Can musiconhold=<class> be included in sip.conf? I want to play music on hold for calling users on the VoIP side. Currently, I can only play moh when the call came from the PSTN (zapata).
----- Original Message ----- From: "Olle E. Johansson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, February 18, 2004 2:58 AM Subject: Re: [Asterisk-Users] SIP config documentation > Costa Tsaousis wrote: > > > > I was trying to figure out all the valid options for a sip.conf and I > > believe I found a few weird things (or just a few things that are weird > > to me :) Anyway, I decided to post this here together with my questions > > and notes in case other people need this info too or have similar > > questions. > Costa, > Thank you for an excellent document. I'll try to merge this into the > wiki docs. I just recently checked what's valid for users and peers > for the chan_sip2 code. You'll see my table in there. Also, in that > beta version of the SIP channel, you'll find some more documentation > and some minor fixes. > > > 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. > > > canreinvite= ; UP yes, no or update. > > > > Q2: What the "update" option does? > > I think the update is not a keyword, but a value. > See here > http://www.voip-info.org/wiki-Asterisk%20sip%20qualify > > > 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. > > > > > > 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? > > > > > > 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. > > > > 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? > > > > 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. > > > 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. > > > mask= ; -P netmask for host= parameter. > This has to be defined *before* thet host= parameter. > What it does? Don't know. Anyone else? Why do Asterisk apply a host mask to > an IP address for a host? > > port= ; -P port for host= parameter. > If host=dynamic this applies to defaultip. > > 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. > > > Q7: I am really lost with these. I understand defaultip pretty well, but > > then, what is exactly the use of host, port and mask, for peers? Does > > this have to do something with nat=yes in order to set our asterisk > > "public view", or what? > 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. > > > 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. > > > context= ; Default context for incoming SIP calls > not coming from users or peers (by IP address) > > > language= ; voice messages language > Not only voice messages, also sets indications. > > > callerid= ; Default caller id (name only & becomes fromuser too) > > > > The effects of the callerid option are very funny. The only thing that > > produces valid SIP headers is just a word. The default is: asterisk > Have you noticed that the realm in authentication is always "asterisk". > In chan_sip2, you can configure this to your domain (also according to > the RFC). > > > autocreatepeer= ; Automatically create peers from incoming calls? > The peer is created by incoming registrations. > > > localnet= ; ??? > > localmask= ; ??? > > externip= ; Address that we're going to put in SIP messages if we're > > behind a NAT > These three options are involved in NAT handling for outbound calls to SIP > proxies we register with. Localnet/localmask is used to determine wheter to > use the externIP or not. > > > disallow= ; Disallow codecs > > allow= ; Allow codecs in order of preference > 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 :-) > > As you've noted, there are things to fix in the SIP channel. > > /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 _______________________________________________ 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
