[Asterisk-Users] Hi...Please help me
Hi Friends, Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone. Now, I want to make calls to US using VoIP Asterisk. I have registered with Vebtel (VoIP IP Telephony Service provider). They had given me one VoIP Modem called "Voice Finder AP 200" and the below values: Inbound Number: 123456789 Public IP Number: 55.23.789.145 Password: xyz (These values are dummy values) Currently we are making US calls using VoIP provided by "Vebtel". Now, I want to make US calls using this Vebtel service from Asterisk. How can I do this? I am unable to understand where to give above mentioned values? What configuration files I should use to implement this using the Vebtel SIP provider? Do I need to provide any more values to implement this using Asterisk from Vebtel? Waiting for your quick response. Thank you. Regards, Chandra. Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___ --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] Hi...Please help me
Chandra, In all honesty if they are proprietary and you want to use them you will need a FXO card. Alternatively there are a few good termination providers out there that are inexpensive. The top 3 most inexpensive that come to mind are: Plainvoip - http://www.plainvoip.com Domestic starting at 1.1c VoipJet - http://www.voipjet.com Domestic starting at 1.3c NuFone - http://www.nufone.net Domestic starting at 2c (I believe) Anyone of these providers can supply you with USA and also international dialing. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 SIP URI: [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy Sent: Monday, May 08, 2006 8:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hi...Please help me Hi Friends, Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone. Now, I want to make calls to US using VoIP Asterisk. I have registered with Vebtel (VoIP IP Telephony Service provider). They had given me one VoIP Modem called Voice Finder AP 200 and the below values: Inbound Number: 123456789 Public IP Number: 55.23.789.145 Password: xyz (These values are dummy values) Currently we are making US calls using VoIP provided by Vebtel. Now, I want to make US calls using this Vebtel service from Asterisk. How can I do this? I am unable to understand where to give above mentioned values? What configuration files I should use to implement this using the Vebtel SIP provider? Do I need to provide any more values to implement this using Asterisk from Vebtel? Waiting for your quick response. Thank you. Regards, Chandra. Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --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] Hi...Please help me
Wouldn't it be easier to replace the callername to the exten.
example:
exten = _x.,1,SetCallerIDname(${EXTEN})
exten = _x.,2,SetCallerIDnum(${CALLERIDNUM})
exten = _x.,3,dial,SIP/number
That way, the Caller Name would show the extension it is ringing and the
callerid will still show the calling party.
Now you don't need the softphone to do it... just a phone with callerid.
bp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, May 02, 2006 5:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Hi...Please help me
On Tuesday 02 May 2006 16:42, hugolivude wrote:
We share SIP phones at the office in a 1:4 ratio. You're probably
asking - how do you know when a ringing phone is for you? Well,
everyone in our office gets an XLite softphone, and I direct calls to
make BOTH the SIP phone AND the XLite ring. If your XLite pops up,
you know that ring phone is for you.
That seems to be humongous overkill... why not just use any of the caller ID
popup apps instead of running that behemoth X-Lite? If the popup comes up,
the phone's for you.
-A.
