Hi,
A bit weird. I tried the same thing in linphonec and it worked, after
I set a stun server. So, I tried setting one up in sflphone, but it
still gives me that error. I prefer sflphone to linphonec though, and
linphonec crashes whenever anything interesting happens.
Sflphone is a free, gpl sip and iax2 compatible softphone. It uses a
client/server model (the core handles calls, audio, settings and other
stuff) while the clients connect to the core and give it commands.
There's supposed to be a console interface for it, but I have yet to
find one.
I was also able to do this from an asterisk prompt
dialplan add extension 100,1,Dial,SIP/linux-speakup.org/3000 into default
console dial 100
That works, but I can't do anything else while asterisk is using my
sound card. Grrrr.
Interestingly, this is the only time I've had this problem. I've been
able to talk to other people just fine, even without setting a stun
server. I'm breakingbigd at ekiga.net if anyone wants to sip me, not
online right now though.
Thanks,
KJ4UFX/AG
{.i doi .tcikoritys. mi cuxna ba'e do}
On 1/29/11, Igor Gueths <igueths at lava-net.com> wrote:
> Hello Derek and all. Most likely that 502 error is due to a proxy issue, in
> that
> it is timing requests out for some reason, hence the busy signal.
> https://secure.wikimedia.org/wikipedia/en/wiki/SIP_Responses for a nice
> reference of SIP protocol responses. Hope this helps.
> On Sat, Jan 29, 2011 at 06:56:16AM -0500, Kirk Reiser wrote:
>> Hello Derek: I am not familiar with sflphone but I have two questions
>> which may sound pretty silly so please be tolerant: is the port the
>> package is using set to 5060? Do you have to specify a protocol
>> string with that client such as sip:3000 at linux-speakup.org? What you
>> did should have worked so I'm a tad confused. If you've already
>> checked those angles possibly give us some more info about your
>> client.
>>
>> On Fri, 28 Jan 2011, Derek Roberts wrote:
>>
>> >Hi,
>> >
>> >I'm trying to call 3000 at linux-speakup.org with sflphone, but it says
>> >502 - Bad Gateway and gives a busy signal. What's up?
>> >
>> >Thanks.
>> >
>> >On 1/28/11, Igor Gueths <igueths at lava-net.com> wrote:
>> >>Well, in SIP, the thing you really have to worry about most is the IP
>> >>information that is contained within the packet's payload itself, since
>> >> that
>> >>is
>> >>really the deciding factor in call negotiation. Basically what would
>> >> happen
>> >>in
>> >>these nstances is that in Twinkle's twinkle.log file, I'd see packets
>> >> going
>> >>from
>> >>my laptop (192.168.1.44), back to 192.168.1.44 even though my Asterisk I
>> >> was
>> >>trying to reach is at 192.168.1.43. For this reason, even though at the
>> >>Twinkle
>> >>CLI I'd type call sip:300 at 192.168.1.43 (a test extension I have here), I
>> >>just
>> >>get put back at a "twinkle>" prompt with no discernible activity
>> >> occurring
>> >>at
>> >>the console.
>> >>On Fri, Jan 28, 2011 at 01:33:27PM -0500, Kitty Litter wrote:
>> >>>I am trying to learn more about the sip protocol and started using
>> >>>tcpdump. Could you tell me what options you use? It isn't alwalys
>> >>>clear to me the source and destination of the packets, I use -A and
>> >>>-t and -c count.
>> >>>
>> >>>_______________________________________________
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>> >>>
>> >>>--
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>> >>>
>> >>
>> >>--
>> >>Igor
>> >>
>> >>--
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>> >>_______________________________________________
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>> >>http://speech.braille.uwo.ca/mailman/listinfo/speakup
>> >>
>> >_______________________________________________
>> >Speakup mailing list
>> >Speakup at braille.uwo.ca
>> >http://speech.braille.uwo.ca/mailman/listinfo/speakup
>> >
>>
>> --
>> Kirk Reiser The Computer Braille Facility
>> e-mail: kirk at braille.uwo.ca University of Western Ontario
>> phone: (519) 661-3061
>> _______________________________________________
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>
> --
> Igor
>
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> dangerous content by MailScanner, and is
> believed to be clean.
>
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