Outbound calls to sip peer fail

When I try to dial outside, I get the following in the Asterisk console:

[Dec 15 00:33:13] WARNING[3163]: chan_sip.c:5441 create_addr: Purely numeric hostname (), and not a peer–rejecting!
[Dec 15 00:33:13] WARNING[3163]: app_dial.c:2274 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)

I am set up with VoicePulse, and started with their sample sip.conf and extensions.conf. The peer is registered, and internal dialing works correctly.

Here’s the relevant portion of the extensions.conf (really simple, just trying for a POC first)

exten => _1NXXNXXXXXX,1,Dial(SIP/+${EXTEN}@${voicepulse-primary})
exten => _1NXXNXXXXXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN},GatewayB)
exten => _1NXXNXXXXXX,n(GatewayB),Dial(SIP/+${EXTEN}@${voicepulse-backup})

And here’s the info from my sip.conf:

[voicepulse-primary]
type=peer
context=voicepulse-in
host=sjc-primary.voicepulse.com
defaultuser=xxxxxxxx
secret=xxxxxxxx
qualify=yes
allow=all
canreinvite=no
dtmfmode=rfc2833
rfc2833compensate=yes
insecure=port,invite
trustrpid=yes

[voicepulse-backup]
type=peer
context=voicepulse-in
host=sjc-backup.voicepulse.com
defaultuser=xxxxxxxx
secret=xxxxxxxx
qualify=yes
allow=all
canreinvite=no
dtmfmode=rfc2833
rfc2833compensate=yes
insecure=port,invite
trustrpid=yes

Any thoughts/help would be greatly appreciated.

What is in the globals section of your extensions.conf? Also running with verbosity 3 would give a clue.

The value of ${voicepulse-primary} is incorrect. I am also not sure if “-” is a permitted character.