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RE: [Asterisk-Users] Hi...Please help me
Hi friends,Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6 version. I have installed Asterisk in my PC and "X-Lite" as softphone in my PC and client PC. Here my user name is "chandra" and client user name is "aarti". I have added these lines to configuration files at the end of file.added contents in sip.conf:[aarti] type=friend username=aarti secret=aarti host=dynamic context=tutorial[chandra] type=friend username=chandra secret=chandra host=dynamic context=tutorialadded contents in extensions.conf:[tutorial] exten = 101,1,Dial(SIP/aarti) exten = 102,1,Dial(SIP/chandra)Here, "aarti" is client, "chandra" is mine and Asterisk is also installed in my PC (chandra) and it is successfully connected to Asterisk server using "X-Lite" softphone. But, when i try to connect from "aarti" system using softphone, it displays an error message "login timedout, contact system admin". Is there any problem with the content of sip.conf file or extensions.conf file? I have not connected any external hardware to my pc. I just want to connect Asterisk server to my collegues PC's like Intercom within my office LAN using headphones. How can I do this? Please tell me. Looking forward for your response. Thank you.Regards, Chandra. Evalyn Wafula [EMAIL PROTECTED] wrote: Hi Chandra, I am also new to Asterisk and I have only just started installing a test system but I probably can help clarify one or two things. I think asterisk "clients" are phones not PCs unless you use"soft phones" which is software onthe PC(somewhat like Skype) that you use to make and answer phone calls. So you might not need to install anything on your PCs if you will use IP phones or ATAs as mentioned by Gonzalo.The hardware you need depends on what you require your asterisk to do. If you will be making only IP calls using IP phones, then you only need asterisk running on your server with no extra hardware. But if you need to connect with analog/digital phone equipment, then you need extra hardware on the server. You do not physically connect your VOIP phone to the asterisk server. You connect it to the network that has the server through a normal network point and configure it to find the server. You probably oughtto take Gonzalo's advice and head over to: http://www.voip-info.org/wiki-Asteriskand do some reading before you even start as it will help you fit many pieces of the asterisk "puzzle" together. It helped me get started. Then you probably will have fewer questions that list members will answer more readily :) Regards Wafula Blab-away for as little as 1¢/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice.___ --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] Hi...Please help me
can u check what this command gives iptables -L or do iptables -F [ Not advisable , but for testing OK ] then try again --- Crazy Boy [EMAIL PROTECTED] wrote: Hi friends, Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6 version. I have installed Asterisk in my PC and X-Lite as softphone in my PC and client PC. Here my user name is chandra and client user name is aarti. I have added these lines to configuration files at the end of file. added contents in sip.conf: [aarti] type=friend username=aarti secret=aarti host=dynamic context=tutorial [chandra] type=friend username=chandra secret=chandra host=dynamic context=tutorial added contents in extensions.conf: [tutorial] exten = 101,1,Dial(SIP/aarti) exten = 102,1,Dial(SIP/chandra) Here, aarti is client, chandra is mine and Asterisk is also installed in my PC (chandra) and it is successfully connected to Asterisk server using X-Lite softphone. But, when i try to connect from aarti system using softphone, it displays an error message login timedout, contact system admin. Is there any problem with the content of sip.conf file or extensions.conf file? I have not connected any external hardware to my pc. I just want to connect Asterisk server to my collegues PC's like Intercom within my office LAN using headphones. How can I do this? Please tell me. Looking forward for your response. Thank you. Regards, Chandra. Evalyn Wafula [EMAIL PROTECTED] wrote: Hi Chandra, I am also new to Asterisk and I have only just started installing a test system but I probably can help clarify one or two things. I think asterisk clients are phones not PCs unless you use soft phones which is software on the PC (somewhat like Skype) that you use to make and answer phone calls. So you might not need to install anything on your PCs if you will use IP phones or ATAs as mentioned by Gonzalo. The hardware you need depends on what you require your asterisk to do. If you will be making only IP calls using IP phones, then you only need asterisk running on your server with no extra hardware. But if you need to connect with analog/digital phone equipment, then you need extra hardware on the server. You do not physically connect your VOIP phone to the asterisk server. You connect it to the network that has the server through a normal network point and configure it to find the server. You probably oughtto take Gonzalo's advice and head over to: http://www.voip-info.org/wiki-Asterisk and do some reading before you even start as it will help you fit many pieces of the asterisk puzzle together. It helped me get started. Then you probably will have fewer questions that list members will answer more readily :) Regards Wafula - Blab-away for as little as 1¢/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice. ___ --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 Send instant messages to your online friends http://uk.messenger.yahoo.com ___ Win tickets to the 2006 FIFA World Cup Germany with Yahoo! Messenger. http://advision.webevents.yahoo.com/fifaworldcup_uk/ ___ --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] Hi...Please help me
