Hi all, (oej, I have lost your e-mail somehow, so I have replied to some other reply to you... sorry!)
>> context= ; UP, the context name for placing calls >> >> Q1: Why is there a context for peers? > > We use peers in some other situations as well. This is strange and > rather undocumented, but an incoming call is first matched by > username with the defined users (including 'friends'). After that, > we match on IP address with the peer table. Hopefully those > peers have authentication (a secret), so we authenticate and accept > the call based on IP. This is used when connecting Asterisk as > a PSTN-gateway to other SIP proxys. Incoming calls are placed > in the peer context. Does this mean that if I have SER for example as the primary proxy for a domain and I want to use asterisk as a media gateway, I can have SER redirect my user agents to asterisk, which will be authenticated as nomral by asterisk (sip.conf entries for each agent), without using autocreatepeers=yes? >> canreinvite= ; UP yes, no or update. >> >> Q2: What the "update" option does? > > I think the update is not a keyword, but a value. Yes, my fault. > See here > http://www.voip-info.org/wiki-Asterisk%20sip%20qualify It is not a value of qualify. It is a value of canreinvite. Do you know what it does? >> callerid= ; U- caller id of the user: "Name <number>". > > Have to check this one. Been working a bit on this problem in the > chan_sip2 channel. I have submitted two bug reports. One includes a patch to chan_sip.c that fixes the problem. See: http://bugs.digium.com/bug_view_page.php?bug_id=0001074 Another is about CALLERIDNUM. This variable seems to strip the dots from the domain without practical reason (also SetCIDNum and the like do this). See: http://bugs.digium.com/bug_view_page.php?bug_id=0001075 >> callgroup= ; UP >> pckupgroup= ; UP >> >> Q4: Since a user cannot accept calls, why to setup call pickup for >> him/her? > Sorry, haven't used or checked call groups. Anyone else? No answer on this yet... >> language= ; U- language for voice messages and indications >> >> Q5: Why a peer does not have a language? What if we want to call >> someone >> with an IVR menu (via a .call file)? How we can choose the language >> for >> him/her? (yes, I know this can be set in the context the call will >> enter, but I think the elegant solution is to have this information >> here). >> > This is a bug fixed in the chan_sip2 channel. ok >> accountcode= ; U- CDR's account code >> incominglimit= ; U- concurrent call limitations ( >= 0 ) >> outgoinglimit= ; U- concurrent call limitations ( >= 0 ) >> >> Q6: How is it possible for a type=user phone to have BOTH incoming and >> outgoing limits? > Interesting question. Anyone else? No help on this either so far. >> nat= ; -P yes, no : Support NAT. (breaks RFC) > Well, yes, it breaks the RFC, but makes SIP work. What nat=yes really > does > is that it ignores the IP data within the registration or invite and use > the IP address Asterisk received the packet from. This works if the > client is contacting us directly, with no outbound proxy in between. > This is rather common in SIP proxy implementations right now. When STUN > and UPNP and other NAT/VoIP solutions are more frequently implemented, > the data sent to thte SIP server will not include any private NAT > networks > any more, but that will not happen overnight. ok >> fromdomain= ; -P Domain to show in the domain field of the outgoing >> call TO the peer. > I think this is mainly used when we REGISTER with an outbound proxy, > like FWD. >> fromuser= ; -P User to show in the user field of the outgoing call TO >> the peer. > Same here. fromdomain and fromuser overwrite the callerid sent TO the peer. The [general] section fromdomain and callerid are the same respectivelly (callerid in [general] is fromuser actually) for destinations not defined as peers in sip.conf (i.e. DNS SRV based lookups). The patch I have sumbitted above allows the type=user entities to have a fromdomain/fromuser given in their callerid: callerid = My Name <[EMAIL PROTECTED]> This allows asterisk serve multiple SIP domains concurrently (I am working on this for some time - I have build a configurator that automatically creates multi-domain configurations for using * for SIP virtual PBX services - If you are interested for this builder, just send me a note to send it to you - it is alpha currently). >> mask= ; -P netmask for host= parameter. > This has to be defined *before* thet host= parameter. Thanks for the hint. I didn't notice this. > What it does? Don't know. Anyone else? Why do Asterisk apply a host mask > to an IP address for a host? This is still open too. >> defaultip= ; -P if the peer does not register with us, where we should >> try to find it by default. > Useful if you restart Asterisk. The SIP device think it's registred for > a while longer, but Asterisk lost contact with them. ok. > host=<ip address> > port=<port no> > means you don't use SIP REGISTER, the SIP ua is always at the same > address. > Again, mask - I don't really know. ok >> username= ; -P username to send to the peer when calling the peer >> >> Q7: Since this is a peer option, it is really very badly documented in >> the various documents. All these documents state that this is used >> when >> the phone's login name is different from the default. But then, since >> this is a peer option it is ignored for type=user agents and is used >> only when asterisk is calling the phone. > > Agree, this is fuzzy. Have you noticed that it changes to peer name > after a while? Do 'sip show peers' at the CLI, and you'll notice. > > The chan_sip2 channel uses the Contact: at registration when we > send subsequent messages, like INVITE. As I see in chan_sip.c/initreqprep() username= is also used in INVITE for peers too. > As I stated earlier, I'm highly suspicious to the "in order of > preference" > part. Since I got no comments or replies on that mail, I suspect I'm > right > :-) What do you mean? Is the order of preference not working? A new question: Is chan_sip2 ready for production? Regards, Costa _______________________________________________ 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
