, being it PSTN or voIP. I am currently reading the Switching to VoIP and Asterisk: The Future of Telephony, so hopefully, I will be less clueless soon :) My first question: if I buy a Wildcard TDM400P, with one X100M and three S100M modules, I would be able to have 1 telephone number given out by my company to come in to my asterisk server, and I could plug in 3 analog phones onto that card, am I correct? Hence, do we have a 1-to-1 relationship here for either modules? My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? Thank you all. Cheers fred 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 questions
hi, My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? An E1 has 30 lines, so you would be perfect with a TE110P. Connecting an E1 to a company PABX is however an expensiveoption, so you might want to compare the prices with 10 analogue lines or maybe 5 BRI lines. I would not let the price of hardware decide this because you will need to pay a fixed cost per month for PSTN lines, so check these prices first. Asterisk is scalable in the sence that you can add more later if you have an available PCI slot. Jan ___ --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 questions
[EMAIL PROTECTED] wrote: hi, My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? An E1 has 30 lines, so you would be perfect with a TE110P. Connecting an E1 to a company PABX is however an expensiveoption, so you might want to compare the prices with 10 analogue lines or maybe 5 BRI lines. I would not let the price of hardware decide this because you will need to pay a fixed cost per month for PSTN lines, so check these prices first. Asterisk is scalable in the sence that you can add more later if you have an available PCI slot. Even though they don't appear to be shipping yet, don't forget the TDM2400P's from Digium. Up to 24 FXO or FXS ports per full length PCI card. -- Christopher L. Wade, CCNA, CCDA, CQS-CIPCES, CQS-CWLSS ___ --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 questions
On Thu, 2005-11-17 at 14:48 -0600, Chris Wade wrote: [EMAIL PROTECTED] wrote: hi, My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? An E1 has 30 lines, so you would be perfect with a TE110P. Connecting an E1 to a company PABX is however an expensiveoption, so you might want to compare the prices with 10 analogue lines or maybe 5 BRI lines. I would not let the price of hardware decide this because you will need to pay a fixed cost per month for PSTN lines, so check these prices first. Asterisk is scalable in the sence that you can add more later if you have an available PCI slot. Even though they don't appear to be shipping yet, don't forget the TDM2400P's from Digium. Up to 24 FXO or FXS ports per full length PCI card. ok. So, with this in mind, if I was to acquire that 24 ports, 1 FXO modules, the rest FSX modules. I could have 1 public telephone number to the PSTN (3 wasted for this example), 20 analog phones inside the company's branch, each phone having its extension in Asterisk? Or could I have just the smaller card allowing me 1 FSO and 3 FSX and some kind of hub to connect some number of analog phones (let's say 20)? If so, does the number of FXS limit the number of my simultaneous telephone calls? Sorry for the dumb questions, but your answers are highly appreciated. 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 questions
Fred Blaise wrote: On Thu, 2005-11-17 at 14:48 -0600, Chris Wade wrote: [EMAIL PROTECTED] wrote: hi, My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? An E1 has 30 lines, so you would be perfect with a TE110P. Connecting an E1 to a company PABX is however an expensiveoption, so you might want to compare the prices with 10 analogue lines or maybe 5 BRI lines. I would not let the price of hardware decide this because you will need to pay a fixed cost per month for PSTN lines, so check these prices first. Asterisk is scalable in the sence that you can add more later if you have an available PCI slot. Even though they don't appear to be shipping yet, don't forget the TDM2400P's from Digium. Up to 24 FXO or FXS ports per full length PCI card. ok. So, with this in mind, if I was to acquire that 24 ports, 1 FXO modules, the rest FSX modules. I could have 1 public telephone number to the PSTN (3 wasted for this example), 20 analog phones inside the company's branch, each phone having its extension in Asterisk? Or could I have just the smaller card allowing me 1 FSO and 3 FSX and some kind of hub to connect some number of analog phones (let's say 20)? If so, does the number of FXS limit the number of my simultaneous telephone calls? Sorry for the dumb questions, but your answers are highly appreciated. Sorry, but there is really no such thing as a hub for telephone lines. Each analog phone must be plugged into its own FXS port. -- Christopher L. Wade, CCNA, CCDA, CQS-CIPCES, CQS-CWLSS ___ --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 questions
On Fri, November 18, 2005 0:02, Chris Wade said: Fred Blaise wrote: Sorry, but there is really no such thing as a hub for telephone lines. Each analog phone must be plugged into its own FXS port. Unless you are willing to put telephones in parallel, like you do when connecting multiple telephones to a single PSTN line without a PBX... The big problems there are that: - you won't be able to call between telephones on the same 'line' - all telephones on the same 'line' can eavesdrop on an already existing conversation on that 'line' - you may exceed the connection rate (power use) on the FXS and blow it up... Good luck! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --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 questions
Hi fred, For the branch office you could consider a 2 (or more) port E1/T1 card. You can utilize one port for an incoming E1 from the telco (BT?) and then the second port run to a T1 Channel Bank such as the Rhino 24 port FXO http://www.myphonecall.co.uk/voip/channelbanks/rhino/default.aspx - this is the closest to the 'hub' you describe. You would need the cabling in your patch panel to break up the Rhino's 50-way 'telco' connector to something you can patch (rj45/rj11) or run to your analogue phones. I don't have any recommendations for this as I'm still looking at this for an installation. This would give you an asterisk setup with ZAP channels 0..31 incomming and 32..56 your extensions. You will have the advantage of reliable faxing (ie with fax machines) with this solution instead of going to sip phones and sip/ata/fax. I'm looking to do this in our office closer to christmas - all mentioned hardware known to have good asterisk support. Chris Fred Blaise wrote: Hi all I am new to this whole field, being it PSTN or voIP. I am currently reading the Switching to VoIP and Asterisk: The Future of Telephony, so hopefully, I will be less clueless soon :) My first question: if I buy a Wildcard TDM400P, with one X100M and three S100M modules, I would be able to have 1 telephone number given out by my company to come in to my asterisk server, and I could plug in 3 analog phones onto that card, am I correct? Hence, do we have a 1-to-1 relationship here for either modules? My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? Thank you all. Cheers fred ___ --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 questions
You can put analogue phones in series, but I am not sure how many phones a FSX connection will drag and I don't think the cards are designed for it - don't know. Sounds to me as you would benefit from downloading Asterisk and play around with a few softphones first and maybe buy Digiums starter kit, cause you need to take into mind that Asterisk is not realy a Plug Play environment, it do require people who are comfortable with the command prompt on Linux and have the time to fiddle around with this (or money to pay someone to do so) jan Chris Wade wrote: Fred Blaise wrote: On Thu, 2005-11-17 at 14:48 -0600, Chris Wade wrote: [EMAIL PROTECTED] wrote: hi, My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? An E1 has 30 lines, so you would be perfect with a TE110P. Connecting an E1 to a company PABX is however an expensiveoption, so you might want to compare the prices with 10 analogue lines or maybe 5 BRI lines. I would not let the price of hardware decide this because you will need to pay a fixed cost per month for PSTN lines, so check these prices first. Asterisk is scalable in the sence that you can add more later if you have an available PCI slot. Even though they don't appear to be shipping yet, don't forget the TDM2400P's from Digium. Up to 24 FXO or FXS ports per full length PCI card. ok. So, with this in mind, if I was to acquire that 24 ports, 1 FXO modules, the rest FSX modules. I could have 1 public telephone number to the PSTN (3 wasted for this example), 20 analog phones inside the company's branch, each phone having its extension in Asterisk? Or could I have just the smaller card allowing me 1 FSO and 3 FSX and some kind of hub to connect some number of analog phones (let's say 20)? If so, does the number of FXS limit the number of my simultaneous telephone calls? Sorry for the dumb questions, but your answers are highly appreciated. Sorry, but there is really no such thing as a hub for telephone lines. Each analog phone must be plugged into its own FXS port. ___ --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 questions
On Fri, 2005-11-18 at 01:09 +0100, [EMAIL PROTECTED] wrote: You can put analogue phones in series, but I am not sure how many phones a FSX connection will drag and I don't think the cards are designed for it - don't know. Sounds to me as you would benefit from downloading Asterisk and play around with a few softphones first and maybe buy Digiums starter kit, cause you need to take into mind that Asterisk is not realy a Plug Play environment, it do require people who are comfortable with the command prompt on Linux and have the time to fiddle around with this (or money to pay someone to do so) Thanks to all. I have a better idea now. Linux is not the issue, I know it quite well. However, I know _nothing_ about telephony (traditional or voIP)... in a process of getting onto that learning curve... jan fred Chris Wade wrote: Fred Blaise wrote: On Thu, 2005-11-17 at 14:48 -0600, Chris Wade wrote: [EMAIL PROTECTED] wrote: hi, My second question: for a branch office of about 20 people, which E1 card do you advise? Would the TE210P be a good choice? (number of concurrent calls would be max 10 for now) Why? An E1 has 30 lines, so you would be perfect with a TE110P. Connecting an E1 to a company PABX is however an expensiveoption, so you might want to compare the prices with 10 analogue lines or maybe 5 BRI lines. I would not let the price of hardware decide this because you will need to pay a fixed cost per month for PSTN lines, so check these prices first. Asterisk is scalable in the sence that you can add more later if you have an available PCI slot. Even though they don't appear to be shipping yet, don't forget the TDM2400P's from Digium. Up to 24 FXO or FXS ports per full length PCI card. ok. So, with this in mind, if I was to acquire that 24 ports, 1 FXO modules, the rest FSX modules. I could have 1 public telephone number to the PSTN (3 wasted for this example), 20 analog phones inside the company's branch, each phone having its extension in Asterisk? Or could I have just the smaller card allowing me 1 FSO and 3 FSX and some kind of hub to connect some number of analog phones (let's say 20)? If so, does the number of FXS limit the number of my simultaneous telephone calls? Sorry for the dumb questions, but your answers are highly appreciated. Sorry, but there is really no such thing as a hub for telephone lines. Each analog phone must be plugged into its own FXS port. ___ --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 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 questions
