[asterisk-users] Hairpinning problems using IAX2 and SIP
Andrew Joakimsen
joakimsen at gmail.com
Mon Nov 6 11:14:41 MST 2006
Sorry for the added traffic, but I had to add, that this was working even
after the call is answered.
On 11/6/06, Andrew Joakimsen <joakimsen at gmail.com> wrote:
>
> Why don't you try 1.12.13? Besides fixing some security issues, it should
> work, I doing a SMB install this weekend (so no concern of billing/CDR
> really) so we didn't use notransfer=yes and it worked perfectly, even behind
> NAT.
>
> Also you should contact your ITSP, maybe they don't allow this?
>
> On 11/5/06, hugolivude <hugolivude at gmail.com > wrote:
> >
> > 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
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