[Asterisk-Users] (Resend) Trouble with all linux sip softphones.... And asterisk/linphone/kphone SRPMs

[Tom Ivar Helbekkmo]

John Morris <asterisk at butchwax.com> writes:

> Running FC1, ThinkPad T22, headset thru the soundcard.  Asterisk is
> asterisk-1.0_RC1.  No NAT.  The phones I've tried so far are as follows.

I've got NetBSD-current on a Thinkpad X31, ear plugs connected to the
built-in sound card, using integral microphone in the laptop.  Running
Asterisk from CVS, freshly updated.  The only soft phone I've tried on
this machine is kphone, which works very well for me.

> ** kphone:  Check out my SRPM at http://www.bigu.org/SRPMS/
>   Sound is fine.  Picks up sound from the microphone, but the echo-test
> repeats it back after passing it through a Mr-Roboto filter.

I have no problems with sound quality in either direction with kphone.

> ** SJphone:
>   Last night:  sound worked fine.  Actually sent sound from the mic,
> which came back after about a 5-second delay, but which sounded quite
> good.

This happens from time to time for me with kphone.  It's outgoing
sound (from kphone to Asterisk) that's being delayed, as far as I can
tell, and yes, it's about 5 seconds.  When this happens, I just hang
up and try again, and everything is fine.

>   Today:  establishes a connection, but absolutely no sound in or out. 

I was plagued with this too, initially, but figured out that it's
usually one of two causes: either you've got codec trouble (which can
be analyzed by invoking Asterisk with -vvv or thereabout), or it's a
firewall problem.  When changing sip.conf to adjust allowed codecs,
remember that you need to restart Asterisk to be sure the change will
"take" -- a 'sip reload' will not always do the right thing.

-tih
-- 
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no  T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145