[TriLUG] SIP Softphone on Suse and Asterisk

[Reginald Reed]

Sorry if I missed this earlier in the thread, but what is what does
your network look like?  Are the clients on the same subnet as the
Asterisk server?  Are you traversing a NAT gateway somewhere in
between?

Also, it'll probably help if you make your configuration files
available with passwords replaces with asterisks (pun intended)!

--Reggie

On 12/29/06, Trevor Little <trevormg19 at gmail.com> wrote:
> I tried with iptables stopped on both the server and client with no luck.
>
> I am noticing a WARNING on the sip debug in the asterisk console. I'm
> wondering if maybe I'm having NAT problems becuas eof the reference here to
> 192.168.0.3
>
> Dec 29 21:02:42 WARNING[27540]: chan_sip.c:1228 retrans_pkt: Maximum retries
> exceeded on transmission ardyoyddhteqojd at 192.168.0.3 for seqno 209
> (Non-critical Response)
> Destroying call 'ardyoyddhteqojd at 192.168.0.3'
> 12 headers, 0 lines
> Reliably Transmitting (NAT) to 68.17.172.79:50157:
> OPTIONS sip:tlittle.creanga.homeip.net at 192.168.0.3 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK5fca1efa;rport
> From: "asterisk" <sip:asterisk at 192.168.0.100>;tag=as4efb2285
> To: <sip:tlittle.creanga.homeip.net at 192.168.0.3>
> Contact: <sip:asterisk at 192.168.0.100>
> Call-ID: 751c8cf2512406744d3f97991e2062d9 at 192.168.0.100
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Sat, 30 Dec 2006 02:03:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
>
> I tried the ethereal trace but am unfamiliar with the protocol so I'm not
> sure if there's anything out of the ordinary there.
>
> Trevor
>
> On 12/28/06, jonc at nc.rr.com <jonc at nc.rr.com> wrote:
> >
> > If that doesn't turn the trick, you can turn on Debuging in SIP inside
> > of Asterisk (sip debug on). SIP passes info back and forth a lot like
> > email. With that you should be able see what the server thinks is going
> > on.
> >
> > Also you can run Ethereal on the Linux box and see what packets are
> > going to/from your soft phone application.
> >
> > Jon Carnes
> >
> > ----- Original Message -----
> > From: Kevin Otte <nivex at nivex.net>
> > Date: Thursday, December 28, 2006 6:48 pm
> > Subject: Re: [TriLUG] SIP Softphone on Suse and Asterisk
> > To: Triangle Linux Users Group discussion list <trilug at trilug.org>
> >
> > > Trevor Little wrote:
> > > > Hello All,
> > > >
> > > > This is my first time posting to this list. I hope to make it to
> > > the next
> > > > meeting.
> > >
> > > Most excellent.  The Asterisk talk in here has been far too little
> > > of late.
> > >
> > > > I have Asterisk running on a P3 centos box at home. When I try to
> > > place a call
> > > > to it from my laptop (SUSE 10.1) using Twinkle I don't get any
> > > sound.> ...
> > > >
> > > > That's the same thing that it shows when  I call from a windows
> > > machine so I'm
> > > > pretty sure that the server is working.
> > > >
> > > > Does anyone have experience withthis kind of problem
> > > > ...
> > >
> > > I have little experience with SuSE, but my gut feeling is that the
> > > firewall is blocking the traffic.  Try doing a 'service iptables stop'
> > > before you start up your client.  If the call succeeds with iptables
> > > off, we can start looking at what rules you need to add.  If it
> > > doesn't, we'll have to try another approach.
> > >
> > > -- Kevin
> > > --
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