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5 posts

Wannabe Geek


# 38634 2-Aug-2009 22:36
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Hi there, I've just changed my VFX connection over to an "asterisk" connection.

I've tried many different configurations (sip/extentions), but I'm not able to get incoming calls to route correctly through to the asterisk box.  Outgoing calls (asterisk to realworld) work fine.  Internal dialing seems to work fine.

Its calls from realworld to asterisk don't seem to reach the box.  (tcpdump shows no sip packets arriving). The box has a real ip address.  Seems to show as registered ok:

asterisk*CLI> sip show registry
Host                            Username       Refresh State                Reg.Time
pan.wxnz.net:5060        7981xxxx            165 Registered        Sun, 02 Aug 2009 22:22:20

Can anyone see anything wrong with my confs?

---sip.conf:
[General]
port = 5060
dtmfmode = rfc2833
disallow=all
;allow=g729  ; tried this in and out..
allow=ulaw
allow=alaw
registertimeout = 20
regseconds = 180
register => 7981xxxx:xxxxxxxxxxxxxxxxxxxx:xxxxxxxxxxxxxxxxxxxx@pan.wxnz.net/7981xxxx

[VFX]
type=peer
fromuser=7981xxxx
host=pan.wxnz.net
insecure=invite,port
canreinvite=no
nat=yes
secret=xxxxxxxxxxxxxxxxxxxx
username=xxxxxxxxxxxxxxxxxxxx
context=default
---

---extensions.conf:
[globals]
static=yes
writeprotect=no
clearglobalvars=no

[general]

[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

exten => 1234,1,Dial(SIP/xlite,30)
exten => 1234,n,Hangup()

exten => i,1,Answer()
exten => i,n,Playback(tt-weasels)
exten => i,n,Hangup()

; added this on in just in case... it wasn't picked up by s above...
exten => 7981xxxx,1,Answer()
exten => 7981xxxx,n,Playback(tt-weasels)
exten => 7981xxxx,n,Hangup()
---


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420 posts

Ultimate Geek
+1 received by user: 2


  # 241905 3-Aug-2009 17:54
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Check that you have an Incoming Route setup to handle the VFX line or at least an Any DID/Any CID incoming route. To handle the VFX line differently, define an incoming route specifically for the VFX trunk (minus the leading zero).

Sorry - reread your post again. You indicate that the SIP packets are not arriving. Have you established pin holes in your router's firewall for port 5060 directing them to your Asterisk box? I have pin holes for the SIP port and RTP plus IAX for external extensions.

Cheers Mike



5 posts

Wannabe Geek


  # 241979 3-Aug-2009 20:07
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Thanks for that, and thanks to Josh P.

I suspect the problem was that when I got the serviced moved from my SPA2102 adapter, it still contained the old registration on the second line. So despite the fact that it was failing to register and asterisk was thinking it was registered, it failed to send the call to the right place.

I'm now getting call notifications to asterisk. I just need to go and configure some dial plans.

Thanks,

S :)

 
 
 
 


5474 posts

Uber Geek
+1 received by user: 2543

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  # 242029 3-Aug-2009 21:33
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Ah, yes, always best to factory reset & re-provision devices when you have moved VFX lines to another device / system.




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