[asterisk-users] Help with sip debug?

[Rich Adamson]

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.