RE: [Asterisk-Users] Newbie's doubt on sip.conf
kaiduan xie wrote: 1) Can I have two or more SIP phones acting as extensions in one Asterisk box, and at the same time, registered to a SIP proxy, say Free World Dialup? If yes, how? Your phone would have to support registering with more than one sip server. It may seem like what you want, but I doubt it is necessary. 2) Why we need a section in the sip.conf for the proxy, say, Free World Dialup's fwd.pulver.com? In the case of 1), how to assign the value to section [fwd.pulver.com], since there are more than one sip phone, each with different FWD number? You should be able to register and listen for calls from multiple fwd numbers and direct them to different extensions. 3) Can anyone explain the meaning of peer, friend, user in more details? For each case, what is the role of Asterisk in SIP world, a UA, a proxy, or others? Peer: A connection that sends calls to asterisk. User: A connection that asterisk sends calls out to. Friend: an attempt at a combination of both, to simplify set up of phones that send and receive calls. (There are several people here who will tell you friend is evil.) 4) If we only use SIP phone as extensions in Asterisk, the SIP phone doesnot associate with outside proxy, does Asterisk act as a proxy for inter-extension call between the SIP phones? In this case, for the outgoing call originating from SIP phone to other network, say, PSTN, does Asterisk act as a gateway? (PSTN connection with Asterisk is assumed.) Hmm, let me give brief examples of things you can do(* = asterisk): sip phone - * - sip phone sip phone - * - hardline adapter - PSTN or T1/E1/BRI/etc. sip phone - * - other voip service(fwd, iaxtel, etc) sip phone - * - other voip service(voicepulse, nufone, iconnecthere) - PSTN Most(if not all) of these can be turned around so that they are still valid reading right to left. - Andrew Thompson http://aktzero.com/ http://www.retirequickly.com/43653 ___ 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's doubt on sip.conf
Andrew Thompson wrote: Peer: A connection that sends calls to asterisk. User: A connection that asterisk sends calls out to. Friend: an attempt at a combination of both, to simplify set up of phones that send and receive calls. (There are several people here who will tell you friend is evil.) Actually, the opposite. * Peers may or may not register so we can place calls to them. * Users place calls to Asterisk. /Olle ___ 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's doubt on sip.conf
3) Can anyone explain the meaning of peer, friend, user in more details? For each case, what is the role of Asterisk in SIP world, a UA, a proxy, or others? In some diagrams, Asterisk take's the role of a SIP Proxy, but it is *not* a SIP proxy by design. Asterisk answers SIP calls and originates calls as just another SIP user agent client/server. Asterisk is also acting as a SIP registrar, keeping track of where SIP ua's are so we can place calls to them ([peer]s). /O ___ 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's doubt on sip.conf
Andrew, Thanks for help. But you still doesnot answer my questions. Please see the inline comments. BTW, I use Getting Started with Asterisk as reference (http://www.automated.it/guidetoasterisk.htm#_Toc49248767). Thanks, kaiduan --- Andrew Thompson [EMAIL PROTECTED] wrote: kaiduan xie wrote: 1) Can I have two or more SIP phones acting as extensions in one Asterisk box, and at the same time, registered to a SIP proxy, say Free World Dialup? If yes, how? Your phone would have to support registering with more than one sip server. It may seem like what you want, but I doubt it is necessary. In my case, each SIP phones register with FWD, and was assigned a extension. I can put more register for each phone in sip.conf. For example, register = 21103:[EMAIL PROTECTED]/1000 register = 21104:[EMAIL PROTECTED]/2000 My concern is how to setup the [fwd.pulver.com] [fwd.pulver.com] type=friend secret=mypassword username=my fwd number host=fwd.pulver.com since there are two FWD numbers, should I set secret and username for each SIP Phones? Also, you still doesnot answer my question, what is the use of section [fwd.pulver.com]? Why we need secret and username in [fwd.pulver.com] since we already provide it in register = 21104:[EMAIL PROTECTED]/2000? 2) Why we need a section in the sip.conf for the proxy, say, Free World Dialup's fwd.pulver.com? In the case of 1), how to assign the value to section [fwd.pulver.com], since there are more than one sip phone, each with different FWD number? You should be able to register and listen for calls from multiple fwd numbers and direct them to different extensions. 