[Asterisk-Users] In/out calls from/to same sip provider

[Pizco Dominguez]

It doesn't seem to work like these, exactly.
On Fri, May 13, 2005 at 12:37:56PM -0700, Adrian A wrote:
> If you have a [provider] peer in sip.conf, what happens if you use a
> register command in sip.conf such as:
> 
> register => user:pass at providerip/provider

I have that line, but, to my knowledge, the "register" line is used to
make the remote server know that we are listening for incoming calls. By
the way, the last item should be "user" or an extension you have already
defined in extensions.conf ready to receive these calls, not "provider".

> and in extensions.conf you have:
> 
> Dial(SIP/${EXTEN:2}@provider)

I also have this line but, in may case, the moment I define [provider]
for outgoing in sip.conf, incoming calls will read it and, finding the
wrong fields, will fail. With some help form this list, I've found out
that if I define a second peer for incoming and I put it after the one
for outgoing, each one with its own fields, it works. If I change the
order, don't ask me why, incoming calls will fail because they read the
wrong peer.

This is my case, I still don't know if it is the same with other
providers.

> 
> 
> On 5/13/05, Pizco Dominguez <pizco at guadawireless.org> wrote:
> > Hi.
> > 
> > I'm new to asterisk and, one way or the other, I manage to get it working
> > for me.
> > 
> > But I'm having a hard time getting calls going to and coming from the
> > same provider, since the definition of the peer in sip.conf seems to be
> > different AND not compatible for incoming and outgoing call.
> > 
> > Outgoing calls need a "secret" and "username" definition in the peer
> > context of sip.conf, while incoming ones will have nothing to do with
> > those fields.
> > 
> > So I can have incoming or outgoing calls regarding one provider, but not
> > both.
> > 
> > I've also tried the sample sintax
> > 
> > "exten =>_42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)"
> > 
> > that comes with the distribution (debian-sarge), but only to get asterisk
> > unable to create sip channel because
> > 
> > "host dialed_number at real_sip_server_address doesn't exist". The address
> > is that of the provider.
> > 
> > voip.org and asteriskdocs.org seems to lead me nowhere.
> > 
> > I must be missing something obvious, but can't figure out what it is.
> > 
> > Anybody?
> > 
> > Thanks.
> > 
> > --
> > Pizco Dominguez
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> > 
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-- 
Pizco Dominguez
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