[Asterisk-Users] Connecting 2 Asterisk using SIP

Alvaro Parres aparres at gmail.com
Tue Dec 6 20:22:43 MST 2005


Why using SIP instead of IAX2 ???

Only a question becouse i always use IAX




On 12/6/05, Waldo Rubinstein <waldo at trianet.net> wrote:
>
> Well... not so perfectly.
>
> What I'm experiencing is that during certain call volumes, many calls
> go thru from box1 to box2. However, there are some cases where I get
> this message:
>
> Dec  6 11:11:19 WARNING[203]: chan_sip.c:9525 handle_response_invite:
> Forbidden - wrong password on authentication for INVITE to
> '"5095551212" <sip:5095551212 at 10.0.0.1>;tag=as3e387d65'
>
> and the caller gets busy signal. However, other callers calling the
> same number get thru with no problems. Why is this?
>
> Thanks,
> Waldo
>
> On Dec 5, 2005, at 10:30 AM, Waldo Rubinstein wrote:
>
> > This worked perfectly.
> >
> > Thanks,
> > Waldo
> >
> > On Dec 5, 2005, at 4:32 AM, xcel wrote:
> >
> >>
> >> Try this
> >>
> >> ___________________________________
> >> 1st Machine sip.conf
> >>
> >> [box2]
> >> username=box1
> >> type=friend
> >> host=10.0.0.2
> >> secret=*****
> >>
> >> in extensions.conf
> >>
> >> exten => _XXXXXX,1,Dial(SIP/box2/${EXTEN})
> >>
> >> __________________________________
> >> 2nd Machine sip.conf
> >>
> >> [box1]
> >> username=box2
> >> type=friend
> >> host=10.0.0.1
> >> secret=*****
> >>
> >> in extensions.conf
> >> exten => _XXXXX,1,Dial(SIP/box1/${EXTEN})
> >>
> >> --xce
> >>
> >>
> >> *********** REPLY SEPARATOR  ***********
> >>
> >> On 12/5/2005 at 12:11 AM Waldo Rubinstein wrote:
> >>
> >>> I have 2 Asterisk servers running 1.2.0. One of them is a PSTN
> >>> gateway. Currently they are connected using IAX2. I wanted to play
> >>> with SIP.
> >>>
> >>> I setup a sip entry (type=friend) in the PSTN gateway box and a sip
> >>> entry (type=user) in the second box in order to send calls using SIP
> >>> to the second box. This works fine. However, when I setup the second
> >>> box as type=friend in order for it to be able to send calls back to
> >>> the gateway box, then calls no longer work from gateway box to the
> >>> second box. The reported error is:
> >>>
> >>> Dec  5 00:07:14 NOTICE[203]: chan_sip.c:9514 handle_response_invite:
> >>> Failed to authenticate on INVITE to '"2125551212" <sip:
> >>> 2125551212 at 10.0.10.1>;tag=as0698b1b9'
> >>>
> >>> In the gateway box, my sip.conf looks like this:
> >>>
> >>> [general]
> >>> allowguest=yes
> >>> autocreatepeer=no
> >>>
> >>> [secondbox]
> >>> type=friend
> >>> host=10.0.0.2
> >>> secret=mysecret
> >>>
> >>> In the second box, my sip.conf looks like this:
> >>>
> >>> [general]
> >>> allowguest=yes
> >>> autocreatepeer=no
> >>>
> >>> [secondbox]
> >>> type=user
> >>> host=10.0.0.1
> >>> secret=mysecret
> >>>
> >>> Any ideas on how to correctly set this up?
> >>>
> >>> Thanks,
> >>> Waldo
> >>> _______________________________________________
> >>> --Bandwidth and Colocation provided by Easynews.com --
> >>>
> >>> Asterisk-Users mailing list
> >>> To UNSUBSCRIBE or update options visit:
> >>>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
> >>
> >> _______________________________________________
> >> --Bandwidth and Colocation provided by Easynews.com --
> >>
> >> Asterisk-Users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051206/4ac18f17/attachment.htm


More information about the asterisk-users mailing list