Hello Hiu, Monday, November 7, 2005, 4:51:35 AM, you wrote: HYO i am pretty new to asterisk. hope to learn more. HYO i have this notice from the console. when i was doing the echo testing HYO by putting the context=default. then, i called out 600 to get the echo HYO test, i can hear the operator talking, but i cant really hear the playback. HYO i am trying to dig around from info from the log files. HYO what does it mean? HYO RFC3389 support incomplete. Turn off on client if possible HYO hope to help..thanks That means that you have to turn off silence suppression in your softphone (in xlite is named transmit silence). Hope it helps! -- Best regards, Alessiomailto:[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] newbie questions
i am pretty new to asterisk. hope to learn more. i have this notice from the console. when i was doing the echo testing by putting the context=default. then, i called out 600 to get the echo test, i can hear the operator talking, but i cant really hear the playback. i am trying to dig around from info from the log files. what does it mean? RFC3389 support incomplete. Turn off on client if possible hope to help..thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie questions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, I've some questions about asterisk, and in general about voip, please help me :) 1. I've SIP accounts on external servers, and I would that my local server will connect with those and redirect all calls from those to an internal SIP account (just one). It's possible to do that? In this case, I think asterisk will work as UA for external accounts, and as sip server for internal. I've to use SER with asterisk? 2. the internal account it's important that will be SIP, or I could forward calls from my external sip account to an h323 account? 3. I could configure a voicemail account (with an internal number) for all calls that I would redirect from all internal phones? 4. I could use a welcome message on an internal account, and/or auto attendant? I hope this is clear. Any advice to put me in the right direction will be appreciated. Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCQ+AHMakHrsrHP9wRAmbDAJ428+4F+R/RSv0CGMVZVwo73z1OAwCgoiDe F6eXRsp/JX4QD78tDE9Jiro= =Zx81 -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
[Asterisk-Users] newbie questions
I'm sorry, I'm sure you get alot of this, but I'm new to Asterisk and could use some help. I'm investigating migrating our small business phone system over to Asterisk and VOIP. Eventually we'll have around 4 incoming SIP (or IAX if I can find one) accounts for PSTN incoming/outgoing, then SIP hardphones in the office. I installed Asterisk on OS X, which might be why I'm having problems. I have Asterisk up and running fine, although it's giving one warning on startup: WARNING[-1610571284]: File chan_iax2.c, Line 5170 (set_config): Ignoring port for now. I'm not too concerned with this, because for now I'm just trying to get an SIP softphone (X-Lite for OS X) connected to Asterisk, so I don't need IAX listening on whatever port isn't working. I setup a very basic config to let X-Lite connect, but all I see is Awaiting Proxy Information in X-Lite. I see with netstat on the server that it has a UDP for *.sip open, so I think it should be listening for incoming, but it seems like it's not. I don't see a firewall running, so I'm not really sure what's going on. I should be getting an SIP hardphone in later this week, but I'd like to try to get this debugged now. If anyone could help I'd be much appreciative. If you guys have any more questions or want to see my config files, please ask. Thanks, -Brian Nehring ___ 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 questions
Ignore the error if it isn't messing anything up. Check out the Wiki here http://www.voip-info.org/tiki-index.php?page=Asterisk A search of X-lite here also yields proper setup info for the softphone to Asterisk connection. The archive of this list can be search via google by entering... site:lists.digium.com some parameter Als try the documentation link at digium.com Regards, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Nehring Sent: Monday, March 07, 2005 2:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] newbie questions I'm sorry, I'm sure you get alot of this, but I'm new to Asterisk and could use some help. I'm investigating migrating our small business phone system over to Asterisk and VOIP. Eventually we'll have around 4 incoming SIP (or IAX if I can find one) accounts for PSTN incoming/outgoing, then SIP hardphones in the office. I installed Asterisk on OS X, which might be why I'm having problems. I have Asterisk up and running fine, although it's giving one warning on startup: WARNING[-1610571284]: File chan_iax2.c, Line 5170 (set_config): Ignoring port for now. I'm not too concerned with this, because for now I'm just trying to get an SIP softphone (X-Lite for OS X) connected to Asterisk, so I don't need IAX listening on whatever port isn't working. I setup a very basic config to let X-Lite connect, but all I see is Awaiting Proxy Information in X-Lite. I see with netstat on the server that it has a UDP for *.sip open, so I think it should be listening for incoming, but it seems like it's not. I don't see a firewall running, so I'm not really sure what's going on. I should be getting an SIP hardphone in later this week, but I'd like to try to get this debugged now. If anyone could help I'd be much appreciative. If you guys have any more questions or want to see my config files, please ask. Thanks, -Brian Nehring ___ 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 questions
I've read through a good amount of documentation on voip-info.org, but hadn't found a solution, so I thought this list might help. I'm not great with linux, and I suspect there might be a port problem... maybe Asterisk isn't listening for SIP clients. How would I go about checking this? X-Lite configuration is pretty straightforward, you just give it username/password and point it at a SIP proxy. However, as far as I can tell it isn't able to register, or it's not listening to Asterisk... hard to tell really. -Brian On Mon, 7 Mar 2005 14:32:57 -0700, Wiley Siler [EMAIL PROTECTED] wrote: Ignore the error if it isn't messing anything up. Check out the Wiki here http://www.voip-info.org/tiki-index.php?page=Asterisk A search of X-lite here also yields proper setup info for the softphone to Asterisk connection. The archive of this list can be search via google by entering... site:lists.digium.com some parameter Als try the documentation link at digium.com Regards, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Nehring Sent: Monday, March 07, 2005 2:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] newbie questions I'm sorry, I'm sure you get alot of this, but I'm new to Asterisk and could use some help. I'm investigating migrating our small business phone system over to Asterisk and VOIP. Eventually we'll have around 4 incoming SIP (or IAX if I can find one) accounts for PSTN incoming/outgoing, then SIP hardphones in the office. I installed Asterisk on OS X, which might be why I'm having problems. I have Asterisk up and running fine, although it's giving one warning on startup: WARNING[-1610571284]: File chan_iax2.c, Line 5170 (set_config): Ignoring port for now. I'm not too concerned with this, because for now I'm just trying to get an SIP softphone (X-Lite for OS X) connected to Asterisk, so I don't need IAX listening on whatever port isn't working. I setup a very basic config to let X-Lite connect, but all I see is Awaiting Proxy Information in X-Lite. I see with netstat on the server that it has a UDP for *.sip open, so I think it should be listening for incoming, but it seems like it's not. I don't see a firewall running, so I'm not really sure what's going on. I should be getting an SIP hardphone in later this week, but I'd like to try to get this debugged now. If anyone could help I'd be much appreciative. If you guys have any more questions or want to see my config files, please ask. Thanks, -Brian Nehring ___ 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 questions
On Mon, March 7, 2005 22:50, Brian Nehring said: I've read through a good amount of documentation on voip-info.org, but hadn't found a solution, so I thought this list might help. I'm not snip just give it username/password and point it at a SIP proxy. However, as far as I can tell it isn't able to register, or it's not listening to Asterisk... hard to tell really. snip On Mon, 7 Mar 2005 14:32:57 -0700, Wiley Siler [EMAIL PROTECTED] wrote: Ignore the error if it isn't messing anything up. Check out the Wiki here http://www.voip-info.org/tiki-index.php?page=Asterisk A search of X-lite here also yields proper setup info for the softphone to Asterisk connection. The archive of this list can be search via google by entering... site:lists.digium.com some parameter snip From your reply above, it is not clear to me whether you even read the reply, or tried what was suggested? Searching the Wiki for 'X-lite conf' gives a link to the X-lite page, which links to the xten page, which has a link to the X0lite and Asterisk configuration PDF file... Took me 30 seconds... If you *did* follow the PDF (which I cannot tell from either your initial post or your reply), then maybe the X-lite specific config data and logs would be helpful?... -- FP Also an *-n00b, just very skilled at Googling... ___ 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 questions
I've read through a good amount of documentation on voip-info.org, but hadn't found a solution, so I thought this list might help. I'm not great with linux, and I suspect there might be a port problem... maybe Asterisk isn't listening for SIP clients. How would I go about checking this? X-Lite configuration is pretty straightforward, you just give it username/password and point it at a SIP proxy. However, as far as I can tell it isn't able to register, or it's not listening to Asterisk... hard to tell really. If you RIGHT-click on the sliver skin of the X-Lite console and LEFT-click on Diagnostic log you will see the debug information as X-Lite is trying to register with Asterisk. The operative part is: SIP/2.0 followed by a number and a status code. You will note that the status code is the same as HTTP status codes I.E. 2XX = OK, did it and 4XX means your client is the problem and 5XX is something is wrong with the server. Looking thru the diag log will get you started in the right direction to troubleshoot. For the codes, look here: http://www.zvon.org/tmRFC/RFC2543/Output/chapter5.html hth ___ 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 questions
I am sure that asterisk is listening for SIP clients. Did you configure your sip.conf correctly? More info to look at... site:lists.digium.com sip x-lite If you are building this form scratch and cannot get the basics compelted, I would just dump it and go to a build of [EMAIL PROTECTED] The built in GUI lets you get basic install completed very quickly. Cheers, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Nehring Sent: Monday, March 07, 2005 2:51 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] newbie questions I've read through a good amount of documentation on voip-info.org, but hadn't found a solution, so I thought this list might help. I'm not great with linux, and I suspect there might be a port problem... maybe Asterisk isn't listening for SIP clients. How would I go about checking this? X-Lite configuration is pretty straightforward, you just give it username/password and point it at a SIP proxy. However, as far as I can tell it isn't able to register, or it's not listening to Asterisk... hard to tell really. -Brian On Mon, 7 Mar 2005 14:32:57 -0700, Wiley Siler [EMAIL PROTECTED] wrote: Ignore the error if it isn't messing anything up. Check out the Wiki here http://www.voip-info.org/tiki-index.php?page=Asterisk A search of X-lite here also yields proper setup info for the softphone to Asterisk connection. The archive of this list can be search via google by entering... site:lists.digium.com some parameter Als try the documentation link at digium.com Regards, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Nehring Sent: Monday, March 07, 2005 2:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] newbie questions I'm sorry, I'm sure you get alot of this, but I'm new to Asterisk and could use some help. I'm investigating migrating our small business phone system over to Asterisk and VOIP. Eventually we'll have around 4 incoming SIP (or IAX if I can find one) accounts for PSTN incoming/outgoing, then SIP hardphones in the office. I installed Asterisk on OS X, which might be why I'm having problems. I have Asterisk up and running fine, although it's giving one warning on startup: WARNING[-1610571284]: File chan_iax2.c, Line 5170 (set_config): Ignoring port for now. I'm not too concerned with this, because for now I'm just trying to get an SIP softphone (X-Lite for OS X) connected to Asterisk, so I don't need IAX listening on whatever port isn't working. I setup a very basic config to let X-Lite connect, but all I see is Awaiting Proxy Information in X-Lite. I see with netstat on the server that it has a UDP for *.sip open, so I think it should be listening for incoming, but it seems like it's not. I don't see a firewall running, so I'm not really sure what's going on. I should be getting an SIP hardphone in later this week, but I'd like to try to get this debugged now. If anyone could help I'd be much appreciative. If you guys have any more questions or want to see my config files, please ask. Thanks, -Brian Nehring ___ 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 questions
