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. > > > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