You are missing the dtmf mode, and most importantly the codec to be used. I would also add the nat=yes, that is probably why your phone isnt registering. See below for example config: [chandra] type=friend username=chandra secret=chandra nat=yes host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw allow=g729 context=tutorial canreinvite=no From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Crazy Boy Sent: Tuesday, May 02, 2006 8:58 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Hi...Please help me Hi friends, Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6 version. I have installed Asterisk in my PC and X-Lite as softphone in my PC and client PC. Here my user name is chandra and client user name is aarti. I have added these lines to configuration files at the end of file. added contents in sip.conf: [aarti] type=friend username=aarti secret=aarti host=dynamic context=tutorial [chandra] type=friend username=chandra secret=chandra host=dynamic context=tutorial added contents in extensions.conf: [tutorial] exten = 101,1,Dial(SIP/aarti) exten = 102,1,Dial(SIP/chandra) Here, aarti is client, chandra is mine and Asterisk is also installed in my PC (chandra) and it is successfully connected to Asterisk server using X-Lite softphone. But, when i try to connect from aarti system using softphone, it displays an error message login timedout, contact system admin. Is there any problem with the content of sip.conf file or extensions.conf file? I have not connected any external hardware to my pc. I just want to connect Asterisk server to my collegues PC's like Intercom within my office LAN using headphones. How can I do this? Please tell me. Looking forward for your response. Thank you. Regards, Chandra. Evalyn Wafula [EMAIL PROTECTED] wrote: Hi Chandra, I am also new to Asterisk and I have only just started installing a test system but I probably can help clarify one or two things. I think asterisk clients are phones not PCs unless you usesoft phones which is software onthe PC(somewhat like Skype) that you use to make and answer phone calls. So you might not need to install anything on your PCs if you will use IP phones or ATAs as mentioned by Gonzalo. The hardware you need depends on what you require your asterisk to do. If you will be making only IP calls using IP phones, then you only need asterisk running on your server with no extra hardware. But if you need to connect with analog/digital phone equipment, then you need extra hardware on the server. You do not physically connect your VOIP phone to the asterisk server. You connect it to the network that has the server through a normal network point and configure it to find the server. You probably ought to take Gonzalo's advice and head over to: http://www.voip-info.org/wiki-Asteriskand do some reading before you even start as it will help you fit many pieces of the asterisk puzzle together. It helped me get started. Then you probably will have fewer questions that list members will answer more readily :) Regards Wafula Blab-away for as little as 1¢/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice. ___ --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] Hi...Please help me
First off, I agree w/ Gonzalo – softphones didn't work out for me either. One thing that did work great tho was a combo. We share SIP phones at the office in a 1:4 ratio. You're probably asking – how do you know when a ringing phone is for you? Well, everyone in our office gets an XLite softphone, and I direct calls to make BOTH the SIP phone AND the XLite ring. If your XLite pops up, you know that ring phone is for you… Here's some answers to your other questions • What I have to install in client PC's? Just the softphone client (e.g. XLite (SIP) Cubix (IAX) http://www.virbiage.com/cubix.php • What hardware I need? Nothing too fancy. Your PCs seem OK. For Asterisk, I'm using an old Pentium 4 beater with 1Gig memory and it handles the whole office (19) just fine. • How can I take decission to buy extra hardware (like Zaptel products) OR no need of buying extra hardware? ( I will be using Asterisk for 70 PC's and a server) This depends on what you want in the way of handsets, and what kind of connectivity you want to the PSTN (Public Switched Telephone Network). You could get away with no extra hardware in a pure VoIP solution. Connect Asterisk to the Internet w/ an Ethernet cable and use SIP based phones that also communicate over a network. Note that if you don't use any Digium hardware, I believe that you need to use ztdummy to control timing (never used it myself) http://www.voip-info.org/wiki-Asterisk+timer+ztdummy • Is it sufficient