Voice drops 1 second after call connection

newboy

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Mar 11, 2009
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#1
Hi

I have recently installed Elastix 1.6, after i had hack attacks due to upgrading to FreePBX version 2.6. I did not realize the sip settings under it allows the anonymous calls to come through. I had Fail2ban installed as well. All my settings were changed by the hacker and i was not able to connect to the PBX at all.

I formatted the system and installed the version 1.6, 32 bit. Installed all the updates to bring it up to FreePBX 2.6. Disabled the SIP settings in FreePBX.

My issue is:

I am unable to make calls between remote extensions. I am able to make calls between the local extensions. But when i call between the remote extensions the voice drops after 1 sec of connection and my internet connection also drops at the router end. I have to reboot the router to get the connection back up.

I have opened the following ports
TCP/UDP: 5060
UDP: 10000 to 20000 verified in rtp.conf the port numbers are right

Another thing i have noticed is that my extensions are not connecting through port 5060. Even the local extensions are not connecting through port 5060.

Name/username Host Dyn Nat ACL Port Status
XXXX/XXXX 123.456.789.123 D N A 32768 UNREACHABLE
XXXX (Unspecified) D N A 0 UNKNOWN
XXXX/XXXX 123.456.789.123 D N A 60850 UNREACHABLE
XXXX/XXXX 192.168.0.9 D N A 31446 OK (115 ms)

The two remote extensions come up as OK (XX ms) initially when i connect but after making a call between them they turn to UNREACHABLE.

When i try to check the sip channels while the call is active, it shows that the call used ulaw.


This is what my sip_general_custom.conf file looks like:

language=au
videosupport=yes
allow=g729
allow=g723
allow=h261
allow=h263
allow=h263p

defaultexpirey=600
maxexpirey=3600
rtptimeout=60
rtpholdtimeout=120

alwaysauthreject=yes

useragent = Elastix


This is how my sip_nat.conf file looks like:

nat=yes
externip=XXX.XXX.XXX.XXX
localnet=10.1.0.0/255.255.0.0
externrefresh=10


Any suggestions on what could be wrong here?
 

fasilkaks

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Feb 6, 2010
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#2
hi newboy,
what trunk are you using for remote connectivity - sip or iax?
 

fasilkaks

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#3
one more thing...
could you paste the trunk settings here? both remote and local
 

newboy

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#4
I am using a sip trunk. It is the same extensions that i am using remotely and or locally.

here is the configuration for 1 of them:

This device uses sip technology.

secret
dtmfmode: rfc2833
canreinvite: no
context: from-internal
host: dynamic
type: friend
nat : yes
port : 5060
qualify : yes
callgroup
pickupgroup
disallow
allow
dial : SIP/XXXX
accountcode:
mailbox : XXXX@device
deny : 0.0.0.0/0.0.0.0
permit: 0.0.0.0/0.0.0.0
 

fasilkaks

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#5
why dont you go for IAX trunks...its very easy to setup and stable. I am using it for many months without failure.
 

newboy

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#6
The IP phones i have only support SIP protocols :( so i can't use IAX extensions
 

fmvillares

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#7
did you have the pbx in external public ip? did you have sip alg firewall enabled controlling that ip? do you use iptables or hardware firewaall, did you test if fail2 ban configs are correct to not block asterisk connections?
remember that sip and nat are close enemies...
 

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