[ncherry]

I'm very confused by what I'm seeing with my Asterisk server. I have the following configuration:

register => cp09876543:065432@voip.callpacket.com:5060/18585551212

[callpacket-out]
  type          = peer
  username      = cp09876543
  secret        = 065432
  port          = 5060
  fromuser      = "Neil Cherry"
  host          = voip.callpacket.com
  dtmfmode      = inband 
; dtmfmode      = rfc2833
  insecure      = very ; To allow registered hosts to call without re-authenticating 
  canreinvite   = no
  disallow      = all
  allow         = ulaw,gsm
; nat           = yes


[callpacket-in]
  type          = friend
  host          = voip.callpacket.com
  dtmfmode      = rfc2833
; context       = from-callpacket
  context       = from-sip-pstn
  insecure      = very ; To allow registered hosts to call without re-authenticating 
; nat           = yes

And my extension has the following:

[to-callpacket]
  exten => _396.,1,SetCallerId("Neil Cherry" <18585551212>)
  exten => _396.,2,Dial(SIP/${EXTEN:3}@voip.callpacket.com)
  exten => _396.,3,Congestion

When I dial 396xxxxxxxxxxxx I get (via the sniffer) a 407 Proxy Authentication required. The weirdest part is that I never see Asterisk register with voip.callpacket.com.

Anyone see anything obviously wrong?

[ncherry]

Ah, found out why registration wasn't working, it's in an #include file. Needs to be in the sip.conf file. That's one step closer. Now I'm seeing stuff from pinger@telepacket.com (???) and I saw some MD5 stuff. My login page knows it an Asterisk server!

Sip Proxy :             voip.callpacket.com
Sip Username :          cp09876543
Sip Password :          065432
Call Packet Soft Phone: Offline
Asterisk PBX:           Online
Phone Number:           +18585551212
TFTP Server:            67.43.156.90

More work to do!

[chandave]

Don't try to change your outgoing CID information. You can't influence the outbound CID info. Callpacket controls your outbound CID at their Call switch. You can only change the CID info by using their Callpacket dashboard. Changing what you present to Callpacket for CID info screws up the validation of your SIP INVITE. The fromuser should reflect your SIP Username.

See ya...

d.c.

[ncherry]

Thanks that seemed help with the incoming call (I made a couple of changes at the same time). I've removed the SetCallerId from the extension. I've changed the fromuser to my cpxxxxxxxxxx id. I've saved the files, restarted the sniffer and Asterisk. The first registration is 401 the second 200 (don't know why I see that). When I attempt to call out I get a 407 (proxy auth). When I call my 858 number I see a now see 200 (possibly seeing "Got 200 OK on REGISTER that isn't a register") and it goes to 'all circuits are busy now' I have no idea what's going on there (my Asterisk or Callpacket). So I have some more work to do. Considering I have no clue as to what I'm doing I'm making good progress ;-)

[ncherry]

Well I really am confused, the context I used only had exten => s, (1-4, 102-104 etc) but jumped into my extension section where _1 was defined (I think I had 2 contexts, I forgot to comment the second one out). So it was dialing out the PSTN (from Callpacket -> Asterisk -> SPA3000 -> PSTN). Anyway I've corrected that now and I can properly receive call from Callpacket. I still haven't resolved the 407 issue and I'm not sure how to yet. I wouldn't be surprised if I had someting else screwed up, anyway more digging.