[asterisk-users] Hairpinning problems using IAX2 and SIP
Andrew Joakimsen
joakimsen at gmail.com
Mon Nov 6 11:14:01 MST 2006
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
> > > _______________________________________________
> > > --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/20061106/f6ecfed3/attachment.htm
More information about the asterisk-users
mailing list