3) Can anyone explain the meaning of peer, friend, user in more details? For each case, what is the role of Asterisk in SIP world, a UA, a proxy, or others? Peer: A connection that sends calls to asterisk. User: A connection that asterisk sends calls out to. Friend: an attempt at a combination of both, to simplify set up of phones that send and receive calls. (There are several people here who will tell you friend is evil.) Just to complain that the definition of peer, user, and Friend is not self-explained, :) 4) If we only use SIP phone as extensions in Asterisk, the SIP phone doesnot associate with outside proxy, does Asterisk act as a proxy for inter-extension call between the SIP phones? In this case, for the outgoing call originating from SIP phone to other network, say, PSTN, does Asterisk act as a gateway? (PSTN connection with Asterisk is assumed.) Hmm, let me give brief examples of things you can do(* = asterisk): sip phone - * - sip phone sip phone - * - hardline adapter - PSTN or T1/E1/BRI/etc. sip phone - * - other voip service(fwd, iaxtel, etc) sip phone - * - other voip service(voicepulse, nufone, iconnecthere) - PSTN Most(if not all) of these can be turned around so that they are still valid reading right to left. - Andrew Thompson http://aktzero.com/ http://www.retirequickly.com/43653 ___ 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 __ Post your free ad now! http://personals.yahoo.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
RE: [Asterisk-Users] Newbie's doubt on sip.conf
Olle E. Johansson wrote: Andrew Thompson wrote: Peer: A connection that sends calls to asterisk. User: A connection that asterisk sends calls out to. Friend: an attempt at a combination of both, to simplify set up of phones that send and receive calls. (There are several people here who will tell you friend is evil.) Actually, the opposite. * Peers may or may not register so we can place calls to them. * Users place calls to Asterisk. /Olle Oops, sorry about that. Guess I should have looked back at the wiki first. I haven't touched my asterisk config in a couple of weeks now... - Andrew Thompson http://aktzero.com/ http://www.retirequickly.com/43653 ___ 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's doubt on sip.conf
kaiduan xie wrote: Andrew, Thanks for help. But you still doesnot answer my questions. Please see the inline comments. I'm sorry that I didn't quite understand what you were asking. I'll try again... BTW, I use Getting Started with Asterisk as reference (http://www.automated.it/guidetoasterisk.htm#_Toc49248767). Thanks, kaiduan --- Andrew Thompson [EMAIL PROTECTED] wrote: kaiduan xie wrote: 1) Can I have two or more SIP phones acting as extensions in one Asterisk box, and at the same time, registered to a SIP proxy, say Free World Dialup? If yes, how? Your phone would have to support registering with more than one sip server. It may seem like what you want, but I doubt it is necessary. In my case, each SIP phones register with FWD, and was assigned a extension. I can put more register for each phone in sip.conf. For example, register = 21103:[EMAIL PROTECTED]/1000 register = 21104:[EMAIL PROTECTED]/2000 The register line tells asterisk to contact the specified server and register the availability of the user(extension) specified. This means any calls destined for that user will be passed to your asterisk. The /1000 tells asterisk to direct any calls to 21103 to extension 1000. (Not specifically asked, but I wanted to elaborate on anyway.) My concern is how to setup the [fwd.pulver.com] [fwd.pulver.com] type=friend secret=mypassword username=my fwd number host=fwd.pulver.com since there are two FWD numbers, should I set secret and username for each SIP Phones? Also, you still doesnot answer my question, what is the use of section [fwd.pulver.com]? Why we need secret and username in [fwd.pulver.com] since we already provide it in register = 21104:[EMAIL PROTECTED]/2000? If asterisk is registering succesfully, you should be able to *receive* calls from fwd. At this point, you need to define an outbound entry for each fwd number you want to create calls as. If you only need for your outbound calls to appear as from one fwd number, a single entry will do. I'm currently referencing: http://www.voip-info.org/wiki-Asterisk+config+sip.conf Create a type=peer section for each fwd number you'll be dialing out as. You will only need this in your Dial() line. I have only a single fwd number right now, so I have not built outbound contexts for more than one fwd number. You'll need to ask for that specifically from the list. - Andrew Thompson http://aktzero.com/ http://www.retirequickly.com/43653 ___ 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's doubt on sip.conf
