[asterisk-users] One Way Audio Problem

[Steve Totaro]

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
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