to buy hardware for server only OR for client PC's also? Again, your PCs seem OK. How you kit out your server depends upon what you want. • How can I connect my VoIP phone to server? Once you have Asterisk installed, you have to configure your VoIP phone to register with it. For example, look here for how to configure Polycom Soundpoint 501s - http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501. You'll also have to have the appropriate entries in SIP.conf for the phone AND to connect to your VoIP service provider http://www.voip-info.org/wiki-Asterisk+config+sip.conf • How can I connect hardware to server? Don't understand this one. If you use telephony boards, you'll need drivers. Depending upon the board you may also have to physically connect your phone to it with a telephone wire (as is the case with TDM boards for example) • How can I connect PSTN line to server PC? Assuming analogue phones you'll need a TDM card with an FXO port (outgoing) for each line you have (http://www.digium.com/en/products/hardware/analogcards.php). You'll also need an FXS port for each phone you have on your TDM card as well. Yours, H On 5/2/06, William Piper [EMAIL PROTECTED] wrote: You are missing the dtmf mode, and most importantly… the codec to be used. I would also add the nat=yes, that is probably why your phone isn't registering. See below for example config: [chandra] type=friend username=chandra secret=chandra nat=yes host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw allow=g729 context=tutorial canreinvite=no From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy Sent: Tuesday, May 02, 2006 8:58 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Hi...Please help me Hi friends, Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6 version. I have installed Asterisk in my PC and X-Lite as softphone in my PC and client PC. Here my user name is chandra and client user name is aarti. I have added these lines to configuration files at the end of file. added contents in sip.conf: [aarti] type=friend username=aarti secret=aarti host=dynamic context=tutorial [chandra] type=friend username=chandra secret=chandra host=dynamic context=tutorial added contents in extensions.conf: [tutorial] exten = 101,1,Dial(SIP/aarti) exten = 102,1,Dial(SIP/chandra) Here, aarti is client, chandra is mine and Asterisk is also installed in my PC (chandra) and it is successfully connected to Asterisk server using X-Lite softphone. But, when i try to connect from aarti system using softphone, it displays an error message login timedout, contact system admin. Is there any problem with the content of sip.conf file or extensions.conf file? I have not connected any external hardware to my pc. I just want to connect Asterisk server to my collegues PC's like Intercom within my office LAN using headphones. How can I do this? Please tell me. Looking forward for your response. Thank you. Regards, Chandra. Evalyn Wafula [EMAIL PROTECTED] wrote: Hi Chandra, I am also new to Asterisk and I have only just started installing a test system but I probably can help clarify one or two things. I think asterisk clients are phones not PCs unless you use soft phones which is software on the PC (somewhat like Skype) that you use
Re: [Asterisk-Users] Hi...Please help me
On Tuesday 02 May 2006 16:42, hugolivude wrote: We share SIP phones at the office in a 1:4 ratio. You're probably asking – how do you know when a ringing phone is for you? Well, everyone in our office gets an XLite softphone, and I direct calls to make BOTH the SIP phone AND the XLite ring. If your XLite pops up, you know that ring phone is for you… That seems to be humongous overkill... why not just use any of the caller ID popup apps instead of running that behemoth X-Lite? If the popup comes up, the phone's for you. -A. ___ --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] Hi...Please help me
Hi Chandra, I am also new to Asterisk and I have only just started installing a test system but I probably can help clarify one or two things. I think asterisk "clients" are phones not PCs unless you use"soft phones" which is software onthe PC(somewhat like Skype) that you use to make and answer phone calls. So you might not need to install anything on your PCs if you will use IP phones or ATAs as mentioned by Gonzalo. The hardware you need depends on what you require your asterisk to do. If you will be making only IP calls using IP phones, then you only need asterisk running on your server with no extra hardware. But if you need to connect with analog/digital phone equipment, then you need extra hardware on the server. You do not physically connect your VOIP phone to the asterisk server. You connect it to the network that has the server through a normal network point and configure it to find the server. You probably ought to take Gonzalo's advice and head over to: http://www.voip-info.org/wiki-Asteriskand do some reading before you even start as it will help you fit many pieces of the asterisk "puzzle" together. It helped me get started. Then you probably will have fewer questions that list members will answer more readily :) Regards Wafula From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy BoySent: 26 April 2006 14:56To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Hi...Please