# One way Audio **URL:** \ **Category:** Asterisk Support\ **Created:** [January 15, 2016, 7:01pm UTC](https://community.asterisk.org/t/one-way-audio/64905 "2016-01-15T19:01:06Z")\ **Posts on this page:** 5\ **Page:** 1
**Author:** ![mettichi](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/mettichi/32/112_2.png) [@mettichi](https://community.asterisk.org/u/mettichi)\ **Post date:** [January 15, 2016, 7:01pm UTC](https://community.asterisk.org/t/one-way-audio/64905/1 "2016-01-15T19:01:06Z")
Hello, I have an Asterisk Server 11.13.1 installed on Centos, my server has 2 interfaces eth0 and eth1, eth0 with public IP and eth1 with private ip, i have created a sip account and trunk SIP, when i register with Public IP every thing is ok, calls are working good but when i try with eth1 ( private ip) rtp is going in one way ( i get one way audio), i have disabled iptables and i have opened rtp ports from 10000 to 20000 and i have disabled nat in the account sip( nat = no, dtmf=rfc2833, canreinvite =no), but still same problem, could you suggest ang modification to resolve this issue please in configuration, do you think if i make this configuration ( nat=yes, canrenvite = nonenat, dtmf=info) the problem will be resolved, another question please does i can put many realm in sip.conf because i need to register to publIC IP and private IP? Thanks in advance ---
**Author:** ![ambiorixg12](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/ambiorixg12/32/1887_2.png) [@ambiorixg12](https://community.asterisk.org/u/ambiorixg12)\ **Post date:** [January 15, 2016, 8:30pm UTC](https://community.asterisk.org/t/one-way-audio/64905/2 "2016-01-15T20:30:15Z")
You havent post your sip configuration file, follow this thread [viewtopic.php?f=1&t=96518&sid=835e684ef3c943584975d328b75b0d5f#p215005](http://forums.digium.com/viewtopic.php?f=1&t=96518&sid=835e684ef3c943584975d328b75b0d5f#p215005) ---
**Author:** ![mettichi](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/mettichi/32/112_2.png) [@mettichi](https://community.asterisk.org/u/mettichi)\ **Post date:** [January 16, 2016, 8:44am UTC](https://community.asterisk.org/t/one-way-audio/64905/3 "2016-01-16T08:44:13Z")
Hello, thanks a lot for your response this is my configuration: [66894672] accountcode=66894672 regexten=66894672 amaflags=billing canreinvite=no context=a2billing dtmfmode=RFC2833 host=dynamic insecure=port,invite language=fr nat=yes qualify=no rtptimeout=600 rtpholdtimeout=600 secret=xxxxxxxxx type=friend username=66894672 disallow=ALL allow=g729 allow=ulaw allow=alaw allow=g723 regseconds=0 cancallforward=no rtpkeepalive=0 SIP.CONF ( GENERAL SECTION) faxdetect=yes vmexten=\*97 context=from-sip-external callerid=Unknown notifyringing=yes notifyhold=yes tos\_sip=cs3 tos\_audio=ef tos\_video=af41 alwaysauthreject=yes useragent=FPBX-2.8.1(11.13.1) disallow=all allow=alaw allow=g723 allow=g726 allow=g729 allow=h264 allowtransfer=no match\_auth\_username=yes allowoverlap=yes realm=10.4.2.5 domain=10.4.2.5 rtsavesysname=no transport=udp udpbindaddr=0.0.0.0 bindport=5060 relaxdtmf=yes dtmfmode=rfc2833 tos\_sip=cs3 tos\_audio=ef tos\_video=af41 tos\_text=af41 cos\_sip=3 cos\_audio=5 cos\_video=4 cos\_text=3 allowexternaldomains=no domainsasrealm=yes domain=10.4.2.5(eth1) domain=public ip(eth0) rtcachefriends=yes pedantic=yes maxforwards=70 preferred\_codec\_only=yes progressinband=never compactheaders=no videosupport=yes maxcallbitrate=384 callevents=yes authfailureevents=yes shrinkcallerid=yes rtpkeepalive=0 sipdebug=no recordhistory=no dumphistory=no faxdetect=yes notifyringing=yes notifyhold=yes notifycid=yes callcounter=yes t38pt\_udptl=yes,fec,maxdatagram=400 directmedia=yes directrtpsetup=no ignoresdpversion=yes autodomain=no callerid=DNID language=en jbenable=no maxcallbitrate=384 minexpiry=60 defaultexpiry=120 t38pt\_udptl=yes g726nonstandard=no srvlookup=yes allowguest=no maxexpiry=3600 videosupport=yes canreinvite=no rtptimeout=600 rtpholdtimeout=600 rtpkeepalive=0 checkmwi=10 notifyringing=yes notifyhold=yes registertimeout=20 registerattempts=10 nat=yes externip=public ip localnet=127.0.0.0/255.0.0.0 Please Help ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [January 16, 2016, 9:05am UTC](https://community.asterisk.org/t/one-way-audio/64905/4 "2016-01-16T09:05:05Z")
You only have one sip.conf entry, whereas you problem description implies at least two (one for each interface). You have localnet and nat=yes, but no means to find your public address (externip, externhost, stunaddr, etc.) localnet is meaningless as it only includes the loopback interface, which would already be included. In particular, it doesn’t include the far side of your, presumably non-natted, eth1 network. Is eth0 really directly on a public address? You haven’t said which way is the one way. What have you configured in your routing tables? You have some obsolete or deprecated option names. ---
**Author:** ![mettichi](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/mettichi/32/112_2.png) [@mettichi](https://community.asterisk.org/u/mettichi)\ **Post date:** [January 16, 2016, 10:36am UTC](https://community.asterisk.org/t/one-way-audio/64905/5 "2016-01-16T10:36:31Z")
Hello, i fact i have configured this account SIP in an xlite on my local network witch is connceted directly to the eth1 interface of my server asterisk, when i call a number ( GSM number) because i have a trunk TDM ( SS7 signalling ), the called party hear me but i don’t hear him, when i use the same account with the same configuration and register the account using my public IP ( eth0), every thing is ok no problem, so in my server there are two interfaces ( eth0: public IP and eth1: private ip 10…) do you need any other information? Thanks in advance