use the wiki as a reference www.voip-info.org - Original Message - From: kaiduan xie [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 12:10 PM Subject: RE: [Asterisk-Users] Newbie's doubt on sip.conf Andrew, Thanks for help. But you still doesnot answer my questions. Please see the inline comments. BTW, I use Getting Started with Asterisk as reference (http://www.automated.it/guidetoasterisk.htm#_Toc49248767). Thanks, kaiduan --- Andrew Thompson [EMAIL PROTECTED] wrote: kaiduan xie wrote: 1) Can I have two or more SIP phones acting as extensions in one Asterisk box, and at the same time, registered to a SIP proxy, say Free World Dialup? If yes, how? Your phone would have to support registering with more than one sip server. It may seem like what you want, but I doubt it is necessary. In my case, each SIP phones register with FWD, and was assigned a extension. I can put more register for each phone in sip.conf. For example, register = 21103:[EMAIL PROTECTED]/1000 register = 21104:[EMAIL PROTECTED]/2000 My concern is how to setup the [fwd.pulver.com] [fwd.pulver.com] type=friend secret=mypassword username=my fwd number host=fwd.pulver.com since there are two FWD numbers, should I set secret and username for each SIP Phones? Also, you still doesnot answer my question, what is the use of section [fwd.pulver.com]? Why we need secret and username in [fwd.pulver.com] since we already provide it in register = 21104:[EMAIL PROTECTED]/2000? 2) Why we need a section in the sip.conf for the proxy, say, Free World Dialup's fwd.pulver.com? In the case of 1), how to assign the value to section [fwd.pulver.com], since there are more than one sip phone, each with different FWD number? You should be able to register and listen for calls from multiple fwd numbers and direct them to different extensions. 3) Can anyone explain the meaning of peer, friend, user in more details? For each case, what is the role of Asterisk in SIP world, a UA, a proxy, or others? Peer: A connection that sends calls to asterisk. User: A connection that asterisk sends calls out to. Friend: an attempt at a combination of both, to simplify set up of phones that send and receive calls. (There are several people here who will tell you friend is evil.) Just to complain that the definition of peer, user, and Friend is not self-explained, :) 4) If we only use SIP phone as extensions in Asterisk, the SIP phone doesnot associate with outside proxy, does Asterisk act as a proxy for inter-extension call between the SIP phones? In this case, for the outgoing call originating from SIP phone to other network, say, PSTN, does Asterisk act as a gateway? (PSTN connection with Asterisk is assumed.) Hmm, let me give brief examples of things you can do(* = asterisk): sip phone - * - sip phone sip phone - * - hardline adapter - PSTN or T1/E1/BRI/etc. sip phone - * - other voip service(fwd, iaxtel, etc) sip phone - * - other voip service(voicepulse, nufone, iconnecthere) - PSTN Most(if not all) of these can be turned around so that they are still valid reading right to left. - Andrew Thompson http://aktzero.com/ http://www.retirequickly.com/43653 ___ 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 __ Post your free ad now! http://personals.yahoo.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 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's doubt on sip.conf
Your probably going to get this url (www.voip-info.org) thrown at you by a few other people too...check there if you haven't already for more information. -mitchel On Wed, 7 Jul 2004 00:45:53 -0400 (EDT), kaiduan xie [EMAIL PROTECTED] wrote: Hi, I have some doubts on sip.conf. 1) Can I have two or more SIP phones acting as extensions in one Asterisk box, and at the same time, registered to a SIP proxy, say Free World Dialup? If yes, how? 2) Why we need a section in the sip.conf for the proxy, say, Free World Dialup's fwd.pulver.com? In the case of 1), how to assign the value to section [fwd.pulver.com], since there are more than one sip phone, each with different FWD number? [fwd.pulver.com] type=friend secret=mypassword username=my fwd number host=fwd.pulver.com 3) Can anyone explain the meaning of peer, friend, user in more details? For each case, what is the role of Asterisk in SIP world, a UA, a proxy, or others? 4) If we only use SIP phone as extensions in Asterisk, the SIP phone doesnot associate with outside proxy, does Asterisk act as a proxy for inter-extension call between the SIP phones? In this case, for the outgoing call originating from SIP phone to other network, say, PSTN, does Asterisk act as a gateway? (PSTN connection with Asterisk is assumed.) Any comments are welcome, thanks, kaiduan __ Post your free ad now! http://personals.yahoo.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 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