I actually got X-Lite talking to the server, finally. I didn't have to change any of my Asterisk servers... I just kept fooling around with X-Lite and watching the diagnostics log and it finally worked. I can't really say what fixed it, I don't even feel like I changed anything. Oh well, thanks for all the advice, the Diagnostics Log in X-Lite and running Asterisk with -vgcd helped quite a bit. [EMAIL PROTECTED] also looks great, I'm going to install that tomorrow, hopefully the GUI will ease some of the learning curve. -Brian ___ 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 questions
On Tue, 2005-03-08 at 14:57, Brian Nehring wrote: I actually got X-Lite talking to the server, finally. I didn't have to change any of my Asterisk servers... I just kept fooling around with X-Lite and watching the diagnostics log and it finally worked. I can't really say what fixed it, I don't even feel like I changed anything. Oh well, thanks for all the advice, the Diagnostics Log in X-Lite and running Asterisk with -vgcd helped quite a bit. [EMAIL PROTECTED] also looks great, I'm going to install that tomorrow, hopefully the GUI will ease some of the learning curve. Is Xlite running on Windows or Linux? -Brian ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 questions
Xlite for OS X actually. On Tue, 08 Mar 2005 15:00:24 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: On Tue, 2005-03-08 at 14:57, Brian Nehring wrote: I actually got X-Lite talking to the server, finally. I didn't have to change any of my Asterisk servers... I just kept fooling around with X-Lite and watching the diagnostics log and it finally worked. I can't really say what fixed it, I don't even feel like I changed anything. Oh well, thanks for all the advice, the Diagnostics Log in X-Lite and running Asterisk with -vgcd helped quite a bit. [EMAIL PROTECTED] also looks great, I'm going to install that tomorrow, hopefully the GUI will ease some of the learning curve. Is Xlite running on Windows or Linux? -Brian ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 questions
On Tue, 2005-03-08 at 16:56, Brian Nehring wrote: Xlite for OS X actually. bummer, I've been wanting to get it running under Linux. On Tue, 08 Mar 2005 15:00:24 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: On Tue, 2005-03-08 at 14:57, Brian Nehring wrote: I actually got X-Lite talking to the server, finally. I didn't have to change any of my Asterisk servers... I just kept fooling around with X-Lite and watching the diagnostics log and it finally worked. I can't really say what fixed it, I don't even feel like I changed anything. Oh well, thanks for all the advice, the Diagnostics Log in X-Lite and running Asterisk with -vgcd helped quite a bit. [EMAIL PROTECTED] also looks great, I'm going to install that tomorrow, hopefully the GUI will ease some of the learning curve. Is Xlite running on Windows or Linux? -Brian ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie questions
Hello, At the office we have a Lucent PBX, which has 3 lines coming from the CO. 2 are used for phones, 1 for fax. In the office, we have 16 phones. All those are connected in the PBX. We do not have an automated system nor voicemail system for now. But this is something we would like to have now. Since we do a lot of work with Linux, I was asked to look into asterisk to deplace our PBX. Software-wise, I don't have any problems yet, doesn't look too bad hard to configure. Now, I know I would need a quad-port FXO card for our lines coming in from the CO in that PC. What would be the best way to connect all those 16 digital phones to the Asterisk box? I could always buy quad-ports FXS cards for now, as we don't use the 16 phones, but I don't think that's going to work well in the future when the company grows and we require more phones. Keep in mind telephony is very very new to me. Any help would be very appreciated. -- Jean-Francois Theroux Systems administrator PrivalODC 514.726.3732 http://www.privalodc.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 questions
Jean-Francois Theroux wrote: Hello, At the office we have a Lucent PBX, which has 3 lines coming from the CO. 2 are used for phones, 1 for fax. In the office, we have 16 phones. All those are connected in the PBX. We do not have an automated system nor voicemail system for now. But this is something we would like to have now. Since we do a lot of work with Linux, I was asked to look into asterisk to deplace our PBX. Software-wise, I don't have any problems yet, doesn't look too bad hard to configure. Now, I know I would need a quad-port FXO card for our lines coming in from the CO in that PC. What would be the best way to connect all those 16 digital phones to the Asterisk box? I could always buy quad-ports FXS cards for now, as we don't use the 16 phones, but I don't think that's going to work well in the future when the company grows and we require more phones. Keep in mind telephony is very very new to me. Any help would be very appreciated. Unfortunately, if you are going to be replacing the Lucent PBX, the phones are goinf to have to go to, unless thay are regular analog phones, which I seriously doubt. You have a few options from there: A channel bank connected to asterisk via a T-1 card, with analog phones connected to the channel bank ports VOIP phones coonnected to the office's network. (this would be my recommendation) There are other configuration options, but these are probably the most popular. ___ 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 questions
First tip, use a descriptive subject line. On Wed, 2005-03-02 at 15:14 -0500, Jean-Francois Theroux wrote: Hello, At the office we have a Lucent PBX, which has 3 lines coming from the CO. 2 are used for phones, 1 for fax. In the office, we have 16 phones. All those are connected in the PBX. We do not have an automated system nor voicemail system for now. But this is something we would like to have now. Since we do a lot of work with Linux, I was asked to look into asterisk to deplace our PBX. Software-wise, I don't have any problems yet, doesn't look too bad hard to configure. Now, I know I would need a quad-port FXO card for our lines coming in from the CO in that PC. What would be the best way to connect all those 16 digital phones to the Asterisk box? I could always buy quad-ports FXS cards for now, as we don't use the 16 phones, but I don't think that's going to work well in the future when the company grows and we require more phones. Unless the phones are IP phones, you won't be able to use them directly. You may wish to look at what is possible with your current PBX to put asterisk to the side of it to handle your IVR and/or your voicemail. You could even use it as a VoIP gateway. It all depends on what features your current PBX will allow you to implement. If you would rather replace your PBX completely with asterisk, you really need to look at a T1 and a channel bank. Adtran and CAC Adit 600 units are modular. The Adit units use 8 port cards to get service. So you could have an 8 port FXO card and 2 8 port FXS port cards. This would let you hook up 8 incoming lines and 16 extensions. Of course since all the channels are programmable, you get a third FXS port and you can mix and match ports 1-8 FXO and 16-24 FXS ports as long as you don't exceed 24 ports. -- Steven Critchfield [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] newbie questions
Ok, so far I know I would need a 4 ports FXO card for the incoming phone lines. I was thinking a Digium TDM04B. Then, I would need a card that would connect to a Lucent 306EC expension modules (3 incoming lines, 8 phones) that goes in a Partner type of PBX system. We would like to keep those cards, and have the Asterisk system interact with it. Would that be possible? What type of Digium (or any other brand) card would I need for that? Since we have 2 expension boards, would I need more than one card to connect to it? Thanks, Nik Martin wrote: Jean-Francois Theroux wrote: Hello, At the office we have a Lucent PBX, which has 3 lines coming from the CO. 2 are used for phones, 1 for fax. In the office, we have 16 phones. All those are connected in the PBX. We do not have an automated system nor voicemail system for now. But this is something we would like to have now. Since we do a lot of work with Linux, I was asked to look into asterisk to deplace our PBX. Software-wise, I don't have any problems yet, doesn't look too bad hard to configure. Now, I know I would need a quad-port FXO card for our lines coming in from the CO in that PC. What would be the best way to connect all those 16 digital phones to the Asterisk box? I could always buy quad-ports FXS cards for now, as we don't use the 16 phones, but I don't think that's going to work well in the future when the company grows and we require more phones. Keep in mind telephony is very very new to me. Any help would be very appreciated. Unfortunately, if you are going to be replacing the Lucent PBX, the phones are goinf to have to go to, unless thay are regular analog phones, which I seriously doubt. You have a few options from there: A channel bank connected to asterisk via a T-1 card, with analog phones connected to the channel bank ports VOIP phones coonnected to the office's network. (this would be my recommendation) There are other configuration options, but these are probably the most popular. ___ 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 -- Jean-Francois Theroux Systems administrator PrivalODC 514.726.3732 http://www.privalodc.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 questions
Jean-Francois Theroux wrote: Ok, so far I know I would need a 4 ports FXO card for the incoming phone lines. I was thinking a Digium TDM04B. Then, I would need a card that would connect to a Lucent 306EC expension modules (3 incoming lines, 8 phones) that goes in a Partner type of PBX system. We would like to keep those cards, and have the Asterisk system interact with it. Would that be possible? What type of Digium (or any other brand) card would I need for that? Since we have 2 expension boards, would I need more than one card to connect to it? Thanks, I'm not up to speed on Lucent cards, but a T-1 (E-1) card may be all you need to communicate with a legacy PBX from Asterisk. Someone with Partner experience will certainly know more than me. Nik Martin wrote: Jean-Francois Theroux wrote: Hello, At the office we have a Lucent PBX, which has 3 lines coming from the CO. 2 are used for phones, 1 for fax. In the office, we have 16 phones. All those are connected in the PBX. We do not have an automated system nor voicemail system for now. But this is something we would like to have now. Since we do a lot of work with Linux, I was asked to look into asterisk to deplace our PBX. Software-wise, I don't have any problems yet, doesn't look too bad hard to configure. Now, I know I would need a quad-port FXO card for our lines coming in from the CO in that PC. What would be the best way to connect all those 16 digital phones to the Asterisk box? I could always buy quad-ports FXS cards for now, as we don't use the 16 phones, but I don't think that's going to work well in the future when the company grows and we require more phones. Keep in mind telephony is very very new to me. Any help would be very appreciated. Unfortunately, if you are going to be replacing the Lucent PBX, the phones are goinf to have to go to, unless thay are regular analog phones, which I seriously doubt. You have a few options from there: A channel bank connected to asterisk via a T-1 card, with analog phones connected to the channel bank ports VOIP phones coonnected to the office's network. (this would be my recommendation) There are other configuration options, but these are probably the most popular. ___ 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 questions
