TRY THIS Vitelity incoming calls fail

[The Deacon]

It was time to rebuild the old Incredible PBX box, so I just built a spankin' new Incredible PBX 11 install and got all the old call logs/etc ported over. Inbound GV calls and outbound calls (using GV, voip.ms and Vitelity) all work GREAT, I'm scratching my head when it comes to any inbound calls from Vitelity.

in the Asterisk console, I see this whenever I call one of my Vitelity hosted numbers:

== Using SIP RTP CoS mark 5
  -- Executing [7075551212@from-sip-external:1] NoOp("SIP/64.2.142.13-00000001", "Received incoming SIP connection from unknown peer to 7075551212") in new stack
  -- Executing [7075551212@from-sip-external:2] Set("SIP/64.2.142.13-00000001", "DID=7075551212") in new stack
  -- Executing [7075551212@from-sip-external:3] Goto("SIP/64.2.142.13-00000001", "s,1") in new stack
  -- Goto (from-sip-external,s,1)
  -- Executing [s@from-sip-external:1] GotoIf("SIP/64.2.142.13-00000001", "0?checklang:noanonymous") in new stack
  -- Goto (from-sip-external,s,5)
  -- Executing [s@from-sip-external:5] Set("SIP/64.2.142.13-00000001", "TIMEOUT(absolute)=15") in new stack
  -- Channel will hangup at 2013-06-29 09:16:55.463 PDT.
  -- Executing [s@from-sip-external:6] Answer("SIP/64.2.142.13-00000001", "") in new stack
      > 0xb7141e18 -- Probation passed - setting RTP source address to 64.2.142.13:17020
  -- Executing [s@from-sip-external:7] Wait("SIP/64.2.142.13-00000001", "2") in new stack
  -- Executing [s@from-sip-external:8] Playback("SIP/64.2.142.13-00000001", "ss-noservice") in new stack
  -- <SIP/64.2.142.13-00000001> Playing 'ss-noservice.gsm' (language 'en')
== Spawn extension (from-sip-external, s, 8) exited non-zero on 'SIP/64.2.142.13-00000001'
  -- Executing [h@from-sip-external:1] Hangup("SIP/64.2.142.13-00000001", "") in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/64.2.142.13-00000001'

The phones/trunks are online:

pbx*CLI> sip show peers
Name/username            Host                                    Dyn Forcerport ACL Port    Status      Description
701/701                  192.168.100.30                          D              A  5061    OK (9 ms)
702/702                  192.168.100.100                          D              A  5060    OK (14 ms)
703/703                  192.168.100.115                          D              A  5060    OK (206 ms)
vitel-inbound/acctname    64.2.142.214                                N            5060    Unmonitored
vitel-outbound/acctname  64.2.142.216                                N            5060    Unmonitored
voipms/XXXXXX            67.215.241.250                              N            5060    Unmonitored
6 sip peers [Monitored: 3 online, 0 offline Unmonitored: 3 online, 0 offline]

pbx*CLI> sip show registry
Host                                    dnsmgr Username      Refresh State                Reg.Time
losangeles.voip.ms:5060                N      XXXXXX            105 Registered          Sat, 29 Jun 2013 12:18:27
inbound7.vitelity.net:5060             N      acctname        45 Registered          Sat, 29 Jun 2013 12:19:21
2 SIP registrations.

Any suggestions as to what I'm missing?

-Rick

 

[briankelly63]

set the context in your PEER trunk configuration to context=from-trunk as it should be .. perhaps you had anonymous allowed in the previous setup.

 

[The Deacon]

I didn't move over any of the settings for the trunks or extensions (configured all of them from scratch). The only things I moved over were the blacklist database as well as the Asteridex and call logs database.

But I did try that, Brian - but still no joy.

Here are the PEER details from that trunk:

username=accountname
type=friend
trustrpid=yes
sendrpid=yes
secret=secret_password
host=inbound1.vitelity.net
fromuser=accountname
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw
insecure=very

 

[Trimline2]

Try this:

username=accountname
type=friend
secret=secret_password
insecure=port,invite
host=inbound28.vitelity.net
dtmfmode=auto
context=from-trunk
canreinvite=no
allow=all

Also, make sure via the online page that your inbound host has not changed. This can happen.

 

[MGD4me]

This probably isn't going to fix your problem, but where you have insecure=very, change it to: insecure=port,invite.

 

[briankelly63]

right about now somebody usually mentions that oh by the way I changed the router ..... Anything new in that regard?

Believe it or not different VOIP providers have equipment that sometimes handles NAT and SIP header packets differently. So based on your router setup one provider make work and another may not.

 

[mp1111]

I didn't move over any of the settings for the trunks or extensions (configured all of them from scratch). The only things I moved over were the blacklist database as well as the Asteridex and call logs database.

But I did try that, Brian - but still no joy.

Here are the PEER details from that trunk:

username=accountname
type=friend
trustrpid=yes
sendrpid=yes
secret=secret_password
host=inbound1.vitelity.net
fromuser=accountname
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw
insecure=very

Can you add the following lines under user details:

nat=yes
qualify=yes
qualifyfreq=120
qualifysmoothing=yes
callevents=yes
insecure=invite&very
pedantic=no

Please try and let me know.

Thanks,

 

[The Deacon]

right about now somebody usually mentions that oh by the way I changed the router ..... Anything new in that regard?

No router / hardware changes.

Try this:

username=accountname
type=friend
secret=secret_password
insecure=port,invite
host=inbound28.vitelity.net
dtmfmode=auto
context=from-trunk
canreinvite=no
allow=all

Also, make sure via the online page that your inbound host has not changed. This can happen.

This was the issue! The inbound host HAD changed. Ergh.

THANKS, Trimline2!