[TriLUG] SIP Softphone on Suse and Asterisk

Sat Dec 30 15:51:09 EST 2006

Guess I should have mentioned the network setup earlier.=)

My server is set up as the DMZ on my router so all ports are forwarded to it
be default. The client machine is my laptop so I am usually connecting from
a different sub-net (coffe shop, neighbors wireless, etc.).

Here are my sip.conf and extentions.conf:

[tlittle at creanga ~]$ cat /etc/asterisk/sip.conf
; **************************************************************
; ** GLOBAL SETTINGS FOR THE ASTERISK SIP CHANNEL
[general]
context=default          ; Default context for incoming calls
realm=creanga.homeip.net
bindport=5060            ; UDP Port to bind
bindaddr=0.0.0.0         ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes            ; Enable DNS SRV lookups on outbound calls
disallow=all             ; First disallow all codecs
;allow=ulaw               ; Allow codecs in order of preference
;allow=gsm
language=en              ; Default language

[tlittle]
type=friend
secret=******
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=default
allow = gsm
allow = ulaw

[alittle]
type=friend
secret=******
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=default
allow=gsm
allow=ulaw
[tlittle at creanga ~]$ cat /etc/asterisk/extensions.conf
[default]
exten => 100,1,Dial(SIP/tlittle)
exten => 200,1,Dial(SIP/alittle)

exten => 900,1,Answer()
exten => 900,n,Playback(hello-world)
exten => 900,n,Hangup()

On 12/30/06, Reginald Reed <reginald.reed at gmail.com> wrote:
>
> 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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