Getting Asterisk to work with the proprietary phones from your Lucent PBX is not likely to happen ... you might be able to use Asterisk to act as a front end to your Lucent PBX by using FXS cards ... you would have Asterisk interface with the CO, then ring the Lucent box ... but you would not be able to make use of most of the PBX features of Asterisk doing this ... about all you would get for your trouble is Attendant and VoiceMail ... how your Lucent phones would then retrieve voicemail would be a real challenge ... suspect they would have to pick up an outside line and dial a code into Asterisk to retrieve it ... transfers from Asterisk to specific Lucent extensions would not work ... You might be able to get a card for the Lucent box that would permit it to accept connections from a standard analog phone ... this would let you connect Asterisk as an extension ... grabbling an outside line would be a dial 9 chore for Asterisk ... but connecting it as an extension would permit the auto attendant to do direct transfers to Lucent phones ... Best thing to do would be to replace the proprietary phones with generic SIP phones ... doing this would make your Asterisk configuration cleaner and easier to manage ... also a lot more predictable ... Check the wiki at http://www.voip-info.org ... there are a number of articles about how to interface Asterisk with legacy phone systems ... some are quite creative and overcome most limitations ... G.Hendershot -Original Message- From: Jean-Francois Theroux [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 02, 2005 3:14 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] newbie questions Hello, At the office we have a Lucent PBX, which has 3 lines coming from the CO. 2 are used for phones, 1 for fax. In the office, we have 16 phones. All those are connected in the PBX. We do not have an automated system nor voicemail system for now. But this is something we would like to have now. Since we do a lot of work with Linux, I was asked to look into asterisk to deplace our PBX. Software-wise, I don't have any problems yet, doesn't look too bad hard to configure. Now, I know I would need a quad-port FXO card for our lines coming in from the CO in that PC. What would be the best way to connect all those 16 digital phones to the Asterisk box? I could always buy quad-ports FXS cards for now, as we don't use the 16 phones, but I don't think that's going to work well in the future when the company grows and we require more phones. Keep in mind telephony is very very new to me. Any help would be very appreciated. -- Jean-Francois Theroux Systems administrator PrivalODC 514.726.3732 http://www.privalodc.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 questions
I've got one freaky budgetone that wont work using dhcp assign ip address via mac code. Basically I need to assign it an ip address using the phones internal web server. Maybe this was your problem as well. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Panco Sent: Tuesday, February 08, 2005 1:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] newbie questions i installed it the other day but from some reason can only get one of my budgetone 100's to register...any thoughts? I have tried upgrading firmare but that didn't seem to work. thanks in advance, ken Steve Rawlings wrote: Why not try [EMAIL PROTECTED], it only takes about an hour to install and be up and running with softphones like x-lite. This takes care of the os and asterisk in one cd. Steve - Original Message - From: Shaoul Jacobson - TELLINK [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 08, 2005 4:44 PM Subject: [Asterisk-Users] newbie questions Hi, I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards) 1. the distro I downloaded a free mandrake 10.0 - 3 CD's) but some packages seem missing (some C or C++ or python ...) (buy the full version ) maybe the latest fedora is more complete ? or easier to complete with rpmfind (I am green to linux too, but I open my windows gates to the tux) (bsd, debian are a bit too tech for me yet, no flaming please.) I prefer ready made rpm's or alike than compile AT THIS TIME. (I promise to improve over time) 2. download any rpm ? or I must download sources and 'make install' ? (I found one iso, but it seemed to require a pstn card) (RTFM a second / third time could is always a good option) 3. pure VoIP is it ok to use it in pure VoIP mode without any 'phone cards' ? all (most) settings samples I see include such cards. Needed or not ? 4. g729 not free. It seems that requires some licensing to digium. Can that be without using any 'card' (just VoIP) ? How to control the licenses then ? (I e-mailed them the question, but got no answer) accounting, cdr's, ... that's for later (first I have to be able to phone) regards, Shaoul Jacobson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie questions
Hi, I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards) 1. the distro I downloaded a free mandrake 10.0 - 3 CD's) but some packages seem missing (some C or C++ or python ...) (buy the full version ) maybe the latest fedora is more complete ? or easier to complete with rpmfind (I am green to linux too, but I open my windows gates to the tux) (bsd, debian are a bit too tech for me yet, no flaming please.) I prefer ready made rpm's or alike than compile AT THIS TIME. (I promise to improve over time) 2. download any rpm ? or I must download sources and 'make install' ? (I found one iso, but it seemed to require a pstn card) (RTFM a second / third time could is always a good option) 3. pure VoIP is it ok to use it in pure VoIP mode without any 'phone cards' ? all (most) settings samples I see include such cards. Needed or not ? 4. g729 not free. It seems that requires some licensing to digium. Can that be without using any 'card' (just VoIP) ? How to control the licenses then ? (I e-mailed them the question, but got no answer) accounting, cdr's, ... that's for later (first I have to be able to phone) regards, Shaoul Jacobson ___ 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 questions
Voip to voip (in whatever form that takes sip-sip sip-iax iax-iax) is what I am using asterisk for. I would have thought mandrake would have been ok - but haven't used it for a while. I'm running FC2 (fedora core2) and asterisk complies and runs without any problems. Dont fear make. Apps, for the most part, compile really easily on linux. Follow the instructions to the letter and you shouldn't go wrong. Its often as simple as typing make waiting a bit for stuff to stop happening and then typing make install. Asterisk prompts you with various other options like make help and make samples (or something like that) so its pretty straight forward. You don't need any cards for asterisk. No phone cards, no sound card just whatever allows you to connect to your lan and/or the internet. g729 - isn't required. There are plenty of other codecs you can use for free. Accounting, cdr, ser etc - I haven't got that far myself either. Shaoul Jacobson - TELLINK wrote: Hi, I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards) 1. the distro I downloaded a free mandrake 10.0 - 3 CD's) but some packages seem missing (some C or C++ or python ...) (buy the full version ) maybe the latest fedora is more complete ? or easier to complete with rpmfind (I am green to linux too, but I open my windows gates to the tux) (bsd, debian are a bit too tech for me yet, no flaming please.) I prefer ready made rpm's or alike than compile AT THIS TIME. (I promise to improve over time) 2. download any rpm ? or I must download sources and 'make install' ? (I found one iso, but it seemed to require a pstn card) (RTFM a second / third time could is always a good option) 3. pure VoIP is it ok to use it in pure VoIP mode without any 'phone cards' ? all (most) settings samples I see include such cards. Needed or not ? 4. g729 not free. It seems that requires some licensing to digium. Can that be without using any 'card' (just VoIP) ? How to control the licenses then ? (I e-mailed them the question, but got no answer) accounting, cdr's, ... that's for later (first I have to be able to phone) regards, Shaoul Jacobson ___ 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 questions
I run Debian, and it's not hard to get a base install running. If you want a GUI and such, then it'll be more than follow the screen prompts. I've been writing some Debian documents, if you're interested, email me off-list. Anyhow, on pretty much any distro, you can make your own packages (RPM, DEB, TGZ, whatever) from compiling source using the program called CheckInstall. It'll make removing your programs much easier, and if you deploy many similar systems, you can reuse the packages. I did this when I ran Mandrake, and it worked great, especially when Mandrake didn't have a lot of the software that I wanted to use. More often than not, Debian has what I want, but if I can't find something (ipkungfu, for example) then I'll use checkinstall to make it easy to remove in the future. -- Dana On Tue, 08 Feb 2005 17:06:14 +, Mark Benson [EMAIL PROTECTED] wrote: Voip to voip (in whatever form that takes sip-sip sip-iax iax-iax) is what I am using asterisk for. I would have thought mandrake would have been ok - but haven't used it for a while. I'm running FC2 (fedora core2) and asterisk complies and runs without any problems. Dont fear make. Apps, for the most part, compile really easily on linux. Follow the instructions to the letter and you shouldn't go wrong. Its often as simple as typing make waiting a bit for stuff to stop happening and then typing make install. Asterisk prompts you with various other options like make help and make samples (or something like that) so its pretty straight forward. You don't need any cards for asterisk. No phone cards, no sound card just whatever allows you to connect to your lan and/or the internet. g729 - isn't required. There are plenty of other codecs you can use for free. Accounting, cdr, ser etc - I haven't got that far myself either. Shaoul Jacobson - TELLINK wrote: Hi, I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards) 1. the distro I downloaded a free mandrake 10.0 - 3 CD's) but some packages seem missing (some C or C++ or python ...) (buy the full version ) maybe the latest fedora is more complete ? or easier to complete with rpmfind (I am green to linux too, but I open my windows gates to the tux) (bsd, debian are a bit too tech for me yet, no flaming please.) I prefer ready made rpm's or alike than compile AT THIS TIME. (I promise to improve over time) 2. download any rpm ? or I must download sources and 'make install' ? (I found one iso, but it seemed to require a pstn card) (RTFM a second / third time could is always a good option) 3. pure VoIP is it ok to use it in pure VoIP mode without any 'phone cards' ? all (most) settings samples I see include such cards. Needed or not ? 4. g729 not free. It seems that requires some licensing to digium. Can that be without using any 'card' (just VoIP) ? How to control the licenses then ? (I e-mailed them the question, but got no answer) accounting, cdr's, ... that's for later (first I have to be able to phone) regards, Shaoul Jacobson ___ 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 questions
