I don't know whats the prob. Getting circuit-busy error.

[chill_master]

I freshly installed a new vicidialnow box. My VOIP provider uses IP authentication. So I tried to configure it. And I got this error everytime I try to call outbound.
=>
SIP/216.xxx.xxx.xxx-09f2db40 is circuit-busy
Everyone is busy/congested at this time (1:0/1/0)

Here is the CLI output when I call:

== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-70ff,2", "8600051|F") in new stack
> Channel Local/8600051@default-70ff,1 was answered.
-- Executing AGI("Local/8600051@default-70ff,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-70ff,1", "SIP/216.xxx.xxx.xxx/1832xxxxxxx||tTor") in new stack
We're at 121.xxx.xxx.xxx port 37364
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 13 lines
Reliably Transmitting (no NAT) to 216.xxx.xxx.xxx:5060:
INVITE sip:1832xxxxxxx@216.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 121.xxx.xxx.xxx:5060;branch=z9hG4bK13b8aa54;rport
From: "M1001181956000000001" <sip:0000000000@121.xxx.xxx.xxx>;tag=as30ecc5a5
To: <sip:1832xxxxxxx@216.xxx.xxx.xxx>
Contact: <sip:0000000000@121.xxx.xxx.xxx>
Call-ID: 1598d88654f6306e5fff11311b098367@121.xxx.xxx.xxx
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M1001181956000000001" <sip:0000000000@121.xxx.xxx.xxx>;privacy=off;screen=no
Date: Thu, 01 Oct 2009 23:19:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 2512 2512 IN IP4 121.xxx.xxx.xxx
s=session
c=IN IP4 121.xxx.xxx.xxx
t=0 0
m=audio 37364 RTP/AVP 18 0 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 216.xxx.xxx.xxx/1832xxxxxxx
vici*CLI>
<-- SIP read from 216.xxx.xxx.xxx:5060:
SIP/2.0 404 Number not in e164 format, example +12125551212
Via: SIP/2.0/UDP 121.xxx.xxx.xxx:5060;branch=z9hG4bK13b8aa54;rport=5060
From: "M1001181956000000001" <sip:0000000000@121.xxx.xxx.xxx>;tag=as30ecc5a5
To: <sip:1832xxxxxxx@216.xxx.xxx.xxx>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.c575
Call-ID: 1598d88654f6306e5fff11311b098367@121.xxx.xxx.xxx
CSeq: 102 INVITE
Server: Bandwidth.com TRM (bw7.gold.13)
Content-Length: 0


--- (8 headers 0 lines) ---
Transmitting (no NAT) to 216.xxx.xxx.xxx:5060:
ACK sip:1832xxxxxxx@216.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 121.xxx.xxx.xxx:5060;branch=z9hG4bK13b8aa54;rport
From: "M1001181956000000001" <sip:0000000000@121.xxx.xxx.xxx>;tag=as30ecc5a5
To: <sip:1832xxxxxxx@216.xxx.xxx.xxx>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.c575
Contact: <sip:0000000000@121.xxx.xxx.xxx>
Call-ID: 1598d88654f6306e5fff11311b098367@121.xxx.xxx.xxx
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M1001181956000000001" <sip:0000000000@121.xxx.xxx.xxx>;privacy=off;screen=no
Content-Length: 0


---
-- SIP/216.xxx.xxx.xxx-09f2db40 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("Local/8600051@default-70ff,1", "") in new stack
== Spawn extension (default, 91832xxxxxxx, 3) exited non-zero on 'Local/8600051@default-70ff,1'
-- Executing DeadAGI("Local/8600051@default-70ff,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Destroying call '1598d88654f6306e5fff11311b098367@121.xxx.xxx.xxx'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-70ff,2'
-- Executing DeadAGI("Local/8600051@default-70ff,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1


Anybody who can help me?

[heinz]

Hi,
could be a codec problem. What codec does your provider use?
What is output of these commands?
sip show registry
show translation

Thanks,
heinz

[chill_master]

Show translation:

Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)

g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc
g723 - - - - - - - - - - -
gsm - - 2 2 2 2 1 2 6 18 -
ulaw - 2 - 1 2 2 1 2 6 18 -
alaw - 2 1 - 2 2 1 2 6 18 -
g726 - 2 2 2 - 2 1 2 6 18 -
adpcm - 2 2 2 2 - 1 2 6 18 -
slin - 1 1 1 1 1 - 1 5 17 -
lpc10 - 2 2 2 2 2 1 - 6 18 -
g729 - 2 2 2 2 2 1 2 - 18 -
speex - 2 2 2 2 2 1 2 6 - -
ilbc - - - - - - - - - - -


My voip provider uses ulaw and g729a.

Sip show registry:
Host Username Refresh State
216.xxx.xxx.xxx:5060 1832xxxxxx 105 Registered

[chill_master]

I've tried setting g729 alone in sip.conf, and also ulaw alone but I got same error.

[chill_master]

one issue might be in the dial plan right? coz of this one:

<-- SIP read from 216.xxx.xxx.xxx:5060:
SIP/2.0 404 Number not in e164 format, example +12125551212
Via: SIP/2.0/UDP 121.xxx.xxx.xxx:5060;branch=z9hG4bK13b8aa54;rport=5060

My provider said they accept only E.164 number patterns.
In my dial plan, I use:

exten => _91NXXNXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,tTor)

Dial prefix is 9 in my campaign setting.

[chill_master]

Solve.

It's with the formatting of the number. I called my voip provider and had them change the format to 11 digit format and remove the + prefix.

Now on to inbound.