[Asterisk-Users] Connecting 2 Asterisk using SIP

[Waldo Rubinstein]

I tried G711 and GSM and in both cases call quality degraded when the  
softphone was conferencing more than 2 people (note: not a meetme room).

- Waldo

On Dec 7, 2005, at 5:45 AM, xcel wrote:

> I did use IAX2 but sound quality wasn't that good which codec are  
> you using with IAX2 ?
>
>
> *********** REPLY SEPARATOR ***********
>
> On 12/6/2005 at 9:22 PM Alvaro Parres wrote:
> 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
> >>> _______________________________________________
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> >>
> >>
> >>
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