[asterisk-users] One Way Audio Problem

[Steve Totaro]

canreinvite defaults to yes, whether specified or not.

http://www.voip-info.org/wiki/view/tips

If you follow these directions adapting to your particular
circumstances and it doesn't work, post your whole sip.conf

Start asterisk with verbose set to 3 or so and turn on sip debugging.
I get somewhere in the debug, you will see local NAT IPs that don't
belong there, or it will just work.

Thanks,
Steve Totaro

On Thu, Oct 16, 2008 at 12:12 AM, Steve Totaro
<stotaro at totarotechnologies.com> wrote:
> Change all canreinvites to no.
>
>
>
> On Wed, Oct 15, 2008 at 9:37 PM, GNUbie <gnubie at gmail.com> wrote:
>> Hello Karsten,
>>
>> On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer <kwem at gmx.de> wrote:
>>>
>>> Please post Your sip.conf.
>>> Which IP-Address do You configure in the snom for Your asterisk? (eth0
>>> or eth1)?
>>
>> The SNOM 300 is using the NET interface beside the DC 5V port to
>> connect to the LAN.
>>
>> The Asterisk box is using the eth1 to connect to the LAN.
>>
>> As per your instruction, below is my /etc/asterisk/sip.conf :
>>
>> - - - < s n i p > - - -
>>
>> [general]
>> realm=pbx.domain.com
>> bindport=5060
>> bindaddr=0.0.0.0
>> rtptimeout=60
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> allow=gsm
>> externip=pbx.domain.com
>> localnet=192.168.101.0/255.255.255.0
>> jbforce=yes
>> allowtransfers=yes
>> maxexpiry=3600
>> minexpiry=1800
>> videosupport=no
>>
>> [internal-phones](!)
>> type=friend
>> host=dynamic
>> context=family
>> dtmfmode=rfc2833
>> insecure=port,invite
>> canreinvite=no
>> nat=no
>> qualify=yes
>> port=5060
>>
>> [102](internal-phones)
>> username=102
>> secret=102
>> callerid="GNUbie"<102>
>> mailbox=102 at family
>>
>> - - - < s n i p > - - -
>>
>> Thank you in advance.
>>
>> Regards,
>>
>> GNUbie
>>
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>
>
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>



-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)