[asterisk-users] Help with sip debug?

Thu Jul 20 06:52:42 MST 2006

Rich,
I had the same problem and the solution was to take out a 'malformed'
callerid value from my sip.conf entry.

Tom

On 7/20/06, Rich Adamson <radamson at routers.com> wrote:
>
> Tried the syslog & debug, but it reports the exact same thing as the sip
> debug shown below. It includes the INVITE, 100 TRYING, AND 486 BUSY
> HERE. There are no hints as to why the Busy Here message is returned.
>
> I was kind of guessing that something in the sip header was not as
> expected for the device, but I don't see anything that seems to be
> inappropriate in the sip debug.
>
> Thoughts?
>
>
> Shanon Swafford wrote:
> > 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-----
> >
> > 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.
>
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