Can't receive incoming calls

I recently built a new Asterisk/FreePBX server to replace a very old one we are running. I built the box from scratch using Ubuntu 9.10 and Asterisk/FreePBX version 2.8.x.x

I have everything configured; office to office and outbound calls work great. Unfortunately, my inbound calls don’t work at all. When someone makes an incoming call there is a few minutes of silence and then a fast busy signal. No entries show up in the log regarding the call but a FreePBX report shows the following:

Calldate            Channel       Source     Clid                       Dst   Description Duration

  1. 2010-11-05 08:20:18 SIP/vitel-… 7038080000 “UNAVAILABLE” <7038080000> tdial NO ANSWER 00:01
  2. 2010-11-05 08:20:18 SIP/vitel-… 7038080000 “UNAVAILABLE” <7038080000> tdial NO ANSWER 00:12
  3. 2010-11-05 08:20:18 SIP/vitel-… 7038080000 “UNAVAILABLE” <7038080000> tdial NO ANSWER 00:12
  4. 2010-11-05 08:20:18 SIP/vitel-… 7038080000 “UNAVAILABLE” <7038080000> tdial NO ANSWER 00:12
  5. 2010-11-05 08:20:17 SIP/vitel-… 7038080000 “UNAVAILABLE” <7038080000> tdial NO ANSWER 00:13
  6. 2010-11-05 08:20:17 SIP/vitel-… 7038080000 “UNAVAILABLE” <7038080000> tdial NO ANSWER 00:13
  7. 2010-11-05 08:20:17 SIP/vitel-… 7038080000 “UNAVAILABLE” <7038080000> tdial NO ANSWER 00:13
  8. 2010-11-05 08:20:16 SIP/vitel-… 7038080000 “UNAVAILABLE” <7038080000> tdial NO ANSWER 00:13

Does anyone have an idea what’s going on and how I can fix it?

BTW, we are using Vitelity as our DID service provider and I have placed some of my configuration info below.


From sip_additional.conf

[vitel-inbound]
type=user
dtmfmode=auto
host=inbound18.vitelity.net
context=ext-did
username=#######
secret=#######
allow=all
insecure=port,invite
canreinvite=no
qualify=yes

[vitel-outbound]
type=peer
dtmfmode=auto
host=outbound.vitelity.net
username=#######
fromuser=#######
trustrpid=yes
sendrpid=yes
secret=#######
allow=all
canreinvite=no
qualify=yes
context=from-trunk-sip-vitel-outbound


From sip_registrations.conf

register=#######:#######@inbound18.vitelity.net:5060


From extensions_additional.conf

[ext-did]
include => ext-did-custom
include => ext-did-0001
include => ext-did-0002
exten => foo,1,Noop(bar)

; end of [ext-did]

[ext-did-0001]
include => ext-did-0001-custom
exten => fax,1,Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)})

; end of [ext-did-0001]

[ext-did-0002]
include => ext-did-0002-custom
exten => fax,1,Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)})
exten => 7036511234,1,Set(__FROM_DID=${EXTEN})
exten => 7036511234,n,Gosub(app-blacklist-check,s,1)
exten => 7036511234,n,ExecIf($[ “${CALLERID(name)}” = “” ] ?Set(CALLERID(name)=${CALLERID(num)}))
exten => 7036511234,n,Set(__CALLINGPRES_SV=${CALLERPRES()})
exten => 7036511234,n,Set(CALLERPRES()=allowed_not_screened)
exten => 7036511234,n(dest-ext),Goto(ext-trunk,1,1)

; end of [ext-did-0002]


From extensions_custom.conf

[ext-did-custom]

; end of [ext-did-custom]

What does a CODEC smell like?

If you turn on SIP debug and you get a “not allowed here” message after the invite with the SDP then you have a CODEC issue.

You should add this to each trunk:

disallow=all
allow=ulaw

I would check which codec the carrier uses to connect to you and make sure you have the prompt files in the appropriate format… really smells like codecs…