Number You Have Dialed Is Unavailable

[rockinthesixstring]

I have a problem with my PBX. I dont know if ALL callers experience this issue, but when I call the system from my mobile phone, I get “The Number You Have Dialed Is Unavailable”.

I currently have an account with CallCentric where they give me a single DID and then I can purchase multiple phone numbers that all operate off of that DID. If I can remember correctly, I had this working before when I called from my Other mobile phone (Bell Mobility), but now when I call from my iPhone (Rogers)… it doesn’t work.

Here is my asterisk trace (did changed for privacy)

[quote]Connected to Asterisk 1.4.21.1 currently running on pbx (pid = 2591)
Verbosity is at least 3
== Connect attempt from ‘192.168.0.195’ unable to authenticate
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Connect attempt from ‘192.168.0.195’ unable to authenticate
– Executing [17771112222@from-sip-external:1] NoOp(“SIP/5060-0a041960”, “Received incoming SIP connection from unknown peer to 17771112222”) in new stack
– Executing [17771112222@from-sip-external:2] Set(“SIP/5060-0a041960”, “DID=17771112222”) in new stack
– Executing [17771112222@from-sip-external:3] Goto(“SIP/5060-0a041960”, “s|1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/5060-0a041960”, “0?from-trunk|17771112222|1”) in new stack
– Executing [s@from-sip-external:2] Set(“SIP/5060-0a041960”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2008-08-16 01:22:10 UTC.
– Executing [s@from-sip-external:3] Answer(“SIP/5060-0a041960”, “”) in new stack
– Executing [s@from-sip-external:4] Wait(“SIP/5060-0a041960”, “2”) in new stack
– Executing [s@from-sip-external:5] Playback(“SIP/5060-0a041960”, “ss-noservice”) in new stack
– <SIP/5060-0a041960> Playing ‘ss-noservice’ (language ‘en’)
== Spawn extension (from-sip-external, s, 5) exited non-zero on ‘SIP/5060-0a041960’
– Executing [h@from-sip-external:1] NoOp(“SIP/5060-0a041960”, “Hangup”) in new stack
– Executing [h@from-sip-external:2] Set(“SIP/5060-0a041960”, “DID=s”) in new stack
– Executing [h@from-sip-external:3] Goto(“SIP/5060-0a041960”, “s|1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/5060-0a041960”, “0?from-trunk|s|1”) in new stack
– Executing [s@from-sip-external:2] Set(“SIP/5060-0a041960”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2008-08-16 01:22:15 UTC.
– Executing [s@from-sip-external:3] Answer(“SIP/5060-0a041960”, “”) in new stack
== Spawn extension (from-sip-external, s, 3) exited non-zero on ‘SIP/5060-0a041960’[/quote]

And here is my current extensions_custom.conf (phone numbers changed for privacy)

; This file contains example extensions_custom.conf entries.
; extensions_custom.conf should be used to include customizations
; to AMP’s Asterisk dialplan.

[incoming]
exten => s,1,Set(Var_TO=${SIP_HEADER(TO)})
exten => s,2,GotoIf($[“${Var_TO}” = “sip:[email protected]”]?extension1,s,1:3)
exten => s,3,GotoIf($[“${Var_TO}” = “sip:[email protected]”]?ext-did,15555551111,1:4)
exten => s,4,GotoIf($[“${Var_TO}” = “sip:[email protected]”]?ext-did,15555552222,1:5)
exten => s,5,GotoIf($[“${Var_TO}” = “sip:[email protected]”]?ext-did,15555553333,1:6)
exten => h,6,Macro(hangupcall)

If I call any one of those phone numbers from my mobile phone… I get that “unavailable” recording.

Any help is greatly appreciated!!!

[hraynor]

I was having exactly the same problem.

I am using an AudioCodes MP-104 FXO gateway with an AT&T line. The AT&T line has Caller ID capability.

When I setup the gateway to NOT deliver Caller ID, everything worked fine, and Asterisk calls the from-trunk context and everything works without issues.

However, when I told the gateway TO deliver Caller ID (which it correctly does in the From line of the INVITE message), it drops to the from-sip-external context and things then do not work…

SOLUTION was to ensure that I put a context=from-trunk line in PEER details for Outgoing Settings (formerly I only had this in Incomning Settings for USER DETAILS) for the Trunk…

Not sure why this worked here, but appears to have solved things and Caller ID can now be delivered…

So my PEER Detials is now:

host=192.168.0.240
allow=ulaw
type=peer
context=from-trunk

And my USER Details is:

allow=ulaw
type=user
context=from-trunk

[fskrotzki]

how about posting a good call trace please. Saying it’s the same when you both will have different systems and version of FreePBX does not help. Also there are errors in the above call trace that make things very unclear as the phone numbers do not match so I can see why it fails from the provided trace at the top (i.e. 1777… will NOT match 1555…).

[hraynor]

Thanks for the quick reply. Actually found the solution right after posting that. I edited the first message to now state that the solution was to add context=from-trunk in the PEER Settings for Outgoing Settings. Not sure why it mattered there, but this fixed it…

[rockinthesixstring]

is it possible that not enough system resources could cause this behavior? I am just testing the software so I have put it on a computer that isn’t very powerful.

[Teacher1965]

help i am new to freepbx,I have an Audiocodes MP-118 Fxo * Port and i want to use it for my trunks to the pstn can anyone help me with the config info on audiocode with freepbx