Why not try [EMAIL PROTECTED], it only takes about an hour to install and be up and running with softphones like x-lite. This takes care of the os and asterisk in one cd. Steve - Original Message - From: Shaoul Jacobson - TELLINK [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 08, 2005 4:44 PM Subject: [Asterisk-Users] newbie questions Hi, I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards) 1. the distro I downloaded a free mandrake 10.0 - 3 CD's) but some packages seem missing (some C or C++ or python ...) (buy the full version ) maybe the latest fedora is more complete ? or easier to complete with rpmfind (I am green to linux too, but I open my windows gates to the tux) (bsd, debian are a bit too tech for me yet, no flaming please.) I prefer ready made rpm's or alike than compile AT THIS TIME. (I promise to improve over time) 2. download any rpm ? or I must download sources and 'make install' ? (I found one iso, but it seemed to require a pstn card) (RTFM a second / third time could is always a good option) 3. pure VoIP is it ok to use it in pure VoIP mode without any 'phone cards' ? all (most) settings samples I see include such cards. Needed or not ? 4. g729 not free. It seems that requires some licensing to digium. Can that be without using any 'card' (just VoIP) ? How to control the licenses then ? (I e-mailed them the question, but got no answer) accounting, cdr's, ... that's for later (first I have to be able to phone) regards, Shaoul Jacobson ___ 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 questions
i installed it the other day but from some reason can only get one of my budgetone 100's to register...any thoughts? I have tried upgrading firmare but that didn't seem to work. thanks in advance, ken Steve Rawlings wrote: Why not try [EMAIL PROTECTED], it only takes about an hour to install and be up and running with softphones like x-lite. This takes care of the os and asterisk in one cd. Steve - Original Message - From: Shaoul Jacobson - TELLINK [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 08, 2005 4:44 PM Subject: [Asterisk-Users] newbie questions Hi, I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards) 1. the distro I downloaded a free mandrake 10.0 - 3 CD's) but some packages seem missing (some C or C++ or python ...) (buy the full version ) maybe the latest fedora is more complete ? or easier to complete with rpmfind (I am green to linux too, but I open my windows gates to the tux) (bsd, debian are a bit too tech for me yet, no flaming please.) I prefer ready made rpm's or alike than compile AT THIS TIME. (I promise to improve over time) 2. download any rpm ? or I must download sources and 'make install' ? (I found one iso, but it seemed to require a pstn card) (RTFM a second / third time could is always a good option) 3. pure VoIP is it ok to use it in pure VoIP mode without any 'phone cards' ? all (most) settings samples I see include such cards. Needed or not ? 4. g729 not free. It seems that requires some licensing to digium. Can that be without using any 'card' (just VoIP) ? How to control the licenses then ? (I e-mailed them the question, but got no answer) accounting, cdr's, ... that's for later (first I have to be able to phone) regards, Shaoul Jacobson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie: questions
I currently subscribe to acedsl for voip service. I have ast. running on an old compaq with 2 clone fxo cards. Everything is going good (thanks to lurking around here). The box answers and dials over the analog. I want to bring in the 2 digitil from acecape. They currenty go to a cisco ata 186. My question are ; Can I address the ata from my ast to get it to ring from my analog as well as it maintaining its current duties being addressed by acedsl? I dont want to switch out till Im finished learning. Or do I need acedsl to set up an sip or iax channel for my ast. to talk to ace. Tia j.p. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie questions / documentation feedback?
I'm slowly but surely bringing up my first asterisk system, plowing through the wiki and the asterisk documentation project's book as I go, trying to understand it all as I go. Needless to say, I'm getting myself quite confused at times. What is the appropriate venue to report my confusion, and possibly give feedback to the book authors in the form of the litany of questions that come to mind as I read, which are not answered within a page of the item that prompted the question? For example: Chapter 4, page 24: What does 'context=default' refer to? Is this the section of the dialplan where incoming calls are to start? What does the channel number 'channel = 1' mean? Is this the 'slot' within the TDM400P that the FXS port is installed in? WHat if I were to install multiple TDM400P's, is there card# address, or do I continue numbering channels 5 - 8? In the case of multiple cards, how do I determine which PCI slot has 1-4, which 5-8, etc? WHat about the X100P? How is its channel # assigned? Page 25: On page 24, you defined channels 1 2, now in the 'recap' of the configuration, you mention channels 1 4. Which is it? IAX The example in the book begins with 'port=5036'. The sample iax.conf created by `make samples` starts with the line 'bindport=4569'. Is this a typo, an alias, or are they two separate parameters, and if so, what for? Is there a convention for IAX ports? I don't want to be too creative at this stage of the game, so if there's a standard convention, I'd just as soon follow it. ...I could go on, but it's clear from reading this list that the vast majority of you are far beyond this point, and I don't want to drag the list down. I'd rather go offline with someone who would be interested in mining my confusion towards the end of clarifying the documentation and making the process easier for the next generation of users. -- Rick Green They that can give up essential liberty to obtain a little temporary safety, deserve neither liberty nor safety. -Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie questions from South Africa: Initial setup
Hi All, I have been researching Asterisk for a few days now and have read hundreds of web pages and other documents. While some things are getting clearer, others are not. I have managed to install Asterisk 1.0.1 on Debian testing (simple as 'apt-get install asterisk'). I understand FXS/FXO, codecs available, feature and functionality available etc. I have found several Getting Started docs. None have proved good enough to get a system working (or to understand how). So my questions: 1. Is there anynone on this list from South Africa using Asterisk? 2. How can I setup a minimal system for testing. Question 2 can be as simple as two SIP phones on my LAN. Preferrable would be to setup my ISDN 2a lines but I cannot figure out how to connect and setup ISDN on Asterisk. I have four MSN numbers on my ISDN 2a line. I think its possible to set this up with an internal ISDN card and soft phones but I am unsure if this is possible (and if so I definitely don't know where to start). Better would be being able to set this up with an external 3COM ISDN modem since I aready have one but I suspect this is not possible. Any help would be greatly appreciated. I feel like I have all the info scattered through my brain and research docs, but distilling it into a plan of action to get a working system is proving very difficult. Thank in advance all. Richard Howes - ResRequest www.resrequest.com ___ 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] Newbie questions from South Africa: Initial setup
Hi Just a shot in the dark, wouldnt this be coming into a CSU/DSU then into a Digium PCI card of some sort, that is where asterisk would pick it up? - Original Message - From: Richard Howes [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 23, 2004 3:41 AM Subject: [Asterisk-Users] Newbie questions from South Africa: Initial setup Hi All, I have been researching Asterisk for a few days now and have read hundreds of web pages and other documents. While some things are getting clearer, others are not. I have managed to install Asterisk 1.0.1 on Debian testing (simple as 'apt-get install asterisk'). I understand FXS/FXO, codecs available, feature and functionality available etc. I have found several Getting Started docs. None have proved good enough to get a system working (or to understand how). So my questions: 1. Is there anynone on this list from South Africa using Asterisk? 2. How can I setup a minimal system for testing. Question 2 can be as simple as two SIP phones on my LAN. Preferrable would be to setup my ISDN 2a lines but I cannot figure out how to connect and setup ISDN on Asterisk. I have four MSN numbers on my ISDN 2a line. I think its possible to set this up with an internal ISDN card and soft phones but I am unsure if this is possible (and if so I definitely don't know where to start). Better would be being able to set this up with an external 3COM ISDN modem since I aready have one but I suspect this is not possible. Any help would be greatly appreciated. I feel like I have all the info scattered through my brain and research docs, but distilling it into a plan of action to get a working system is proving very difficult. Thank in advance all. Richard Howes - ResRequest www.resrequest.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie Questions
Hi everyone, I'm going to be helping to set * up for the company I work for, and in doing all my research about it, have found it to be a very viable solution for my SOHO side business at home. I do however have a few questions, forgive me if they're stupid but I'm new to all of this. 1. I want to be able to handle 3 analogue phone lines, with a regular bell telephone line coming into the house. So am I to assume that I want one PCI card for a P300mhz or above with three FXS ports and one FXO port? (the TDM31B). Am I correct in the card that I want? 2. Relating to my first question, say I instead get a card with only one FXS port and one FXO port, can I 'chain' my phones together from the one FXS port and still get the same functionality? (what i mean is one phone line coming out, with a splitter going to my three telephones)? 3. In the future I will be wanting to upgrade to VOIP capabilities for my SOHO Long Distance, is this as simple as getting another card with a T1 interface and an interface port for the phone, then plug it into my existing LAN to get internet connectivity, and still use the TDM31B for regular analogue conversations? 4. Does * support 'ring tone identification' ? Currently I have the outside line coming into the house, then it's split to go off to two phones, then from one of the phones the 'second extension jack' is going to my fax machine, which recognizes the distinctive ring the phone company gave me for the fax #. Will this still work with asterix, or would the fax machine have to be coming directly off the port on the PCI card? 5. Relating back to the splitting of the phone lines, if I have a card with two FXS jacks, and one FXO, and I only wanted two extensions on the line (upstairs, downstairs), could I chain the upstairs lines on one analogue line, and then if i transfer a caller to that extension it will ring on both phones upstairs? Hopefully I'm clear on my questions, Thanks a lot in advance. Matt Gibson Unix Administrator Experthost / NJ Tech Solutions ___ 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] Newbie Questions
