One way Audio

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One way Audio

Postby mettichi » Fri Jan 15, 2016 1:01 pm

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
mettichi
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Re: One way Audio

Postby ambiorixg12 » Fri Jan 15, 2016 2:30 pm

You havent post your sip configuration file, follow this thread viewtopic.php?f=1&t=96518&sid=835e684ef3c943584975d328b75b0d5f#p215005
ambiorixg12
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Location: Dominican Republic

Re: One way Audio

Postby mettichi » Sat Jan 16, 2016 2:44 am

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
mettichi
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Joined: Thu Oct 09, 2008 6:25 am

Re: One way Audio

Postby david55 » Sat Jan 16, 2016 3:05 am

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.
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Re: One way Audio

Postby mettichi » Sat Jan 16, 2016 4:36 am

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
mettichi
Newsterisk
 
Posts: 7
Joined: Thu Oct 09, 2008 6:25 am


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