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
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 <kaiduanx@yahoo.ca> 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 > 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 >
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
Changing rxgain and txgain in zapata.conf does not seem to have any effect when I test with ztmonitor. I reloaded the conf files in *, but still no difference. Wonder what needs to be done to experiment with rxgain / txgain for a X100P Thanks for any pointers Hariom --------------------------------- Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040707/52df67ba/attachment.htm
> Changing rxgain and txgain in zapata.conf does not seem to have > any effect when I test with ztmonitor. I reloaded the conf > files in *, but still no difference. > > Wonder what needs to be done to experiment with rxgain / txgain > for a X100PReloading does not affect the gain settings. You actually have to stop asterisk. There is also a high probablility that you need to stop/start the drivers as well.