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13 posts

Geek
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Topic # 106551 26-Jul-2012 11:53
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Hi All,

I have the the openVFX service from Xnet and I seem to be getting audio problems intermittently,

My DSL is DMZ'd to my asterisk box, and the firewall is set to just allow any incoming connection from the WXC server (58.28.20.150) and sip/ssh/rtp from anyone (minus a couple of IP blocks from spammers), and anything outgoing is allowed

The system was working perfectly however one day the outgoing audio to WXC just stopped working, Extension to Extension worked fine, and Extension to the Echo service in asterisk worked fine, calls via WXC would dial/ring and I could hear the person, but they could not hear me. even after I changed the dial plan so all incoming calls answered and echoed, I still had no audio.

This also happened a few weeks ago and came right the next day (after a call to wxc tech support who said nothing was wrong) but now this time it doesn't seem to be fixing itself...

Anyone else experience this problem? currently it is working fine after setting up my PAP2T to go direct to WXC

I am running Linux 2.6 with Asterisk 1.6.2 and I have hand made/clean configs (no gui's etc)
I have 1x line registered with Xnet, and 4 SIP extensions, Asterisk is on a separate DSL to the 4x extensions

Naturally this is frustrating so any leads with this would be appreciated,

Regards,
Blair

My SIP/RTP configs are:

/etc/asterisk/sip.conf
[general]
port=5060
bindport=5060
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=gsm
nat=yes
qualify=yes
registertimeout=1200
regseconds=1800
allowguest=no
alwaysauthreject=yes
localnet=10.1.1.0/255.255.255.0
externip=[my public ip]
context=guestincomming
allowoverlap=no
canreinvite=yes

register = [my vfx registration]

[VFX]
type=peer
fromuser=[my number]
host=pan.wxnz.net
insecure=invite,port
canreinvite=yes
secret=[my password]
username=[my username]
context=incomming

[601]
type=peer
host=dynamic
secret=[ext password]
context=usersall
mailbox=601@default

[602]
type=peer
host=dynamic
secret=[ext password]
context=usersall
mailbox=601@default

... (2 more extensions)

/etc/init.d/rtp.conf
[general]
rtpstart=35384
rtpend=37384

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174 posts

Master Geek

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  Reply # 662435 26-Jul-2012 15:21
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Hello,

Looking at a couple example calls I can see you aren't sending us any audio at all. Does the router in front of the box have a SIP ALG on it ? As the most likely cause is going to be the audio not getting through your router. Your first step would be to verify the audio is leaving the asterisk box then work upstream from there.




Nicholas Cuc

Network Support
WorldxChange Communications
www.xnet.co.nz





13 posts

Geek
+1 received by user: 1


  Reply # 662450 26-Jul-2012 15:32
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The router is pretty basic and doesn't have any VoIP options at all so had to go with DMZ, whats weird is it works fine to my external extensions, and that it worked fine for about 2 months, then one day just stopped working. I'm pretty sure its not a firewall/dsl issue because incoming audio works fine. and enabling rtp debug shows that its sending the audio (both to wxc and to my extension) it just doesn't seem to get there

 
 
 
 


4997 posts

Uber Geek
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Subscriber

  Reply # 662598 26-Jul-2012 19:50
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Also remove GSM codec.




Chorus has spent $1.4 billion on making their xDSL broadband network faster. If your still stuck on ADSL or VDSL, why not spend from $150 on a master filter install to make sure you are getting the most out of your connection?
I install - Naked DSL, DSL Master Splitters, VoIP, data cabling and general computer support for home and small business.
Rural Broadband RBI installer for Ultimate Broadband and Full Flavour

 

Need help in Auckland, Waikato or BoP? Click my email button, or email me direct: [my user name] at geekzonemail dot com




13 posts

Geek
+1 received by user: 1


  Reply # 685538 13-Sep-2012 13:04
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Well my vfx line came right the next day as if nothing was wrong (with the exception of me removing the GSM codec) and it has been working flawlessly ever since. but yesterday and today I seem to be back to having no outgoing audio.. any thoughts? RTP debug shows I'm sending the audio data, .. maybe the internet is stealing it haha

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