Two asterisk servers circuit-busy

Hi,

We have got two Asterisk servers in differnet cities. Sometimes, when calling to another city, we get an error:

– Called lviv-peer/732
– SIP/lviv-peer-0caedd80 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)

What is the problem?

You will need SIP debug output for this, as various different conditions get mapped onto busy and congestion. However the most likely reason is that the number you are calling is busy!

Or maybe the sip registration fail.

No, that number is not busy, because the phone is near me =)
Here is debug:

[code]<------------->
— (14 headers 15 lines) —
Sending to 194.44.237.ХХХ : 37889 (NAT)
Using INVITE request as basis request - e4ea2458ad7cf0ae@10.10.20.107
Found user ‘750’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 99
Peer audio RTP is at port 10.10.20.107:5006
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format iLBC for ID 99
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.10.20.107:5006
Looking for 1732 in from-internal (domain 192.168.0.102)
list_route: hop: sip:750@10.10.20.107

<— Transmitting (NAT) to 194.44.237.ХХХ:37889 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.20.107;branch=z9hG4bKad32c9856000c4ad;received=194.44.237.ХХХ
From: “750” sip:750@192.168.0.102:5060;tag=6197b0685938c7b0
To: sip:1732@192.168.0.102:5060
Call-ID: e4ea2458ad7cf0ae@10.10.20.107
CSeq: 57218 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:1732@82.207.98.ХХХ
Content-Length: 0

<------------>
Audio is at 82.207.98.ХХХ port 15728
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP

<— Transmitting (NAT) to 194.44.237.ХХХ:37889 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.10.20.107;branch=z9hG4bKad32c9856000c4ad;received=194.44.237.ХХХ
From: “750” sip:750@192.168.0.102:5060;tag=6197b0685938c7b0
To: sip:1732@192.168.0.102:5060;tag=as358a4f11
Call-ID: e4ea2458ad7cf0ae@10.10.20.107
CSeq: 57218 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:1732@82.207.98.ХХХ
Content-Type: application/sdp
Content-Length: 208

v=0
o=root 3318 3318 IN IP4 82.207.98.ХХХ
s=session
c=IN IP4 82.207.98.ХХХ
t=0 0
m=audio 15728 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[/code]

There is no final status on that trace. The decision as to what category to use will be based on the final status, typically a 4xx or 5xx status for a failed call. 1xx ones are interim results and the call will not terminate on receiving them.