SIP inbound is working outbound is not

General Help
1

Hello all,

We moved SIP service from SIPstation to Cox Communications. SIPstation was working fine but they were dinging us for using the Unlimited Plan too much (a big ? on that).

Cox installed a SIP gateway (for lack of a better term) on premises. The PBX shows successful registration to that device. Inbound calls work fine on the telephone numbers provided by Cox.

Outbound calls get an “All circuits are busy message” if outbound route is set to SIP trunk only

  • AsteriskNOW install
  • FreePBX 2.9.0.7
  • Asterisk 1.6.2.11
  • there are analog lines that work fine
  • if I put the analog trunk(s) in front of the SIP trunk, dialing works fine
  • if I put the SIP trunk in front of the analog trunk it seems to skip the SIP and dial out through analog
  • I un-installed the SIPstation module but no change
  • I did the asterisk -vcr and saw this error message: Got SIP response 604 “Does not exist anywhere”

From SIP trunk configuration:

PEER
host=10.0.0.19
type=peer
fromuser=XXXXXXXXXXX
username=XXXXXXXX
fromdomain=coxbusiness.com
dtmfmode=rfc2833
disallow=all
allow=ulaw&alaw
insecure=invite,port
context=from-trunk
nat=yes
canreinvite=no
secret=XXXXXXX

USER
dtmfmode=inband

Any ideas? I’ve been searching the web and found some references to configuring the dialplan but I’m not sure if that is the right path.

Regards,
Rich.

2

The config does not make any sense, you have credential then you have insecure set. Did you copy this from somewhere or create it yourself?

What are you trying to talk to?

3

Have to agree with SkyKing, this configurations makes little sense.

Try type=friend and delete the user entry.
Then nat=no Since you said “Cox installed a SIP gateway (for lack of a better term) on premises.” and your host is on an RFC-1918 local subnet. Then check the values in /etc/asterisk/sip_nat.conf.

Restart with an amportal stop followed by amportal start…

4

This was copied and pasted from Cox’s documentation according to their onsite tech.

I changed
nat=no
type=friend
wasn’t clear on the ‘delete the user entry’ so I removed username (still bad) removed fromuser (still bad) and USER dtmfmode=inband (still bad) - removed each one individually then ran amportal stop/start

5

Can you be more specific on the trying to talk to, do you mean the device Cox installed locally?

6

Can a couple people post their a copy of their working examples? I’m looking for the PEER and USER details. Obviously XXX out your specific details on fromuser and username. If you have a working Cox VoIP setup all the better!

Also, I have the ‘Asterisk SIP Settings’ module installed, do I need anything else?

Thanks,
Rich.

7

My guess is inbound is not really working to the peer, do you have anonymous SIP turned on?

8

Here’s the config that works. Mind you this is using Cox as an ITSP. They install an Edgewater EdgeMarc 4550 device in your network to talk to their VoIP backbone.

For your setup change:
host = IP address of your EdgeMarc 4550
username = given to you by Cox
fromuser = your 10 digit tele given to you by Cox
secret = given to you by Cox

PEER DETAILS:
type=peer
host=10.0.0.19
username=476111111111
fromuser=7025551212
dtmfmode=rfc2833
disallow=all
allow=ulaw&alaw
qualify=yes
insecure=port,invite
fromdomain=coxbusiness.com
context=from-trunk
canreinvite=no
secret=4767025551212
trustrpid=yes
sendrpid=yes
usereqphone=yes

Registration string:
[email protected]:4767025551212:[email protected]/4767025551212