I have just set up my first Asterisk, and I have the basics up an running.
I have set it up with two SIP phones only. I can call between them, and I
can call out to FWD phones. However, on receiving calls from FWD, my
Asterisk blocks the calls with the following message:
May 30 20:19:24 NOTICE[
Add "insecure=very" to your FWD peer context in sip.conf.
The "in-between fix" really wasn't a fix. Chan_sip was modified some time
ago
(5/24 or so) to require authentication for inbound calls also. To turn this
required authentication off, you need to add "insecure=very" to your peer
definition.
Am 30.05.2004 um 21:35 schrieb Thor Atle Rustad:
However, on receiving calls from FWD, my Asterisk blocks the calls
with the following message:
May 30 20:19:24 NOTICE[180236]: chan_sip.c:6397 handle_request: Failed
to authenticate user . Obviously, I
want FWD users to be able to call me withou
At 21:51 30.05.2004, you wrote:
Am 30.05.2004 um 21:35 schrieb Thor Atle Rustad:
However, on receiving calls from FWD, my Asterisk blocks the calls with
the following message:
May 30 20:19:24 NOTICE[180236]: chan_sip.c:6397 handle_request: Failed to
authenticate user . Obviously, I want
FWD us
Hi!
Thor, you are right here, the problem has crept back into CVS! :-(
In between it had been fixed, though.
Cheers, Philipp
> >>However, on receiving calls from FWD, my Asterisk blocks the calls with
> >>the following message:
> >>
> >>May 30 20:19:24 NOTICE[180236]: chan_sip.c:6397 handle_re
On 30/05/2004 at 21:35 Thor Atle Rustad wrote:
>I have just set up my first Asterisk, and I have the basics up an
>running.
>I have set it up with two SIP phones only. I can call between them, and I
>can call out to FWD phones. However, on receiving calls from FWD, my
>Asterisk blocks the calls w
Andy Powell wrote:
Anything that's added to * that breaks how protocols work should be by default OFF not ON,
but that's just IMO...
I agree 100%, this has been very frustrating trying to work out why
Asterisk suddenly stopped accepting calls from FWD and other PSTN based
services.
I very much
On Mon, 2004-05-31 at 10:16, Duane wrote:
> Andy Powell wrote:
>
> > Anything that's added to * that breaks how protocols work should be by default OFF
> > not ON,
> > but that's just IMO...
>
> I agree 100%, this has been very frustrating trying to work out why
> Asterisk suddenly stopped acc
Eric Wieling wrote:
The problem is that too many people had a context= in [general] and
didn't realize that incoming SIP calls that didn't match anything would
be accepted and sent to the context= that was specified in [genera].
I wonder where they picked that up from, default config perhaps?
--
Be
The problem is that too many people had a context= in [general] and
didn't realize that incoming SIP calls that didn't match anything would
be accepted and sent to the context= that was specified in [genera].
Thats the way we prefer it (the old way). Its nice to be able to
publish a sip phone
On 31/05/2004 at 10:47 Eric Wieling wrote:
>On Mon, 2004-05-31 at 10:16, Duane wrote:
>> Andy Powell wrote:
>>
>> > Anything that's added to * that breaks how protocols work should be by
>default OFF not ON,
>> > but that's just IMO...
>>
>> I agree 100%, this has been very frustrating trying to w
*** REPLY SEPARATOR ***
On 31/05/2004 at 11:13 Andres wrote:
>Thats the way we prefer it (the old way). Its nice to be able to
>publish a sip phone number to anybody out there(for example I can just
>say that my number is sip:[EMAIL PROTECTED]). When the
>call comes into
Eric Wieling wrote:
On Mon, 2004-05-31 at 10:16, Duane wrote:
Andy Powell wrote:
Anything that's added to * that breaks how protocols work should be by default OFF not ON,
but that's just IMO...
I agree 100%, this has been very frustrating trying to work out why
Asterisk suddenly s
Andy Powell wrote:
On 31/05/2004 at 10:47 Eric Wieling wrote:
On Mon, 2004-05-31 at 10:16, Duane wrote:
Andy Powell wrote:
Anything that's added to * that breaks how protocols work should be by
default OFF not ON,
but that's just IMO...
I agree 100%, this has
Brian Cuthie wrote:
Except that I *want* anyone to be able to call me directly from the
Internet. That's the whole point -- we're trying to remove the necessity
for a phone-company-like entity in the middle.
Instead, I suggest setting the default context for sip to something like
"sip-incoming
Eric Wieling wrote:
On Mon, 2004-05-31 at 10:16, Duane wrote:
Andy Powell wrote:
Anything that's added to * that breaks how protocols work should be by default OFF not ON,
but that's just IMO...
I agree 100%, this has been very frustrating trying to work out why
Asterisk suddenly stopped accepti
> The "in-between fix" really wasn't a fix. Chan_sip was modified some time
> ago
> (5/24 or so) to require authentication for inbound calls also. To turn
this
> required authentication off, you need to add "insecure=very" to your peer
> definition.
5/24 is some time ago? The age we live in!
-
17 matches
Mail list logo
