On Fri, 22 Oct 2004 14:11:53 +0300, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> thanx, now it works..
Good. Remember, debug is always your best friend. Other feedback may
lie to you (ie "everybody is busy" is most often a lie and it really
means "No clue why it's not working") but debug will al
thanx, now it works..
I've set the both phones on auto-codecs ..
In fact the results are :
If both phones are on auto, g729, gsm610 or g711a i can hear each other.
If they are on different codecs "no hear". F.e.
phone1 <--> asterisk <--> phone2
if phone1 is set on g729 and phone2 auto, then pho
On Fri, 22 Oct 2004 09:47:21 +0300, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> hmm... now tring.. somone to know how can I redirect the output of the "sip debug"
> into file 'cause it is really hard to grasp (several pages is just one call)
if you are running a fairly recent version of asteris
[EMAIL PROTECTED] wrote:
> hmm... now tring.. somone to know how can I redirect the output of
> the "sip debug" into file 'cause it is really hard to grasp (several
> pages is just one call)
>
>
Try this..
http://www.voip-info.org/wiki-Asterisk+debugging
If you are having problems catching inte
hmm... now tring.. somone to know how can I redirect the output of the "sip debug"
into file 'cause it is really hard to grasp (several pages is just one call)
> On Fri, 22 Oct 2004 01:24:24 +0300, raptor <[EMAIL PROTECTED]> wrote:
> > On Fri, 22 Oct 2004 01:36:29 +0900
> > Benjamin on Asterisk
On Fri, 22 Oct 2004 01:24:24 +0300, raptor <[EMAIL PROTECTED]> wrote:
> On Fri, 22 Oct 2004 01:36:29 +0900
> Benjamin on Asterisk Mailing Lists <[EMAIL PROTECTED]> wrote:
>
> |The problem is SIP NAT traversal
> |
> |The solution is 'canreinvite=no'
>
> ]- no it is not, the phones are on the same
On Fri, 22 Oct 2004 01:36:29 +0900
Benjamin on Asterisk Mailing Lists <[EMAIL PROTECTED]> wrote:
|The problem is SIP NAT traversal
|
|The solution is 'canreinvite=no'
]- no it is not, the phones are on the same subnet :"(
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On Thu, 21 Oct 2004 17:09:46 +0300, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> I just setup two ip phones..
[snip]
>
> If I set canreinvite=no, then everyhtink works..
> what is the problem ?!? a solution
The problem is SIP NAT traversal
The solution is 'canreinvite=no'
for more information
I just setup two ip phones..
The problem is that when they connect trough asterisk nothing
is heard or sometimes garbage.. both phones are set on uLaw,
With or w/o allow=ulaw in sip.conf
the result is the same...
If I set canreinvite=no, then everyhtink works..
what is the problem ?!? a solution
