[TriLUG] SIP Softphone on Suse and Asterisk
trevormg19 at gmail.com
Fri Dec 29 21:34:53 EST 2006
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
Dec 29 21:02:42 WARNING: chan_sip.c:1228 retrans_pkt: Maximum retries
exceeded on transmission ardyoyddhteqojd at 192.168.0.3 for seqno 209
Destroying call 'ardyoyddhteqojd at 192.168.0.3'
12 headers, 0 lines
Reliably Transmitting (NAT) to 188.8.131.52:50157:
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
Date: Sat, 30 Dec 2006 02:03:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
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.
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
> 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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