[Asterisk-Users] SIP config documentation

Wed Feb 18 09:06:19 MST 2004

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" <oej at edvina.net>
To: <asterisk-users at lists.digium.com>
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
> Asterisk-Users at 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