Re: [asterisk-users] asterisk setup w/ voIP phones
Which grandstream phone should I buy, this is going to be for small office for testing purposes. I am on a budget, hoping to find someone here who has some used to sell or point me in the direction of a seller. Hi Mike, If you're set on the Grandstreams, and it's just for testing the BT200 or BT201 will suit you fine. If you want to test more features, the GXP2000 is relatively cheap. Have you looked at Aastra? They offer some quality phones in the same range as the GrandStreams. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk setup w/ voIP phones
On Wed, 12 Nov 2008, Mike wrote: Hi All, I have setup asterisk 1.4.22; so far everything good. Except, I am still searching for voIP phones. Which grandstream phone should I buy, this is going to be for small office for testing purposes. I am on a budget, hoping to find someone here who has some used to sell or point me in the direction of a seller. If you're on a budget and you're going to buy Grandstream phones, then you need the Budget phone - er, Budgephone 200 - BT200. Small phone, numbers only display, single line phone, but handles transfers 3-way calling if required. Has a built-in 10/100 switch. (The BT101102 only has a 10Mb hub, but I think they've been end of lined now) The web interface is easy to use too. I've deployed quite a few of these on customer sites without any real issues so-far. (The display will look funny if you send it alphanumeric caller ID though - as the phone actually tries to represent characters on the 7-segment display!) If your budget will afford it, look at the GXP1200s though - smallish display, but it's alpha-numeric, so you get to see proper caller ID. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk setup w/ voIP phones
On Thu, Nov 13, 2008 at 9:56 AM, Matt Gibson [EMAIL PROTECTED]wrote: Which grandstream phone should I buy, this is going to be for small office for testing purposes. I am on a budget, hoping to find someone here who has some used to sell or point me in the direction of a seller. Hi Mike, If you're set on the Grandstreams, and it's just for testing the BT200 or BT201 will suit you fine. If you want to test more features, the GXP2000 is relatively cheap. Have you looked at Aastra? They offer some quality phones in the same range as the GrandStreams. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com I have found Grandstream to make many too many issues behind NAT. I was told to upgrade the phone to the latest firmware which only made it worse. After an upgrade to the latest firmware the phone is as some one on the biz list once said A great door stopper. ___ -- 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] asterisk setup w/ voIP phones
Hi All, I have setup asterisk 1.4.22; so far everything good. Except, I am still searching for voIP phones. Which grandstream phone should I buy, this is going to be for small office for testing purposes. I am on a budget, hoping to find someone here who has some used to sell or point me in the direction of a seller. I am in the US. thanks, Mike ___ -- 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] asterisk setup
Hi folks, Am new to asterisk pbx systems. I am trying to figure out what to do, I'll list and folks feel free to give feedback and advice. MAIN purpose for usage: 1.exposure to setup an asterisk box 2.get home phone service via VOIP/internet connection. tasks so far -- 1. setup and install asterisk (1.4.x) -- DONE -currently configuring sip.conf and extenstion.conf files. -using soft-phone till I find a cheap voip phone(recommendations ??) 2. trying to get a DID number -alot of search results, not sure which are reliable, most are charging per month usage and some are charging setup charges -also looking for DID provider which will interface with asterisk pbx (sip and iax protocols, is that ok? ) -Am I missing anything else here? -Are there any providers folks would recommend? Looking for affordable and reliable) thanks in advance, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk setup
On Mon, 20 Oct 2008, Mike wrote: Hi folks, Am new to asterisk pbx systems. I am trying to figure out what to do, I'll list and folks feel free to give feedback and advice. MAIN purpose for usage: 1.exposure to setup an asterisk box 2.get home phone service via VOIP/internet connection. tasks so far -- 1. setup and install asterisk (1.4.x) -- DONE Heh... You'll never be done! Read the starfish book cover to cover (free online/PDF, or buy a paper copy). Read the Voip-wiki. -currently configuring sip.conf and extenstion.conf files. -using soft-phone till I find a cheap voip phone(recommendations ??) If you want a cheap VoIP phone look for a Grandstream - BT200 is cheap and functional. GXP1200 will give you something with a small display (BT200 is numeric only) GXP2000 has a bigger display and more buttons. However holy wars have been fought over Grandstream - some people like them, some hate them. You get what you pay for and I've deployed 100's of the damn things. 2. trying to get a DID number -alot of search results, not sure which are reliable, most are charging per month usage and some are charging setup charges -also looking for DID provider which will interface with asterisk pbx (sip and iax protocols, is that ok? ) -Am I missing anything else here? -Are there any providers folks would recommend? Looking for affordable and reliable) What country are you in? This is a truly global marketplace and mailing list. We have people from the UK, Ireland, Oztrailia, New Zealand, Bolivia, Russia, China, India, Argentina, etc. All over the world, really. Saying what country you need the DID/DDI in will narrow it down somewhat. Well done so-far and Good luck! Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk setup