From: Matt G [EMAIL PROTECTED] Date: 2004/07/28 Wed PM 08:50:03 GMT To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie Questions Hi everyone, I'm going to be helping to set * up for the company I work for, and in doing all my research about it, have found it to be a very viable solution for my SOHO side business at home. I do however have a few questions, forgive me if they're stupid but I'm new to all of this. 1. I want to be able to handle 3 analogue phone lines, with a regular bell telephone line coming into the house. So am I to assume that I want one PCI card for a P300mhz or above with three FXS ports and one FXO port? (the TDM31B). Am I correct in the card that I want? I believe so. You will need one FXO and three FXS ports, and you should be able to get them all on one card. 2. Relating to my first question, say I instead get a card with only one FXS port and one FXO port, can I 'chain' my phones together from the one FXS port and still get the same functionality? (what i mean is one phone line coming out, with a splitter going to my three telephones)? Yes, that should work, but there's a limit on how many telephones one can drive. 3. In the future I will be wanting to upgrade to VOIP capabilities for my SOHO Long Distance, is this as simple as getting another card with a T1 interface and an interface port for the phone, then plug it into my existing LAN to get internet connectivity, and still use the TDM31B for regular analogue conversations? No. A T1 card does not plug into a LAN. You will need to use the ethernet port on your server, or add a NIC. Once you do so, and get all of that configured, you can bridge calls between your analog ports and SIP/IAX/other VoIP phones. 4. Does * support 'ring tone identification' ? Currently I have the outside line coming into the house, then it's split to go off to two phones, then from one of the phones the 'second extension jack' is going to my fax machine, which recognizes the distinctive ring the phone company gave me for the fax #. Will this still work with asterix, or would the fax machine have to be coming directly off the port on the PCI card? Asterisk has distinctive ring support, but I have not worked with it yet. 5. Relating back to the splitting of the phone lines, if I have a card with two FXS jacks, and one FXO, and I only wanted two extensions on the line (upstairs, downstairs), could I chain the upstairs lines on one analogue line, and then if i transfer a caller to that extension it will ring on both phones upstairs? Yes, it should. No different than a normal analog line that you get from a telco. Again, there's a limit on how many phones one line can drive. Thanks a lot in advance. You're welcome... ___ 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] Newbie Questions
You can plug several phones into an FXS port, but they look like the same phone/extension to Asterisk - so they will all ring together, if one is in use the others will be as well, etc. The big question I see here is whether or not you want each individual phone instrument (or group of instruments) to have their own identity - e.g., do you want to be able to call the kitchen from the bedroom? Do you want to be able to have someone on the phone in the kitchen having a conversation with X in Florida, while someone else in the bedroom is having a conversation with Y in New York? Do you want to be able to set things up so that some phones don't ring late at night, or such that it's possible for an outside caller to be routed to a particular instrument? If you want those sorts of things, you will want to have several FXS ports (or several VoIP phones) so that each phone can have its own identity. If you don't need that sort of thing, then it's not important that each phone get its own FXS port. That being said, you don't want to plug so many phones in that you exceed the REN capacity of the port (the ability of the device to provide the higher voltage used to signal (and, on older phones, to generate) the ringing sound). I don't know what the REN capacity of the cards is, but I know you don't want to go above it. Once upon a time, most phones had a REN of 1, and the telco would guarantee at least 5 REN worth of juice. In the modern age, I've got no idea, but my impression is that modern electronic telephones, particularly those with their own power supplies, are more likely to have a REN in the neighborhood of 0.1, or similar. If you want to use VoIP to get long distance service for your Asterisk box, you don't need to monkey around with a T1 interface - you can use your existing Ethernet and cable/DSL connection to get service, so long as your bandwidth needs are modest. This works smoothest if you sign up with one of the LD providers who's set up to interconnect with Asterisk. If you wanted to, you could also sign up for Vonage or one of the other more consumer-oriented VoIP providers, and plug their network device into an FXO port, and just pretend it's an old-fashioned POTS line. The theoretical downside to that is that you're doing an extra digital - analog - digital conversion, which may cost you some voice quality. The practical downside to that is that the consumer-oriented services are also more expensive, at least at a low call volume, since they're more likely to be priced at $20-30-40-50/month for many minutes or unlimited service, whereas the Asterisk-friendly providers are probably going to charge you on a per-minute (or fractional minute) basis for time used, plus a flat rate per incoming number (if you even want/need one). I have Vonage's 500 minutes/month service and never come close to using up my 500 minutes. Also, I've found voice quality to be better using Asterisk and SIP phones - I suspect it's because I've got more control over the codecs used, but it's hard to say for sure. -- Greg Broiles, JD, EA [EMAIL PROTECTED] (Lists only. Not for confidential communications.) Law Office of Gregory A. Broiles San Jose, CA ___ 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] newbie questions
Hi, I am new to asterisk. And I have some newbie questions :-) I like to use asterisk I our office (around 20 phones) but we need to see if a user is at the moment using his phone so he can not get a second call from someone. Sometimes this is called a telephonecenter where I can see the used lines. Is it possible to configure the frontend to autoaccept internal incoming calls? Our employees often use this feature to search each other Yours, Nicolai Diese Nachricht wurde auf Viren und andere gefaehrliche Inhalte sowie Spam untersucht. ___ 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] Newbie questions about ISDNzapata.conf, outbound dialing, TDMoE
Hi, I spent some hours working my way through the WIKI and a number of other documentations, but after all, three questions are still left: 1. I'm using a Fritz!Card with the i4l driver - no problem at all, my Grandstream BT 100 rings when I diall a regular phone number. Is there any need for me to configure the zapata.conf for the ISDN card to get a special asterisk feature or is this file really only necessary for zapata devices? 2. Is outbound dialing only configured in the extension.conf or do I have to edit another file? 3. I want to connect two asterisk servers over a vpn tunnel (perhaps with static, maybe with dynamic addresses). Is TDMoE the thing I have to look at to set this up? The corresponding wiki article says that I must have a zaptel interface ?! Thanks a lot for any helpful comment. RTFM is okay, as long as you tell me the right page(s), too. ;.)) Stefan -- * in-put GbR - Das Linux-Systemhaus Stefan-Michael Günther Moltkestraße 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de * ___ 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] Newbie questions about ISDNzapata.conf, outbound dialing, TDMoE
On Fri, 4 Jun 2004, Stefan-Michael. [iso-8859-15] Günther (in-put GbR) wrote: 1. I'm using a Fritz!Card with the i4l driver - no problem at all, my Grandstream BT 100 rings when I diall a regular phone number. Is there any need for me to configure the zapata.conf for the ISDN card to get a special asterisk feature or is this file really only necessary for zapata devices? zapata.conf is only for Zap devices. I believe modem.conf configures isdn4linux. You should definitely look at the CAPI driver that Kapejod wrote, though. www.junghanns.net. It delivers much better latency, and more features. 2. Is outbound dialing only configured in the extension.conf or do I have to edit another file? There's stuff in modem.conf. But like I said you should look at Kapejod's CAPI driver. 3. I want to connect two asterisk servers over a vpn tunnel (perhaps with static, maybe with dynamic addresses). Is TDMoE the thing I have to look at to set this up? The corresponding wiki article says that I must have a zaptel interface ?! You'll use IAX to link your two servers. Steve ___ 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] Newbie Questions
Ill apologize right away for asking stupid questions. J System Setup: SER = Proxy Asterisk = Voicemail All sip based setup. What Is required to make asterisk NOT- accept inbound calls/signaling from an unknown host? I tried the peers in sip.conf but it still allows unknown hosts to send it calls. Does anyone have a suggestion or maybe some sample configs? Im trying to extensions.conf dynamic. Is there any other alternative to the DynamicDB program to do something like that at this time? Im trying to avoid having to restart * every time we make a change/addition. Im going to be rolling out a fairly large installation of Asterisk. What is the best way to have them all have the same configs/be synchronized? Does anyone have any good tips/advice on SER+Asterisk integration? I appreciate it. - Darren
Re: [Asterisk-Users] Newbie Questions
Darren Sessions wrote: Ill apologize right away for asking stupid questions. J System Setup: SER = Proxy Asterisk = Voicemail All sip based setup. 1. What Is required to make asterisk NOT- accept inbound calls/signaling from an unknown host? I tried the peers in sip.conf but it still allows unknown hosts to send it calls. Does anyone have a suggestion or maybe some sample configs? Don't put a context directive in the general section. 2. Im trying to extensions.conf dynamic. Is there any other alternative to the DynamicDB program to do something like that at this time? Im trying to avoid having to restart * every time we make a change/addition. DynExtenDB is the definition of EVIL. You don't have to restart asterisk, just issue a reload. You have to reload apache, sendmail, named, and so on so when you make changes, what's so hard about doing it with Asterisk? 3. Im going to be rolling out a fairly large installation of Asterisk. What is the best way to have them all have the same configs/be synchronized? Build a provisioning system around asterisk. I've done itIts not very hard, if you understand the whole concept of Asterisk's architecture. (and no i'm not going to waste time explaining it or providing any of my provisioning system code) 4. Does anyone have any good tips/advice on SER+Asterisk integration? Prepare for hair loss and heartburn. SER is a joke in my book. (Along with RADIUS in a VoIP environment, java and anything Microsoft, but that's off topic) Jeremy McNamara ___ 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] Newbie questions, call waiting/700 calling/etc...
Hi gang, I've just set up my Asterisk server with a X100P (talking to a Vonage Motorola do-dad), and a Cisco IP7960 SIP phone. All is working quite will with outbound and inbound calling. However, I have a few questions. First, regarding call waiting on Vonage/X100P, how do I click over to the call from my Cisco phone? I searched the archives and docs but can't seem to find any information on this. Along those same lines, is there any way to get call waiting caller ID working with this as well? I have what I believe are the appropriate lines in my Zapata.conf file (usecallerid=yes, callerid=asreceived, callwaiting=yes, calleridcallwaiting=yes), but it doesn't seem to work. Second, I got myself set up with an IAXtel number, I was wondering if there are any 700 test numbers out there that will read back the calling number, do an echo test, request a call back, etc. Mostly I want this for testing, and don't want to disturb any 700 number users. :) Third, are there any VoIP providers that I can have Asterisk talk to natively (i.e. via IAX or SIP, not the way I have Vonage set up now)? I'd be looking for a Chicago land number (630 specifically). I looked at VoicePulse, but they don't have any local numbers. Thanks for your help! Brian ___ 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] Newbie questions, call waiting/700 calling/etc...
