I think u need to add the
fromuser=peername
in u r sip.conf
Thanks
lloyd
On 2/6/07, David Sfiligoi <[EMAIL PROTECTED]> wrote:
Hi Dave,
Nothing really special in the console. The calls does go through, and it
is anwsered by presumably a les.net robot which is the voice of a woman
saying that the peer was not configured properly. I guess you could say I
get audio in one direction, but it is difficult to confirm wheter the robot
understand what I say ;-). I forgot to mention that I can actually receive a
call on the peer. voice goes both direction.
set verbose 20
Verbosity was 18 and is now 20
-- Executing Dial("SIP/2201-40ce", "SIP/lesnet_peer/19053164746") in
new stack
-- Called lesnet_peer/19053164746
-- SIP/lesnet_peer-8350 answered SIP/2201-40ce
-- Attempting native bridge of SIP/2201-40ce and SIP/lesnet_peer-8350
== Spawn extension (default, 619053164746, 1) exited non-zero on
'SIP/2201-40ce'
I have checked again the sip settings and dialplan, everything looks
good. I have also attempted to debug using sip debug on the peer, only
thing I noticed and rectified was that my firewall was blocking the rtp
ports. I ran into the same issue while setting up iax. I managed to debug it
with the iax debug communication traces. It boiled down to setting up an
entry in iax matching the peer number as a type=user.
To respond to Lloyd, I could talk to Les, but really it works in iax so
why should I bother him, as far as I am concerned it is not a "production"
problem. I am asking the community because I am curious to find the
solution. I am sure other people uses les and they can validate settings
and assumptions.If I don't get anywhere with our toronto community, I will
definitely ask les.
Thanks
David
On Tue, 6 Feb 2007 10:16:44 -0500
"Dave Donovan" <[EMAIL PROTECTED]> wrote:
> Hi David,
>
> Do you get an error message on the Asterisk console? It's often helpful
to
> see what's happening by turning on a verbose logging mode with "set
verbose
> 20" or something like that. Consider posting that here, it's often
helpful
> in debugging.
>
> Do you get audio in one direction?
>
> I haven't used Les.net but maybe if you are able to post some more info,
> someone on the list will be able to help. Personally most of my "SIP
> connection problems" have boiled down to a typo in an IP address or the
> dialplan or something.
>
> Dave
>
> On 2/5/07, David Sfiligoi <[EMAIL PROTECTED]> wrote:
> >
> > Hi,
> >
> > Based on an earlier thread where some taug users les.net was pretty
good
> > I decided to give it a try. I have been banging my head against this
problem
> > for a while since probably it is my own problem. So far no problem
> > receiving phone call when configured on both SIP and IAX. However, I
cannot
> > seem to make outbound phone call work on SIP(it does woek on IAX).
WhenI
> > phone I get some lday saying that the peer is incorectly
configured. I have
> > tried many things(including what les recommends), just can't seem to
get
> > it. Using some other voip company on sip I have no issue on sip.
Anybody
> > has some sample config that works well?(please remove your personal
info if
> > you post your config) Alternatively anybody that can help.
> >
> > Here the relevant sip.conf
> >
> > [lesnet_peer]
> > type=friend
> > host=did.voip.les.net
> > canreinvite=no
> > dtmfmode=rfc2833
> > insecure=very
> > disallow=all
> > allow=ulaw
> > allow=gsm
> > nat=yes
> > username=peername
> > secret=itssecret
> >
> >
> > register => peername:[EMAIL PROTECTED]/peername
> >
> > and extensions.conf
> >
> > exten => _6NX.,1,Dial(SIP/lesnet_peer/${EXTEN:1})
> > exten => _6NX.,2,Congestion
> > exten => _61NX.,1,Dial(SIP/lesnet_peer/${EXTEN:1})
> > exten => _61NX.,2,Congestion
> >
> > thanks,
> > David
> >
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> >
> >
>
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