[asterisk-users] Ongoing problem with 1.8

Tilghman Lesher tilghman at meg.abyt.es
Tue Jan 18 12:33:51 CST 2011


On Tuesday 18 January 2011 11:31:07 Ira wrote:
> At 01:00 AM 1/18/2011, you wrote:
> >On Tuesday 18 January 2011 01:05:20 Ira wrote:
> > > I have tried installing many of the beta versions and most of the
> > > release versions of 1.8. I have 3 SIP phones which we use for all
> > > our calls. After installing 1.8 the first thing I try is calling
> > > out port one of my Digium TDM04 back into port 2. I can see that
> > > the call comes in and tries to call all three SIP phones but the
> > > phones never ring. Eventually the call goes to voice mail and these
> > > error messages pop up. I've read doc/sip-retransmit.txt and as far
> > > as I can tell, there's nothing there for me to try.
> > > 
> > > Is there anything else I might try or do to help troubleshoot this.
> >
> >Try running a tcpdump for udp port 5060 while this is occurring.  Also,
> >what type of SIP phones are you using?
> 
> Aastra 480i-CT phones.  Is "tcpdump port 5060" the syntax you'd like
> me to use?

Nope, "tcpdump 'udp port 5060'".

> And I may have neglected to point out, the same system has been
> running since 1.2.11 or so with basically no issues.

While that's a useful data point, it's not relevant to the problem.  A
significant portion of the SIP stack was re-implemented in 1.8, and Polycom
phones are on the desktops of nearly every Asterisk developer.  Since you
aren't using a Polycom, the SIP stack on that device is implemented
differently, causing possible incompatibilities.  This is why the tcpdump
will be helpful:  to figure out what is different and why it doesn't work.

-- 
Tilghman



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