Third, are there any VoIP providers that I can have Asterisk talk to natively (i.e. via IAX or SIP, not the way I have Vonage set up now)? I'd be looking for a Chicago land number (630 specifically). Check out www.iconnecthere.com - I think they've got pretty good coverage nationally. Cheers Paul ___ 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] Newbie Questions
Look into www.digium.com. Digium's cards are you best choice. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brez Sent: Monday, November 03, 2003 4:03 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie Questions hello, I am completely new to things but was wondering if some one could steer me in the right direction [i.e. i was volunteered to get a PBX running with little or knowledge] good news is, i got a lot of experience with open source / linux / etc. anyhow. we have 4 lines coming in and need 16 extensions. we have the PC and the 16 analog phones. the question is what type of hardware will i need? i.e. modem, a phone 'hub' [or whatever it is called for pluggin all the phone lines into] - basically a small office environment. if any of you using asterisk in a similar environment could spell out exactly what hardware youre using [and perhaps where to buy it] for your office, i would really appreciate the help. thanks. ___ 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] Newbie Questions
hello, I am completely new to things but was wondering if some one could steer me in the right direction [i.e. i was volunteered to get a PBX running with little or knowledge] good news is, i got a lot of experience with open source / linux / etc. anyhow. we have 4 lines coming in and need 16 extensions. we have the PC and the 16 analog phones. the question is what type of hardware will i need? i.e. modem, a phone 'hub' [or whatever it is called for pluggin all the phone lines into] - basically a small office environment. if any of you using asterisk in a similar environment could spell out exactly what hardware youre using [and perhaps where to buy it] for your office, i would really appreciate the help. thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Questions
I built something very similar using: - Adtran TA750 bought off Ebay for around $400 (you can do much better, I was in a hurry.) - A Digium Wildcard T100P - A 4 port FXO card for the TA750 (I searched Google for Adtran FXO and clicked one of the sposored links.) You might have to pick up some other misc bits pieces depending on what the Channel bank you get off EBay has. I had to buy a 25 pair cable with an Amphenol connector and a Type 66 punch down block. Right now my set up has an evil hum on outgoing calls. I suspect the home brew T1 reverse cable I'm using. HTH. brez wrote: hello, I am completely new to things but was wondering if some one could steer me in the right direction [i.e. i was volunteered to get a PBX running with little or knowledge] good news is, i got a lot of experience with open source / linux / etc. anyhow. we have 4 lines coming in and need 16 extensions. we have the PC and the 16 analog phones. the question is what type of hardware will i need? i.e. modem, a phone 'hub' [or whatever it is called for pluggin all the phone lines into] - basically a small office environment. if any of you using asterisk in a similar environment could spell out exactly what hardware youre using [and perhaps where to buy it] for your office, i would really appreciate the help. thanks. ___ 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
SV: SV: [Asterisk-Users] Newbie questions.....
Hi - Jeremy Well - I only refer to what I've been told : Cisco CallManager use skinny protocol communicate to their IP phones, which is a subset of H.323 - But communication between CCM and Voicegateway like 6608-E1/T1 board is using Skinny Client Control Protocol. But then again Cisco could say these things in order to sell more AVVID solution - But they must use some sort of propriaty standard on their IP phones which makes it difficult to use non-cisco IP phones (Other Vendors H.323 devices has to be specified as H.323 endpoints). It can have been some sort of sales trick to make it easy understandable for us guys and our customers. I've never tried to setup a Cisco Callmanager to work with non-cisco products only through a PRI interface. So I had no reason to question their statement. These statements is a year old or so - I haven't worked with Cisco for about a year. But lots of things has happened since that time. I tried looking into is on CCO - Did'nt get clear definitions on IP phones, only about SCCP towards gateway, so you're probably right. Cheers, Johnny Witt Netværksspecialist CSIS - Combined Services Integrated Solutions Majsmarken 9 DK2680 Solrød Strand Tlf : 56 13 11 83 Mobil 28 66 28 48 [EMAIL PROTECTED] -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Jeremy McNamara Sendt: 28. juni 2003 23:20 Til: [EMAIL PROTECTED] Emne: Re: SV: [Asterisk-Users] Newbie questions. Johnny Witt wrote: CallManager).am I right in saying that Cisco phones using Skinny will not work with asterisk? Is it ever likely too? Cisco own Skinny Protocol is not supported directly. But Cisco SE always told me that Skinny is a subset of H.323. But I would'nt count on it to be functioning correctly. Skinny is not even closely related to H.323 or SIP or MGCP or anything else that is out there.. other than using RTP for the audio transport. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users BEGIN:VCARD VERSION:2.1 N:Witt;Johnny FN:Johnny Witt ([EMAIL PROTECTED]) ORG:CSIS - Combined Services Integrated Solutions TITLE:Netværks Specialist TEL;WORK;VOICE:+45 56 13 11 83 TEL;CELL;VOICE:+45 28 66 28 48 ADR;WORK:;;Majsmarken 9;DK-2680;;;Danmark LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Majsmarken 9=0D=0ADK-2680=0D=0ADanmark URL;WORK:http://www.csis.dk EMAIL;PREF;INTERNET:[EMAIL PROTECTED] REV:20030412T003448Z END:VCARD
Re: SV: [Asterisk-Users] Newbie questions.....
Johnny Witt wrote: CallManager).am I right in saying that Cisco phones using Skinny will not work with asterisk? Is it ever likely too? Cisco own Skinny Protocol is not supported directly. But Cisco SE always told me that Skinny is a subset of H.323. But I would'nt count on it to be functioning correctly. Skinny is not even closely related to H.323 or SIP or MGCP or anything else that is out there.. other than using RTP for the audio transport. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Newbie questions.....
Check to see if you can get a IOS code leverl that supports SIP on the 6500. then maybe you can use your E1 card directly. you can also get a SIP version of the code for the 7960's etc Dave [EMAIL PROTECTED] 6/28/2003 2:56:12 PM Hi Chris I've done a lot of things with Cisco AVVID solutions in the past. CallManager).am I right in saying that Cisco phones using Skinny will not work with asterisk? Is it ever likely too? Cisco own Skinny Protocol is not supported directly. But Cisco SE always told me that Skinny is a subset of H.323. But I would'nt count on it to be functioning correctly. * We have an 8 port E1 card in a Cisco 6509 which takes our main phone trunk to the public network. Can we connect an Asterisk PBX server with an E1 card to this? If so, could we then connect the Asterisk PBX to the callmanager? (Perhaps with another extension range).and if so, how? Yes, it's possible. I've done it to trombone calls through a Operator system (Trio) and to interconnect with both Ericsson, Nortel PBXs and iPBXs. On CM you set it up as a trunk line, create a route map which forward all calls to that specific E1 / T1 port on the 6608, which are connected to the Asterisk Pri port. On the Asterisk you do the same. Beware that top-down, bottom-up is opposite on the 2 system. Then you should be able to get it working. There could be a few minor adjustments which needs to be done on the CCM - but trying to recall the configuration Page from memory is'nt that easy :-) But it was actually very few steps in getting it to interconnect with Nortel Meridians and Ericsson MD-110 through PRI trunks. But you actually dont need it. You can specify H.323 trunks/endpoints/zones in CCM where you can specify The IP address and numbering plan of the Asterisk system. This way would be better because you have to apply codecs (involve DSP) four times using the 6608 board. This would probably give very long delays more than 300 ms depending on the payload. * .or could we connect Asterisk to the 6509 over IP and so make it part of the main phone system? Already answered - But asterisk can not directly communicate with 6608 on the 6509. All out going calls using the 6608 has to go through the Callmanager. * We have a Nortel Meridian PBX on our other campus which is connected to our IP telephony system via an E1 link to a Cisco Vg200 voice H.323 gateway...would there be any way to point asterisk at this gateway and make it part of our main phone system that way? again if so how? Hm This one is a bit difficult to answer. As I recall vg200 is a limited voice gateway which only can be used by Cisco Callmanager. Which makes your scenario to be : Meridian - PRI trunk - VG200 - Router WAN/LAN - Router - Cisco Callmanger - 6608 -PSTN Am I correct ? The you can replace the Meridian and VG200 with Asterisk or any of these with Asterisk. Or you can add it separately on the LAN on same location as the VG200, and specify asterisk as a H.323 Trunk/endpoint on the Cisco callmanager. In general all solutions depends on the equipments ability to apply to the standards. If some of it does'nt you would loose functionality. But I would really think twice before connecting too many different iPBX Components in one VoIP solution. I can understand if you're going to test the asterisk project as an alternative to either Cisco AVVID solution or the PBX's in your campus network. But a working enviroment I would really be very careful. It would be almost impossible to troubleshoot. It is very important to know your current setup in details, and what you hope to gain by using Asterisk. If my English is to bad then I apologize ... :-) Cheers, Johnny Witt Netværksspecialist CSIS - Combined Services Integrated Solutions Majsmarken 9 * DK2680 Solrød Strand * Denmark Tlf : (+45) 56 13 11 83 * Mobil (+45) 28 66 28 48 * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Newbie questions.....
