[asterisk-users] Asterisk -> SIP-ROUTER -> Internet = no audio

[Yves Arikoglu]

thanks brian,

yes, i am aware that sip is only responsible for signalling and therefor 
my conclusion was, that it
has got something to do with nat / firewall / the router...
meanwhile i´ve got it solved... although the sip-provider tried to 
convince me, that the misconfiguration
is on my asterisks´ side, i penetrated the support until they looked 
over it again.... and... what should i
say... finally they had to admit, that the router had a wrong acesslist. 
they corrected it and now it works.

yves

Brian schrieb:
> On Fri, 2010-02-12 at 02:18 +0100, Yves Arikoglu wrote:
>   
>> Hi,
>>
>> I am breaking my fingers in configuring an asterisk (1.6) to 
>> successfully transmit audio with the following setup:
>>
>> asterisk, resides in local network, ip is 10.26.208.252
>> versatel business router (directly connected to a dsl, configured by 
>> sip-provider), WAN ip 89.244.13.25
>> versatel sip-proxy ip 89.244.13.10
>>
>>
>> in sip.conf I have:
>> [general]
>> bindaddr=0.0.0.0
>> externip=89.244.13.25
>> localnet=10.26.208.0/255.255.252.0
>> nat=yes
>> qualify=yes
>>
>>
>> the local sip phones register correctly and can make calls between each 
>> other with audio.
>> the local sip phones CAN make outbound calls via the sip-provider... 
>> will say, destination phone rings, but there is no audio (on both legs)
>> after pickup...
>> external phones can call my sip-number... the call comes into the 
>> asterisk, the sip-extension rings, but after pickup... no audio at all.
>> even if i route the call from external to a queue or something else... i 
>> see, that asterisk is playing voicefiles, but the caller does not hear
>> anything.
>> because sip-signalling works in any ways, but audio not, i think its got 
>> something to do with nat... but there is no firewall between asterisk
>> and the router or between the router and the internetconnection from 
>> versatel... and i already tried millions of combinations of using
>> nat=yes/no/route, qualify=yes/no, canreinvite=yes/no and and and and i´m 
>> stuck as i was never ever stuck before :-(((((
>>
>> any hints? anybody?
>>
>>     
> You are aware that SIP only sets up, monitors and takes the call down?
> The audio stream is RDP and on higher ports. My guess is that the audio
> stream on inbound calls is not arriving where it should be - or is
> blocked. This could be router or nat, but one thing jumps out to me:
> Does your Asterisk Server itself have something set up in the built in
> iptables firewall blocking udp inbound traffic in the port range
> 15000:20000? The output of the command 'iptables -nvL' will tell you
> pretty quickly.
>
> HTH.
>
>
>
>