On Mon, Oct 20, 2008 at 8:42 AM, Mike [EMAIL PROTECTED] wrote: Am new to asterisk pbx systems. Hi Mike, Welcome to the wonderful world of asterisk! I am trying to figure out what to do, I'll list and folks feel free to give feedback and advice. You don't mention if you have read one of the several books out on asterisk or whether you are familiar with sources like http://voip-info.org. If you haven't already, look into the O'Reilly book: http://oreilly.com/catalog/9780596510480/ -- 1. setup and install asterisk (1.4.x) -- DONE -currently configuring sip.conf and extenstion.conf files. -using soft-phone till I find a cheap voip phone(recommendations ??) Define cheap. I believe for home use there are entry-level phones by known manufacturers for well under $200, but that's 10 times what a cheap cordless costs, right? You can get a Grandstream for under $100, but people will already be groaning at the mention of these. You can also purchase FXS/FXO cards to connect regular phones to your asterisk box, among these the Digium TDM400 can accept four modules. Pricey perhaps, but this avoids buying phones if you already have some. Ypour budget will depend on how serious your needs are. If you just want an answering machine to experiment with, cheap iup phones maybe from Ebay will do it for you. (Ebay is the Devil, I do not recommend it personally) 2. trying to get a DID number -alot of search results, not sure which are reliable, most are charging per month usage and some are charging setup charges -also looking for DID provider which will interface with asterisk pbx (sip and iax protocols, is that ok? ) -Am I missing anything else here? -Are there any providers folks would recommend? Looking for affordable and reliable) There are many SIP providers, but among those that are part of or support the asterisk community, Junction Networks, Teliax, VoicePulse Connect loom large. I have had accounts with all of those and find them to be reasonable in price and service. This stuff is very subjective, though. Maybe look through the mialing list for references to those names and other providers. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk setup
What country are you in? This is a truly global marketplace and mailing list. We have people from the UK, Ireland, Oztrailia, New Zealand, Bolivia, Russia, China, India, Argentina, etc. All over the world, really. Saying what country you need the DID/DDI in will narrow it down somewhat. I am in the US. Hope someone here can help. Gordon, thanks for your email. ___ -- 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] asterisk setup for church / conference call / speaker system integration
We have several people in our church that recently became disabled. I am thinking of setting up an asterisk server and several phone lined so that they can call in to church during services to listen to the service. The phone lines at church are also used by our private school during the week. So I am thinking that I need to setup a schedule so that the lines used for the service will automatically go to the conference during normal service times and ring the phones the rest of the time. We have 2 analog lines currently, DSL is available, and I'm checking now to see if we can get a voip provider that will provide 3 - 10 local numbers. to connect to the speaker system I either need to trigger a ring on a analog line to the phone interface on our speaker system, it picks up on the first ring, or we can manully push a button that picks up the line. If we do the second we would have to have something in asterisk connect it to the conference when it picks up. the next consideration is that sometimes we might want to connect out to a conference call from another church. so we need to be able to dial out from one of the phones and then somehow trigger the speaker systems incoming side to pickup. ( I just thought of it now - I suppose a call transfer to that line would be all thats needed for that) ___ --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] asterisk setup for church / conference call / speaker system integration