Hi Chris I've done a lot of things with Cisco AVVID solutions in the past. CallManager).am I right in saying that Cisco phones using Skinny will not work with asterisk? Is it ever likely too? Cisco own Skinny Protocol is not supported directly. But Cisco SE always told me that Skinny is a subset of H.323. But I would'nt count on it to be functioning correctly. * We have an 8 port E1 card in a Cisco 6509 which takes our main phone trunk to the public network. Can we connect an Asterisk PBX server with an E1 card to this? If so, could we then connect the Asterisk PBX to the callmanager? (Perhaps with another extension range).and if so, how? Yes, it's possible. I've done it to trombone calls through a Operator system (Trio) and to interconnect with both Ericsson, Nortel PBXs and iPBXs. On CM you set it up as a trunk line, create a route map which forward all calls to that specific E1 / T1 port on the 6608, which are connected to the Asterisk Pri port. On the Asterisk you do the same. Beware that top-down, bottom-up is opposite on the 2 system. Then you should be able to get it working. There could be a few minor adjustments which needs to be done on the CCM - but trying to recall the configuration Page from memory is'nt that easy :-) But it was actually very few steps in getting it to interconnect with Nortel Meridians and Ericsson MD-110 through PRI trunks. But you actually dont need it. You can specify H.323 trunks/endpoints/zones in CCM where you can specify The IP address and numbering plan of the Asterisk system. This way would be better because you have to apply codecs (involve DSP) four times using the 6608 board. This would probably give very long delays more than 300 ms depending on the payload. * .or could we connect Asterisk to the 6509 over IP and so make it part of the main phone system? Already answered - But asterisk can not directly communicate with 6608 on the 6509. All out going calls using the 6608 has to go through the Callmanager. * We have a Nortel Meridian PBX on our other campus which is connected to our IP telephony system via an E1 link to a Cisco Vg200 voice H.323 gateway...would there be any way to point asterisk at this gateway and make it part of our main phone system that way? again if so how? Hm This one is a bit difficult to answer. As I recall vg200 is a limited voice gateway which only can be used by Cisco Callmanager. Which makes your scenario to be : Meridian - PRI trunk - VG200 - Router WAN/LAN - Router - Cisco Callmanger - 6608 -PSTN Am I correct ? The you can replace the Meridian and VG200 with Asterisk or any of these with Asterisk. Or you can add it separately on the LAN on same location as the VG200, and specify asterisk as a H.323 Trunk/endpoint on the Cisco callmanager. In general all solutions depends on the equipments ability to apply to the standards. If some of it does'nt you would loose functionality. But I would really think twice before connecting too many different iPBX Components in one VoIP solution. I can understand if you're going to test the asterisk project as an alternative to either Cisco AVVID solution or the PBX's in your campus network. But a working enviroment I would really be very careful. It would be almost impossible to troubleshoot. It is very important to know your current setup in details, and what you hope to gain by using Asterisk. If my English is to bad then I apologize ... :-) Cheers, Johnny Witt Netværksspecialist CSIS - Combined Services Integrated Solutions Majsmarken 9 DK2680 Solrød Strand Denmark Tlf : (+45) 56 13 11 83 Mobil (+45) 28 66 28 48 [EMAIL PROTECTED] BEGIN:VCARD VERSION:2.1 N:Witt;Johnny FN:Johnny Witt ([EMAIL PROTECTED]) ORG:CSIS - Combined Services Integrated Solutions TITLE:Netværks Specialist TEL;WORK;VOICE:+45 56 13 11 83 TEL;CELL;VOICE:+45 28 66 28 48 ADR;WORK:;;Majsmarken 9;DK-2680;;;Danmark LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Majsmarken 9=0D=0ADK-2680=0D=0ADanmark URL;WORK:http://www.csis.dk EMAIL;PREF;INTERNET:[EMAIL PROTECTED] REV:20030412T003448Z END:VCARD
[Asterisk-Users] Newbie questions
Hi.I am new to this software, and I want to implementa client (SIP or IAX) with PHP or at least to pass the main functions (connection,call, transfer, hangup, call id etc) to a CRM. Does anyone know if I could achive a project like that with AGI ? Any example using AGI with PHP ? Do I have all the functionality with AGI ? What about call id ? What is depend on ? (As I know * does not support SS7, so is there a problem for the call id ?) Thanks in advance. Konstantinos.
[Asterisk-Users] Newbie questions.....
Hi. I have just successfully setup Asterisk with 2 Cisco 7940 phones (converted for SIP) and a SIP softphone on a W2K box.and it all seems to work very well.to those who wrote this software, it is really cool. Anyway, I am new to this software, and I have a lot of questions which I am hoping someone on the mailing list might be able to answer for me.I am basically trying to get an idea of how/what I can do with Asterisk that I am already doing with our existing phone system Sorry about the length of the mailthe docs don't seem to cover some of the topics below. Thanx in advance for any help. Chris. * We currently have a Cisco IP telephony system (using their CallManager).am I right in saying that Cisco phones using Skinny will not work with asterisk? Is it ever likely too? * When we connect and power on a Cisco 79X0 phone for the first time, it automatically registers with the CallManager and is assigned a temporary number. We then do into the CallManager admin interface and assign it to its owner, give it its permanent number etc. Among the things which happen are the TFTP files for the phone (eg: SEPmac_address.cnf) get created as part of the automatic registration. When I converted the phones to SIP, I had to manually create config files for each phone (SIPmac_address.cnf etc.).is there any way I can have this happen automatically? * We have an 8 port E1 card in a Cisco 6509 which takes our main phone trunk to the public network. Can we connect an Asterisk PBX server with an E1 card to this? If so, could we then connect the Asterisk PBX to the callmanager? (Perhaps with another extension range).and if so, how? * .or could we connect Asterisk to the 6509 over IP and so make it part of the main phone system? * We have a Nortel Meridian PBX on our other campus which is connected to our IP telephony system via an E1 link to a Cisco Vg200 voice H.323 gateway...would there be any way to point asterisk at this gateway and make it part of our main phone system that way? again if so how? _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie questions.....
Hi. I have just successfully setup Asterisk with 2 Cisco 7940 phones (converted for SIP) and a SIP softphone on a W2K box.and it all seems to work very well.to those who wrote this software, it is really cool. Anyway, I am new to this software, and I have a lot of questions which I am hoping someone on the mailing list might be able to answer for me.I am basically trying to get an idea of how/what I can do with Asterisk that I am already doing with our existing phone system Sorry about the length of the mailthe docs don't seem to cover some of the topics below. Thanx in advance for any help. Chris. * We currently have a Cisco IP telephony system (using their CallManager).am I right in saying that Cisco phones using Skinny will not work with asterisk? Is it ever likely too? Skinny is not included as a channel in Asterisk at this time. There are reports of a Skinny channel well into development - see http://www.sf.net/projects/sccp and we await further testing. * When we connect and power on a Cisco 79X0 phone for the first time, it automatically registers with the CallManager and is assigned a temporary number. We then do into the CallManager admin interface and assign it to its owner, give it its permanent number etc. Among the things which happen are the TFTP files for the phone (eg: SEPmac_address.cnf) get created as part of the automatic registration. When I converted the phones to SIP, I had to manually create config files for each phone (SIPmac_address.cnf etc.).is there any way I can have this happen automatically? Yes and no. You still will have to create a file called SIPmac address.cnf which contains the extensions that you expect the phone to use. However, if you have an RFC compliant DHCP server, you should be able to make everything happen automatically except for the generation of that extension. There are almost no hooks between any of the very sophisticated Cisco configuration files and Asterisk; they are _separate_ systems. Asterisk simply deals with SIP devices and their SIP transactions - Asterisk does _not_ configure SIP devices, and Asterisk is not Cisco-specific in any treatment of SIP transactions. I seem to recall that there is a Cisco 79xx administration tool in the http://www.vovida.org/ pages somewhere. * We have an 8 port E1 card in a Cisco 6509 which takes our main phone trunk to the public network. Can we connect an Asterisk PBX server with an E1 card to this? If so, could we then connect the Asterisk PBX to the callmanager? (Perhaps with another extension range).and if so, how? Yes. The manual should explain further details. * .or could we connect Asterisk to the 6509 over IP and so make it part of the main phone system? I don't know. Does the 6509 talk SIP? * We have a Nortel Meridian PBX on our other campus which is connected to our IP telephony system via an E1 link to a Cisco Vg200 voice H.323 gateway...would there be any way to point asterisk at this gateway and make it part of our main phone system that way? again if so how? Yes. That's too complex to explain adequately here, but you should try setting it up to answer the question yourself. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie questions.....
* .or could we connect Asterisk to the 6509 over IP and so make it part of the main phone system? I don't know. Does the 6509 talk SIP? It doesn't appear to. I would love to be wrong. It does support MGCP, though. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie questions.....
Thanx for the infounfortunately, I think we would need an Communications Media Modulewhich we don't have Chris. From: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Newbie questions. Date: Fri, 20 Jun 2003 15:21:53 -0500 * .or could we connect Asterisk to the 6509 over IP and so make it part of the main phone system? I don't know. Does the 6509 talk SIP? It doesn't appear to. I would love to be wrong. It does support MGCP, though. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ The new MSN 8: smart spam protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie questions.....
Hi Thanx for the info.sorry to hassle you, but I have follow on questions below. I seem to recall that there is a Cisco 79xx administration tool in the http://www.vovida.org/ pages somewhere. Had a look at this.this tool will certainly make managing Cisco SIP phones easierthanx Yes. The manual should explain further details. I am not a voice comms expert and our Cisco IP Telephony system was installed by an outside company Are you referring to the Asterisk manual here? Just looking at the manual, it seems to me that I could use a Zaptel E1 card and configure the zapata.conf file appropriately..am I correct? If so, I would just need to learn how to configure an E1 port on the 6509.and configure the CallManager to know where Asterisk is in the number range * We have a Nortel Meridian PBX on our other campus which is connected to our IP telephony system via an E1 link to a Cisco Vg200 voice H.323 gateway...would there be any way to point asterisk at this gateway and make it part of our main phone system that way? again if so how? Yes. That's too complex to explain adequately here, but you should try setting it up to answer the question yourself. Again, I am at a bit of a loss since I am not a voice comms expertwhere would I begin in Asterisk...is there a H.323 channel (as there is for SIP) in Asterisk?...or is this a silly question.? I don't see any mention of H.323 in the conf files and lastly, one further question.I got voicemail working, but the red light on the Cisco 79X0 phone doesn't light when a voice mail is waiting.is there a way to enable this? Thanx very much for the info, and thanx in advance for any further info Chris. _ MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*. http://join.msn.com/?page=features/virus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