help me Hi,Thank you for your response. Basically, I follow "O Reilly AsteriskTFOT.pdf" book and some other eBooks. They have mentioned how to install Asterisk in server. But, they have not mentioned What I have to install in client PC's? What hardware I need? How can I take decission to buy extra hardware (like Zaptel products) OR no need of buying extra hardware? ( I will be using Asterisk for 70 PC's and a server) Is it sufficient to buy hardware for server only OR for client PC's also? How can I connect my VoIP phone to server? How can I connect hardware to server? How can I connect PSTN line to server PC?Please guide me to complete this task. Waiting for your response. Thank you.Regards,Chandra.Gonzalo Servat [EMAIL PROTECTED] wrote: On 4/24/06, Crazy Boy <[EMAIL PROTECTED]>wrote: Hi Friends,[..snip..] --- Employee 1 PC (Softphone i.e., Headphones with Mic) --- Employee 2 PC (Softphone i.e., Headphones with Mic) --- Employee 3 PC (Softphone i.e., Headphones with Mic) --- -- --- -- --- Employee 10 PC (Softphone i.e., Headphones with Mic) and vice versa. How can I implement this? Is it possible to implement this using "Asterisk" software? If It can be implemented using "Asterisk" software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware?[..snip..]It can be done with Asterisk. For the server side, you would need toinstall Asterisk on your Fedora 5 box, Zaptel and lots of Wikireading.I don't recommend using softphones for your employee PCs. It lookslike an attractive solution at first (from a cost perspective) but inreality it's not very practical (at least that was my experience).Buying 5 x 2 port ATAs will cost you around $300-$350 which is notreally expensive considering the kind of powerful PBX you will have atyour disposal. I would have suggested some Digium hardware for the FXS(extensions) but I think it will be a lot more expensive (for 10extensions) than the ATAs solution. You could also look into a channelbank, but again it will be more expensive than the 5 ATAs. As for theFXO (incoming/outgoing PSTN) I recommend buying Digium hardware(TDM400P).Hope this helps, and good luck!Regards,Gonzalo.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Blab-away for as little as 1¢/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice. ___ --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] Hi...Please help me
Hi,Thank you for your response. Basically, I follow "O Reilly AsteriskTFOT.pdf" book and some other eBooks. They have mentioned how to install Asterisk in server. But, they have not mentioned What I have to install in client PC's?What hardware I need?How can I take decission to buy extra hardware (like Zaptel products) OR no need of buying extra hardware? ( I will be using Asterisk for 70 PC's and a server)Is it sufficient to buy hardware for server only OR for client PC's also?How can I connect my VoIP phone to server?How can I connect hardware to server?How can I connect PSTN line to server PC?Please guide me to complete this task. Waiting for your response. Thank you.Regards,Chandra.Gonzalo Servat [EMAIL PROTECTED] wrote: On 4/24/06, Crazy Boy <[EMAIL PROTECTED]> wrote: Hi Friends,[..snip..] --- Employee 1 PC (Softphone i.e., Headphones with Mic) --- Employee 2 PC (Softphone i.e., Headphones with Mic) --- Employee 3 PC (Softphone i.e., Headphones with Mic) ----- ----- --- Employee 10 PC (Softphone i.e., Headphones with Mic) and vice versa. How can I implement this? Is it possible to implement this using "Asterisk" software? If It can be implemented using "Asterisk" software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware?[..snip..]It can be done with Asterisk. For the server side, you would need toinstall Asterisk on your Fedora 5 box, Zaptel and lots of Wikireading.I don't recommend using softphones for your employee PCs. It lookslike an attractive solution at first (from a cost perspective) but inreality it's not very practical (at least that was my experience).Buying 5 x 2 port ATAs will cost you around $300-$350 which is notreally expensive considering the kind of powerful PBX you will have atyour disposal. I would have suggested some Digium hardware for the FXS(extensions) but I think it will be a lot more expensive (for 10extensions) than the ATAs solution. You could also look into a channelbank, but again it will be more expensive than the 5 ATAs. As for theFXO (incoming/outgoing PSTN) I recommend buying Digium hardware(TDM400P).Hope this helps, and good luck!Regards,Gonzalo.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Blab-away for as little as 1¢/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice.___ --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] Hi...Please help me
