[asterisk-users] Problem receiving audio with Asterisk 1.4.5 with SIP trunk

[Rafael Canchola]

Hi

You can tray to making a tcpdump for detect where stay the audio 
packets (RTP) and/or stopping the iptables.
Also you may check the out route (route -n) and get the default GW, 
it should be the Public GW.




At 09:17 a.m. 26/11/2007, Zaheer K. Master wrote:
>Hi All,
>
>I'm running asterisk 1.4.5 (on AsteriskNOW beta 6 appliance) and using Snom
>360/370 phones with direct SIP trunking from bandwidth.com. I can make
>outgoing calls, and the person on the receiving end can hear my voice, but I
>cannot hear them. I also cannot receive incoming calls to my DID number.
>
>Here is my current setup:
>Asterisk is running on a dell poweredge server with an ip of 192.168.1.55
>I have setup a 1:1 NAT for asterisk with my public IP of 72.127.218.XXX
>The phones have IPs of 192.168.1.150-160
>I can register my phones and they work correctly for intercom, voicemail,
>etc.
>
>I think the problem is that after the SIP session has initiated, the phones
>are giving an IP of 192.168.1.151 for the return audio, and those packets
>are getting dropped. I'm not sure where to go from here to get the incoming
>calls/audio working. Do I have to give the Asterisk box a public IP? I tried
>this, and when I did I was unable to get the phones to register - probably
>since they had private IPs.
>
>Any help or suggestions would be greatly appreciated. Thanks in advance!
>
>Regards,
>Zaheer
>
>
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----------
RafaelCanchola
Product Development Engineer,
FonetGlobal Inc.
rcm at fonetglobal.com
http://www.fonetglobal.com
Ph. + 52 800 022 10 21 ext. 214
       + 52 442 167 08 00
VoIP 523663899
d00d! cyberalph
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