On 5/17/07, Tim Litwiller [EMAIL PROTECTED] wrote: We have several people in our church that recently became disabled. I am thinking of setting up an asterisk server and several phone lined so that they can call in to church during services to listen to the service. If it were me, I would: 1. create a conference room 2. create a .call file that dials into the speaker system 3. create .call files to dial the participants, muting them This can obviously be scripted very easily. A simple cron job copying the .call files should do nicely. If the disabled persons wish to no longer participate, you can simply delete that .call file; conversely, if you need to add someone, you can just create another one. Then you won't have to worry about incoming phone numbers and coordinating with the private school, you can bring it in when you need it, etc. ___ --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] asterisk setup for church / conference call / speaker system integration
David Gomillion wrote: On 5/17/07, *Tim Litwiller* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We have several people in our church that recently became disabled. I am thinking of setting up an asterisk server and several phone lined so that they can call in to church during services to listen to the service. If it were me, I would: 1. create a conference room 2. create a .call file that dials into the speaker system 3. create .call files to dial the participants, muting them This can obviously be scripted very easily. A simple cron job copying the .call files should do nicely. If the disabled persons wish to no longer participate, you can simply delete that .call file; conversely, if you need to add someone, you can just create another one. Then you won't have to worry about incoming phone numbers and coordinating with the private school, you can bring it in when you need it, etc. That does simplify it a bit - but is probably not flexible enough. I don't want to have to have someone call and request to be added to the conference if they are sick sunday morning. We will publish the number for anyone in the congregation to use as they need it. For those that always listen at home tho this would work fine. Oh and we will also want to record the services so that if someone wants a copy or to listen later they can call in to listen or we can burn then a copy. So I'll need to program a recording menu system that is somewhat automated and lists the last few weeks of services by date or something. We are currently using freeconferencecall.com but it is a long distance call from church and from all of our members. ___ --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] asterisk setup for church / conference call / speaker system integration
Tim Litwiller wrote: David Gomillion wrote: On 5/17/07, *Tim Litwiller* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We have several people in our church that recently became disabled. I am thinking of setting up an asterisk server and several phone lined so that they can call in to church during services to listen to the service. If it were me, I would: 1. create a conference room 2. create a .call file that dials into the speaker system 3. create .call files to dial the participants, muting them This can obviously be scripted very easily. A simple cron job copying the .call files should do nicely. If the disabled persons wish to no longer participate, you can simply delete that .call file; conversely, if you need to add someone, you can just create another one. Then you won't have to worry about incoming phone numbers and coordinating with the private school, you can bring it in when you need it, etc. That does simplify it a bit - but is probably not flexible enough. I don't want to have to have someone call and request to be added to the conference if they are sick sunday morning. We will publish the number for anyone in the congregation to use as they need it. For those that always listen at home tho this would work fine. Oh and we will also want to record the services so that if someone wants a copy or to listen later they can call in to listen or we can burn then a copy. So I'll need to program a recording menu system that is somewhat automated and lists the last few weeks of services by date or something. We are currently using freeconferencecall.com but it is a long distance call from church and from all of our members. Use GotoIfTime to divert incoming calls, only during regular service times, to a voice menu with a choice of listen live, or make a call to extension. Use David's call file suggestion to connect the speakers to the conference and record (Monitor) that channel. Have an extension that callers can dial at anytime for a voice menu with a choice of recordings. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.com ___ --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] asterisk setup for church / conference call / speaker system integration
Tim Litwiller wrote: Oh and we will also want to record the services so that if someone wants a copy or to listen later they can call in to listen or we can burn then a copy. So I'll need to program a recording menu system that is somewhat automated and lists the last few weeks of services by date or something. This part sounds like a podcast to me. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] asterisk setup for church / conference call/ speaker system integration
