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rphenix

901 posts

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#20385 24-Mar-2008 18:36
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Figured I was doing something dumb on my vanilla asterisk box so I've temporarily setup trixbox on an old laptop but still cant get things to work.

Making calls outwards is fine but inbound dont work work I get the following whenever I try and register (where MyVFXNumber = without leading 0)
:

Reliably Transmitting (NAT) to 58.28.20.150:5060:
REGISTER sip:as.wxcnz.net SIP/2.0
Via: SIP/2.0/UDP LANIP:5060;branch=z9hG4bK21ef4d27;rport
From: <sip:register=MyVFXNumber@as.wxcnz.net>;tag=as753146ae
To: <sip:register=MyVFXNumber@as.wxcnz.net>
Call-ID: 19bca31c54183f097dc38d0837cdefa6@127.0.0.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:MyVFXNumber@LANIP>
Event: registration
Content-Length: 0


---

<--- SIP read from 58.28.20.150:5060 --->
SIP/2.0 404 Not found
From: <sip:register=MyVFXNumber@as.wxcnz.net>;tag=as753146ae
To: <sip:register=MyVFXNumber@as.wxcnz.net>;tag=1429232167-1206335151738
Call-ID: 19bca31c54183f097dc38d0837cdefa6@127.0.0.1
CSeq: 102 REGISTER
Via: SIP/2.0/UDP LANIP:5060;received=MyPubStaticIP;rport=5060;branch=z9hG4bK                                    21ef4d27
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '19bca31c54183f097dc38d0837cdefa6@127.0.0.1' Method                                    : REGISTER


sip show registry produces:
as.wxcnz.net:5060               register=995       120 Request Sent

Any ideas?



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sbiddle
29266 posts

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  #118428 24-Mar-2008 18:52
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Do incoming calls hit your Asterisk box or go to VFX voicemail?

Do you have an incoming route set to a valid extension/group/IVR menu?

rphenix

901 posts

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  #118433 24-Mar-2008 19:32
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I've kept it simple with my routes any inbound call goes to my  extension.

The call goes straight to voicemail if I enable voicemail on my VFX account.


 
 
 
 


jedney
39 posts

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#118603 25-Mar-2008 18:01
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I'm no expert on Asterisk, so I may be miles out on this one.. but in your snipit you have

Contact: <sip:MyVFXNumber@LANIP>

I believe this should actually be
 
Contact: <sip:MyVFXNumber@WANIP>

I seem to remember I had to put my real world IP in my sip.conf file as Asterisk is behind a NAT router. Whilst on the subject of Routers, have you set your's up to port forward 5060 TCP/UDP and 10000-20000 UDP to go to the LAN IP of your Asterisk box?

Hope that's of some help.

rphenix

901 posts

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  #118613 25-Mar-2008 18:52
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jedney: I'm no expert on Asterisk, so I may be miles out on this one.. but in your snipit you have


No such thing as a bad response :)

Regarding WAN IP I thought about that but... I thought the line:

Reliably Transmitting (NAT)  meant it should be fine?

I might define the PUB IP Address anyway (it is static).

Rob

richms
23652 posts

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  #118614 25-Mar-2008 18:57
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For me to get asterisk to woerk with 2talk I had to put my wan ip in the sip config meaning I needed a static IP address. other providers worked ok with the wrong ip in there since clearly they ignored the ip in the payload.

Would be nice if they would get stun support in asterisk when running as a client since you cant always guarentee a 1:1 incoming port mapping in anycase when behind nat, but sadly it seems that asterisk developmenet is at a snails pace.




Richard rich.ms

rphenix

901 posts

Ultimate Geek


  #118622 25-Mar-2008 20:27
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Ok this time pub ip is in the sip register command but same difference :(

Double checked my firewall rules look right on the openwrt:
UDP ports 10000:20000 forwarded to asterisk box
UDP ports 5060:5082 forwarded to asterisk box
TCP Port 5060 forwarded to asterisk (probably not required).

rphenix

901 posts

Ultimate Geek


  #118657 25-Mar-2008 22:32
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Got it sorted, not sure what it was now, setup asterisk again still had a problem then powered off everything (openwrt router, dsl modem, switch etc..) things work now so really not sure what it was should have tried that earlier I guess!

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