It's all possible. Paul Hales -- Paul Hales Technical Manager Asterisk IT bus: 03 8320 8100 mob: 0434 225 491 Crazy Boy wrote: Hi Friends, I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP phone and have registered with VoIP service provider. Now, I want to implement VOIP PBX facility to all of my systems. The structure for the same is: PSTN (Phone call) --- VOIP phone --- Server system --- --- Employee 1 PC (Softphone i.e., Headphones with Mic) --- Employee 2 PC (Softphone i.e., Headphones with Mic) --- Employee 3 PC (Softphone i.e., Headphones with Mic) ----- ----- --- Employee 10 PC (Softphone i.e., Headphones with Mic) and vice versa. How can I implement this? Is it possible to implement this using Asterisk software? If It can be implemented using Asterisk software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware? Please reply me. Thank you Thanks Regards, Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://beta.messenger.yahoo.com http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://beta.messenger.yahoo.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 http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://beta.messenger.yahoo.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] Hi...Please help me
On 4/24/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi Friends, [..snip..] --- Employee 1 PC (Softphone i.e., Headphones with Mic) --- Employee 2 PC (Softphone i.e., Headphones with Mic) --- Employee 3 PC (Softphone i.e., Headphones with Mic) ----- ----- --- Employee 10 PC (Softphone i.e., Headphones with Mic) and vice versa. How can I implement this? Is it possible to implement this using Asterisk software? If It can be implemented using Asterisk software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware? [..snip..] It can be done with Asterisk. For the server side, you would need to install Asterisk on your Fedora 5 box, Zaptel and lots of Wiki reading. I don't recommend using softphones for your employee PCs. It looks like an attractive solution at first (from a cost perspective) but in reality it's not very practical (at least that was my experience). Buying 5 x 2 port ATAs will cost you around $300-$350 which is not really expensive considering the kind of powerful PBX you will have at your disposal. I would have suggested some Digium hardware for the FXS (extensions) but I think it will be a lot more expensive (for 10 extensions) than the ATAs solution. You could also look into a channel bank, but again it will be more expensive than the 5 ATAs. As for the FXO (incoming/outgoing PSTN) I recommend buying Digium hardware (TDM400P). Hope this helps, and good luck! Regards, Gonzalo. ___ --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] Hi...Please help me
Hi Friends,I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP phone and have registered with VoIP service provider. Now, I want to implement VOIP PBX facility to all of my systems. The structure for the same is:PSTN (Phone call) --- VOIP phone --- Server system --- --- Employee 1 PC (Softphone i.e., Headphones with Mic)--- Employee 2 PC (Softphone i.e., Headphones with Mic)--- Employee 3 PC (Softphone i.e., Headphones with Mic) --- -- --- ----- Employee 10 PC (Softphone i.e., Headphones with Mic)and vice versa.How can I implement this? Is it possible to implement this using "Asterisk" software? If It can be implemented using "Asterisk" software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware? Please reply me. Thank youThanks Regards,Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1/min.___ --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] Hi...Please help me
Yes it is possible - check out the Asterisk manual or nice book from O'Reilly - Asterisk PBX (The Furute of telephony) Marcel Crazy Boy wrote: Hi Friends, I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP phone and have registered with VoIP service provider. Now, I want to implement VOIP PBX facility to all of my systems. The structure for the same is: PSTN (Phone call) --- VOIP phone --- Server system --- --- Employee 1 PC (Softphone i.e., Headphones with Mic) --- Employee 2 PC (Softphone i.e., Headphones with Mic) --- Employee 3 PC (Softphone i.e., Headphones with Mic) ----- ----- --- Employee 10 PC (Softphone i.e., Headphones with Mic) and vice versa. How can I implement this? Is it possible to implement this using Asterisk software? If It can be implemented using Asterisk software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware? Please reply me. Thank you Thanks Regards, Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://beta.messenger.yahoo.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 ___ --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] Hi...Please help me
For hardware check out this page: http://www.digium.com/en/products/hardware/ Marcel Crazy Boy wrote: Hi Friends, I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP phone and have registered with VoIP service provider. Now, I want to implement VOIP PBX facility to all of my systems. The structure for the same is: PSTN (Phone call) --- VOIP phone --- Server system --- --- Employee 1 PC (Softphone i.e., Headphones with Mic) --- Employee 2 PC (Softphone i.e., Headphones with Mic) --- Employee 3 PC (Softphone i.e., Headphones with Mic) ----- ----- --- Employee 10 PC (Softphone i.e., Headphones with Mic) and vice versa. How can I implement this? Is it possible to implement this using Asterisk software? If It can be implemented using Asterisk software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware? Please reply me. Thank you Thanks Regards, Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://beta.messenger.yahoo.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 ___ --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