Worth looking at using Talkshoe for this application maybe? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Prior Sent: Thursday, 17 May 2007 12:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk setup for church / conference call/ speaker system integration Tim Litwiller wrote: Oh and we will also want to record the services so that if someone wants a copy or to listen later they can call in to listen or we can burn then a copy. So I'll need to program a recording menu system that is somewhat automated and lists the last few weeks of services by date or something. This part sounds like a podcast to me. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] asterisk setup for church / conference call / speaker system integration
Drew Gibson wrote: Tim Litwiller wrote: David Gomillion wrote: On 5/17/07, *Tim Litwiller* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We have several people in our church that recently became disabled. I am thinking of setting up an asterisk server and several phone lined so that they can call in to church during services to listen to the service. If it were me, I would: 1. create a conference room 2. create a .call file that dials into the speaker system 3. create .call files to dial the participants, muting them This can obviously be scripted very easily. A simple cron job copying the .call files should do nicely. If the disabled persons wish to no longer participate, you can simply delete that .call file; conversely, if you need to add someone, you can just create another one. Then you won't have to worry about incoming phone numbers and coordinating with the private school, you can bring it in when you need it, etc. That does simplify it a bit - but is probably not flexible enough. I don't want to have to have someone call and request to be added to the conference if they are sick sunday morning. We will publish the number for anyone in the congregation to use as they need it. For those that always listen at home tho this would work fine. Oh and we will also want to record the services so that if someone wants a copy or to listen later they can call in to listen or we can burn then a copy. So I'll need to program a recording menu system that is somewhat automated and lists the last few weeks of services by date or something. We are currently using freeconferencecall.com but it is a long distance call from church and from all of our members. Use GotoIfTime to divert incoming calls, only during regular service times, to a voice menu with a choice of listen live, or make a call to extension. Use David's call file suggestion to connect the speakers to the conference and record (Monitor) that channel. Have an extension that callers can dial at anytime for a voice menu with a choice of recordings. regards, Drew Good suggestions everyone, thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk setup for church / conference call/ speaker system integration
Dean Collins wrote: Worth looking at using Talkshoe for this application maybe? My personal podcast authoring tool is vi (and you thought it was just a website authoring tool...) so I don't have any user friendly recommendations, but there's got to be a lot out there. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Setup Question -- Please Help
I've had success using oh323 to create a trunk between a CCM3.2 and asterisk.. I wouldn't put any analog devices in your asterisk box if you already have CM. You could also move completely to Asterisk and flash your 79xx's into SIP mode and turn your Cisco boxes off. BTW, you could also have asterisk just do something like: exten = _9XX,1,Dial(OH323/[EMAIL PROTECTED]) So you don't have to do fancy 78 stuff to get an outside line. I'm also playing around with something like: exten = 1234,1,Dial(SIP/SomeUser) exten = 1235,1,Dial(SIP/SomeOtherUser) ; we didn't find any users here .. let's try Cisco exten = _,1,Dial(OH323/[EMAIL PROTECTED]) On this Cisco side, when I move people to Asterisk, I'll just hard code a dial plan to the h323 gate on my asterisk box. Then, I can flash hardphone users one at a time and get them using asterisk as their call server at my leisure (or not move people at all). It's mostly working now, I'm just having problems with including contexts and precedence -- I haven't quite figured it all out yet. I hope that's a source of inspiration :) On 1/25/06, Goran Donev [EMAIL PROTECTED] wrote: I have a question on Asterisk and whether it will work with the following design. Install ASTERISK on the external side of the Network. Purchase an AudioCodes 4/8 port Analog Fx0 gateway. So far everything seems straight forward. Here is the twist. The company currently has Cisco Call Manager 3.3 which does not support SIP Trunking. But it does have a VG248. I would like to place 4 lines through the Cisco Call Manager. I want to set up a dial plan where 7 would grab the fx0 line for internal and the users would be able to place internal calls through the Cisco Call Manager. I envision people dialing 7 (4 digit extension.) This would call internally. I then envision setting up a calling plan where 7 would grab the trunk and 8 would grab an outside line from the Cisco Call Manager and then dial the 10 digit telephone number. 