[asterisk-users] Help with sip debug?

[Shanon Swafford]

I always like to activate the syslog and debug on my SPA's.  Sometimes this
will tell you what they are doing.

Shanon



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rich Adamson
Sent: Wednesday, July 19, 2006 8:30 PM
To: Asterisk Users-List
Subject: [asterisk-users] Help with sip debug?



Need a little help trying to understand what's happening here.

spa941 -> Asterisk-A -> iax2 -> Asterisk-B -> spa942

When the spa941 (x3000) calls spa942 (x1004), the spa942 returns a "busy 
here" sip message. The spa942 is not busy and does not have DND or any 
other option set to cause a busy-here message. Asterisk-B is v1.2.10 
updated to current svn. (Seeing the exact same issue with an spa3k.)

A sip debug from Asterisk-B shows the following three packets:

localhost*CLI> sip debug peer 1004
SIP Debugging Enabled for IP: 160.80.40.201:5060   <== x1004
     -- Registered IAX2 to '151.213.193.101', who sees us as 
153.222.216.140:1963 with no messages waiting

     -- Accepting UNAUTHENTICATED call from 151.213.193.101:
        > requested format = gsm,
        > requested prefs = (g726|gsm|ilbc),
        > actual format = g726,
        > host prefs = (g726|gsm|ilbc),
        > priority = mine
     -- Executing Dial("IAX2/to-npi-3", "SIP/1004|15|r") in new stack
We're at 160.80.40.4 port 13382
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 160.80.40.201:5060:
INVITE sip:1004 at 160.80.40.201:5060 SIP/2.0
Via: SIP/2.0/UDP 160.80.40.4:5060;branch=z9hG4bK544dbabe;rport
From: "NPI-Rich" <sip:3000 at 160.80.40.4>;tag=as0e37bb22
To: <sip:1004 at 160.80.40.201:5060>
Contact: <sip:3000 at 160.80.40.4>
Call-ID: 176eea4944e5fd1f63179a042ba51c06 at 160.80.40.4
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 19 Jul 2006 22:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 18182 18182 IN IP4 160.80.40.4
s=session
c=IN IP4 160.80.40.4
t=0 0
m=audio 13382 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
     -- Called 1004
localhost*CLI>
<-- SIP read from 160.80.40.201:5060:
SIP/2.0 100 Trying
To: <sip:1004 at 160.80.40.201:5060>
From: "NPI-Rich" <sip:3000 at 160.80.40.4>;tag=as0e37bb22
Call-ID: 176eea4944e5fd1f63179a042ba51c06 at 160.80.40.4
CSeq: 102 INVITE
Via: SIP/2.0/UDP 160.80.40.4:5060;branch=z9hG4bK544dbabe
Server: Sipura/SPA942-4.1.10(e)
Content-Length: 0


--- (8 headers 0 lines)---
localhost*CLI>
<-- SIP read from 160.80.40.201:5060:
SIP/2.0 486 Busy Here
To: <sip:1004 at 160.80.40.201:5060>;tag=e434eff616a11501i0
From: "NPI-Rich" <sip:3000 at 160.80.40.4>;tag=as0e37bb22
Call-ID: 176eea4944e5fd1f63179a042ba51c06 at 160.80.40.4
CSeq: 102 INVITE
Via: SIP/2.0/UDP 160.80.40.4:5060;branch=z9hG4bK544dbabe
Server: Sipura/SPA942-4.1.10(e)
Content-Length: 0


--- (8 headers 0 lines)---
     -- Got SIP response 486 "Busy Here" back from 160.80.40.201
Transmitting (no NAT) to 160.80.40.201:5060:
ACK sip:1004 at 160.80.40.201:5060 SIP/2.0
Via: SIP/2.0/UDP 160.80.40.4:5060;branch=z9hG4bK544dbabe;rport
From: "NPI-Rich" <sip:3000 at 160.80.40.4>;tag=as0e37bb22
To: <sip:1004 at 160.80.40.201:5060>;tag=e434eff616a11501i0
Contact: <sip:3000 at 160.80.40.4>
Call-ID: 176eea4944e5fd1f63179a042ba51c06 at 160.80.40.4
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
     -- SIP/1004-081e9c08 is busy
   == Everyone is busy/congested at this time (1:1/0/0)
     -- Executing VoiceMail("IAX2/to-npi-3", "1004|ug(6)") in new stack
     -- Playing 'vm-theperson' (language 'en')
Destroying call '176eea4944e5fd1f63179a042ba51c06 at 160.80.40.4'
     -- Playing 'digits/1' (language 'en')
     -- Playing 'digits/0' (language 'en')
     -- Playing 'digits/0' (language 'en')
   == Spawn extension (from-sip, 1004, 2) exited non-zero on 'IAX2/to-npi-3'
     -- Executing Hangup("IAX2/to-npi-3", "") in new stack
   == Spawn extension (from-sip, h, 1) exited non-zero on 'IAX2/to-npi-3'
     -- Hungup 'IAX2/to-npi-3'

In addition, if I access the spa942 via a web browser, all lines/extns 
are idle. Does not seem to be any reason for the 'busy here' message 
that I can see.  Placing a call to another spa942 on the same Asterisk-B 
and on the same wire works fine.  Yesterday the first spa942 was working 
fine as well.

Can anyone see anything strange in the sip debug that would cause this?

R.

_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users