[asterisk-users] Hairpinning problems using IAX2 and SIP

[hugolivude]

Thanks for responding.

Yes I am doing pretty much exactly what you showed.  When I try to
dial without answering, I get a busy tone on the DiD (the local Telco
offers to let them notify me when it becomes available).  Sometimes I
get half a ring on the destination cell phone b4 receiving the busy
signal.

Not sure whether this sheds any light.  I'm going to have to answer
the call anyway in order to implement an auto attendant, but it was
worth trying.

Cheers,
H


*CLI>     -- Accepting AUTHENTICATED call from <ip>:
       > requested format = ulaw,
       > requested prefs = (),
       > actual format = g729,
       > host prefs = (g729),
       > priority = mine
    -- Executing Dial("IAX2/<ITSP>-1", "IAX2/<ITSP>/6135551234") in new stack
    -- Called <ITSP>/6135551234
    -- Call accepted by <ip> (format g729)
    -- Format for call is g729
    -- IAX2/<ITSP>-2 is making progress passing it to IAX2/<ITSP>-1
    -- Hungup 'IAX2/<ITSP>-2'
  == Spawn extension (incoming-iax, 6135551234", 1) exited non-zero on
'IAX2/<ITSP>1'
    -- Executing Hangup("IAX2/<ITSP>-1", "") in new stack
  == Spawn extension (incoming-iax, h, 1) exited non-zero on 'IAX2/<ITSP>-1'
    -- Hungup 'IAX2/<ITSP>-1'



On 11/4/06, Andrew Joakimsen <joakimsen at gmail.com> wrote:
> When you say you answer the call, I assume you have something like this:
>
> exten => 5551212,1,Answer
> exten => 5551212,1,Dial(SIP/provider/10005551212)
>
> Try to not answer the call and see if the behviour changes, it could just be
> your ITSP configuration....
>
>
> On 11/4/06, hugolivude <hugolivude at gmail.com> wrote:
> >
> > Asterisk 1.2.7
> > RedHat 9.0
> >
> > I frequently have the need to redirect calls that come in on a DiD
> > provisioned by my ITSP, back to the ITSP so that they can terminate
> > the call on the PSTN.  For example when an external call comes in, I
> > often have to send it to a cell phone.  I believe that this is
> > referred to as "hairpinning" the call.
> >
> > I do this by answering the incoming call and then I use a simple
> > "dial" command to send it back to my ISTP using a SIP or IAX channel
> > and the ITSP terminates it on the cell phone.    One of my main goals
> > is to keep my Asterisk box out of the media path and let the ITSP
> > handle all the provisioning for the call.  I understand that the
> > default behaviour of the "dial" command is supposed to do just that,
> > but I've run into problems though on both SIP & IAX channels.
> >
> > With IAX I use a simple dial command:
> >
> >    Dial(IAX2/myIAX/7775551234)
> >
> > Things seem to work great, I can see the handshaking in the CLI as the
> > call gets redirected and once both end points are connected, I can
> > actually disconnect my box from the ethernet and the call is
> > uninterruoted.  Unfortuanately the call quality is terrible!  Low
> > volume, choppy and so on.
> >
> > It seemed to me that since I had stepped my * box out of the network,
> > the problem must be with the  ITSP.  They suggested I try SIP.
> >
> > With SIP I use:
> >
> >    Dial(SIP/7775551234 at mySIP)
> >
> > Unfortuantely I don't get the handshakes and the whole call ends up
> > passing through my box, which is something I'm desperate to avoid.  I
> > have canreinvite=yes as seen from my sip.conf:
> >
> >    [mySIP]
> >
> >    type=peer
> >
> >    auth=md5
> >
> >    username=<UID>
> >    fromuser=<UID>
> >    fromdomain=<domain>
> >
> >    secret=<pw>
> >    host=<domain>
> >
> >    port=5060
> >
> >    nat=yes
> >
> >    canreinvite=yes
> >
> >    qualify=no
> >
> >    disallow=all
> >
> >    allow=g729
> >
> >    dtmfmode=rfc2833
> >
> >    insecure=very
> >    context=incoming-sip
> >
> >
> > Now the questions:
> >
> > 1) Given that I can see the handshaking and I can disconnect my * box
> > during the call, I think that the IAX call quality problems are on my
> > ITSP's end, but I could be wrong.  Is there anything I can do to
> > improve call quality when using IAX this way?
> >
> > 2) What about SIP?  Why doesn't that work?  I always thought that
> > "dial" would do exactly what I'm after (hairpin/redirect the call) if
> > I avoided options like t or T.
> >
> > Any direction you can provide is highly appreciated.
> >
> > Thanks,
> > H
> > _______________________________________________
> > --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
>
>
>