78xx. This would allow them to place external calls through the call manager. Is this something that would be feasible? Since the company is not looking to invest a lot in upgrading the Cisco yet they want to allow external sales reps to work from home. Would there be a way through Asterisk where I can then program the FX0 extension coming in from the Cisco Call Manager to ring into the Audiocodes and be dialed directly to an extension in the Asterisk server? Example - 1300---200 on the Asterisk. This would allow people calling the company to directly dial their sales people and be forwarded to the extension attached to the audiocodes. If this is feasible please let me know as I would like to propose this solution to the company. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Asterisk Setup Question -- Please Help
I have a question on Asterisk and whether it will work with the following design. Install ASTERISK on the external side of the Network. Purchase an AudioCodes 4/8 port Analog Fx0 gateway. So far everything seems straight forward. Here is the twist. The company currently has Cisco Call Manager 3.3 which does not support SIP Trunking. But it does have a VG248. I would like to place 4 lines through the Cisco Call Manager. I want to set up a dial plan where 7 would grab the fx0 line for internal and the users would be able to place internal calls through the Cisco Call Manager. I envision people dialing 7 (4 digit extension.) This would call internally. I then envision setting up a calling plan where 7 would grab the trunk and 8 would grab an outside line from the Cisco Call Manager and then dial the 10 digit telephone number. 78xx. This would allow them to place external calls through the call manager. Is this something that would be feasible? Since the company is not looking to invest a lot in upgrading the Cisco yet they want to allow external sales reps to work from home. Would there be a way through Asterisk where I can then program the FX0 extension coming in from the Cisco Call Manager to ring into the Audiocodes and be dialed directly to an extension in the Asterisk server? Example - 1300---200 on the Asterisk. This would allow people calling the company to directly dial their sales people and be forwarded to the extension attached to the audiocodes. If this is feasible please let me know as I would like to propose this solution to the company. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk setup
Hi all, We are in the process of asterisk setup in our office. The setup should contain analog phones connected through USB or PCI slot supported by softphone or device driver to the workstation running linux.Further all workstations with analog phones should connected to asterisk server in LAN with normal cables.We don,t have telephone cables. Is this type of setup is possible.If so what type of devices are need to be bought. We came to know about S100U .Is this device is reliable to by or it satisfy our setup procedure. If anyone will help would be appreciated. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] asterisk setup
Hi, I just joined the list, anyways i am trying to setup an @home box with a x100p card and so far i can't even get the box to pickup the incoming call and in the amp management under the section send calls from PSTN too page all the radio buttons are blank and i want to use the digital receptionist, also when i try to setup digital receptionist via uploading wav file and save, it says file uploaded successfully but when i go back in there nothing is in the digital receptionist page. Kurt Fankhauser WaveLinc www.wavelinc.com 114 S. Walnut St. Bucyrus, OH 44820 419-562-6405 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.8 - Release Date: 2/14/2005 ___ 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] asterisk setup
Kurt, Did you follow the instructions and remove the semi colon in zapata.conf before the channel =1 what happens on the console cli when you make a call. What type of handsets are you using? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt Fankhauser Sent: Saturday, February 19, 2005 11:30 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk setup Hi, I just joined the list, anyways i am trying to setup an @home box with a x100p card and so far i can't even get the box to pickup the incoming call and in the amp management under the section send calls from PSTN too page all the radio buttons are blank and i want to use the digital receptionist, also when i try to setup digital receptionist via uploading wav file and save, it says file uploaded successfully but when i go back in there nothing is in the digital receptionist page. Kurt Fankhauser WaveLinc www.wavelinc.com 114 S. Walnut St. Bucyrus, OH 44820 419-562-6405 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.8 - Release Date: 2/14/2005 ___ 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] Asterisk Setup Documentation
Title: Message Hello all: Can anyone help mewith finding the best locations for getting setup and other documentation for *. Thank you. Phil Menico www.xtend.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] Asterisk Setup Documentation
Title: Message Phil, http://www.voip-info.org/wiki-Asterisk is a good place to start, and will point you to most resources. Paul From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Menico Sent: 10 January 2005 15:20 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Asterisk Setup Documentation Hello all: Can anyone help mewith finding the best locations for getting setup and other documentation for *. Thank you. Phil Menico www.xtend.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] Asterisk Setup Documentation
Title: Message An alternative is http://www.asteriskdocs.org/ . Its only getting going but there is some getting started stuff. The challenge of the wiki is that its relatively fragmented and assumes a level of knowledge. I found the wiki much more useful when I had a running system and wanted to find expert ideas on how to achieve a specific objective. Regards Bill Seddon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Brock Sent: January 10, 2005 3:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk Setup Documentation Phil, http://www.voip-info.org/wiki-Asterisk is a good place to start, and will point you to most resources. Paul From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Menico Sent: 10 January 2005 15:20 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Asterisk Setup Documentation Hello all: Can anyone help mewith finding the best locations for getting setup and other documentation for *. Thank you. Phil Menico www.xtend.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] Asterisk Setup Documentation
The Guide to Asterisk at http://asterisk.automated.it/ is pretty thorough/detailed for those not too familiar with linux/telephony. Quoting Bill Seddon [EMAIL PROTECTED]: An alternative is http://www.asteriskdocs.org/ . It's only getting going but there is some getting started stuff. The challenge of the wiki is that its relatively fragmented and assumes a level of knowledge. I found the wiki much more useful when I had a running system and wanted to find expert ideas on how to achieve a specific objective. ___ 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] Asterisk Setup Documentation
http://www.asteriskdocs.org is a work in progress document project for Asterisk between that and the wiki you should be ok. If that isn't enough there is plenty of posts in the archives of this list and odds are someone else has already had the issue you are faced with. ___ 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] Asterisk Setup and configuration help
Hi, I have a client that wants to set up an Asterisk-based VOIP solution. While we can easily handle most of their IT needs, we've never really used Asterisk. We're looking for an experienced Asterisk tech to give us some help getting the box up and running, configured appropriately. Once up and running, I imagine that this person would give us a little training and provide third-level support as required. If interested, please email me off list - make sure to put Asterisk in the subject line so I don't mistake it for spam. Regards Mike ___ 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] Asterisk setup.-
Hi, I recently received my development kit with 1 x100p and one tdm400p (1) fxs port. I installed everything from the digium disk that i received with my kit, however, i dont; know what to do next. I would like to be able to call through the internet using xten (pc2phone) and terminate the call in my gateway. anyone has a standard setup ? thanks, Francisco - Original Message - From: Steven E. Frazier [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 06, 2004 12:30 AM Subject: [Asterisk-Users] Adding another X100P after X100P and TDM400P is already configured History: 1. Added X100P to my system 2. Added TDM400P (2 port) Worked fine so far 3. Now I want to add an additional X100P Is the following configs files ok and is there any issue with adding the X100P (channel 4) after my 2 analog FXS channels? Thanks. Steve Here is my /etc/zaptel.conf fxsks=1,4 fxols=2-3 loadzone = us defaultzone = us Here is my /etc/asterisk/zapata.conf ; Zapata telephony interface sample configuration file ; [channels] ; ; X100P plugged into PSTN ; X100P # 1 context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel = 1 ; ; ; ; TDM200B Port #1 plugged into analog Phone ; ; context=toll-access signalling=fxo_ls callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes callerid=Livingroom 2201 mailbox=2201 channel = 2 ; ; TDM200B Port #2 ; ; context=toll-access signalling=fxo_ls callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes callerid=Kitchen 2202 mailbox=2202 channel = 3 ; X100P # 2 context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel = 4 ___ 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
