Rich,<br>I had the same problem and the solution was to take out a 'malformed' callerid value from my sip.conf entry.<br><br>Tom<br><br><div><span class="gmail_quote">On 7/20/06, <b class="gmail_sendername">Rich Adamson</b>
<<a href="mailto:radamson@routers.com">radamson@routers.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Tried the syslog & debug, but it reports the exact same thing as the sip
<br>debug shown below. It includes the INVITE, 100 TRYING, AND 486 BUSY<br>HERE. There are no hints as to why the Busy Here message is returned.<br><br>I was kind of guessing that something in the sip header was not as<br>
expected for the device, but I don't see anything that seems to be<br>inappropriate in the sip debug.<br><br>Thoughts?<br><br><br>Shanon Swafford wrote:<br>> I always like to activate the syslog and debug on my SPA's. Sometimes this
<br>> will tell you what they are doing.<br>><br>> Shanon<br>><br>><br>><br>> -----Original Message-----<br>><br>> Need a little help trying to understand what's happening here.<br>><br>> spa941 -> Asterisk-A -> iax2 -> Asterisk-B -> spa942
<br>><br>> When the spa941 (x3000) calls spa942 (x1004), the spa942 returns a "busy<br>> here" sip message. The spa942 is not busy and does not have DND or any<br>> other option set to cause a busy-here message. Asterisk-B is
v1.2.10<br>> updated to current svn. (Seeing the exact same issue with an spa3k.)<br>><br>> A sip debug from Asterisk-B shows the following three packets:<br>><br>> localhost*CLI> sip debug peer 1004<br>
> SIP Debugging Enabled for IP: <a href="http://160.80.40.201:5060">160.80.40.201:5060</a> <== x1004<br>> -- Registered IAX2 to '<a href="http://151.213.193.101">151.213.193.101</a>', who sees us as<br>>
153.22<a href="snap://2.216.140:1963" id="dyn">2.216.140:1963</a> with no messages waiting<br>><br>> -- Accepting UNAUTHENTICATED call from <a href="http://151.213.193.101">151.213.193.101</a>:<br>> > requested format = gsm,
<br>> > requested prefs = (g726|gsm|ilbc),<br>> > actual format = g726,<br>> > host prefs = (g726|gsm|ilbc),<br>> > priority = mine<br>> -- Executing Dial("IAX2/to-npi-3", "SIP/1004|15|r") in new stack
<br>> We're at <a href="http://160.80.40.4">160.80.40.4</a> port 13382<br>> Adding codec 0x2 (gsm) to SDP<br>> Adding codec 0x4 (ulaw) to SDP<br>> Adding codec 0x8 (alaw) to SDP<br>> Adding non-codec 0x1 (telephone-event) to SDP
<br>> 13 headers, 12 lines<br>> Reliably Transmitting (no NAT) to <a href="http://160.80.40.201:5060">160.80.40.201:5060</a>:<br>> INVITE sip:1004@160.80.40.201:5060 SIP/2.0<br>> Via: SIP/2.0/UDP <a href="http://160.80.40.4:5060">
160.80.40.4:5060</a>;branch=z9hG4bK544dbabe;rport<br>> From: "NPI-Rich" <<a href="mailto:sip:3000@160.80.40.4">sip:3000@160.80.40.4</a>>;tag=as0e37bb22<br>> To: <sip:1004@160.80.40.201:5060><br>
> Contact: <<a href="mailto:sip:3000@160.80.40.4">sip:3000@160.80.40.4</a>><br>> Call-ID: <a href="mailto:176eea4944e5fd1f63179a042ba51c06@160.80.40.4">176eea4944e5fd1f63179a042ba51c06@160.80.40.4</a><br>> CSeq: 102 INVITE
<br>> User-Agent: Asterisk PBX<br>> Max-Forwards: 70<br>> Date: Wed, 19 Jul 2006 22:27:31 GMT<br>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>> Content-Type: application/sdp<br>> Content-Length: 261
<br>><br>> v=0<br>> o=root 18182 18182 IN IP4 <a href="http://160.80.40.4">160.80.40.4</a><br>> s=session<br>> c=IN IP4 <a href="http://160.80.40.4">160.80.40.4</a><br>> t=0 0<br>> m=audio 13382 RTP/AVP 3 0 8 101
<br>> a=rtpmap:3 GSM/8000<br>> a=rtpmap:0 PCMU/8000<br>> a=rtpmap:8 PCMA/8000<br>> a=rtpmap:101 telephone-event/8000<br>> a=fmtp:101 0-16<br>> a=silenceSupp:off - - - -<br>><br>> ---<br>> -- Called 1004
<br>> localhost*CLI><br>> <-- SIP read from <a href="http://160.80.40.201:5060">160.80.40.201:5060</a>:<br>> SIP/2.0 100 Trying<br>> To: <sip:1004@160.80.40.201:5060><br>> From: "NPI-Rich" <
<a href="mailto:sip:3000@160.80.40.4">sip:3000@160.80.40.4</a>>;tag=as0e37bb22<br>> Call-ID: <a href="mailto:176eea4944e5fd1f63179a042ba51c06@160.80.40.4">176eea4944e5fd1f63179a042ba51c06@160.80.40.4</a><br>> CSeq: 102 INVITE
<br>> Via: SIP/2.0/UDP <a href="http://160.80.40.4:5060">160.80.40.4:5060</a>;branch=z9hG4bK544dbabe<br>> Server: Sipura/SPA942-4.1.10(e)<br>> Content-Length: 0<br>><br>><br>> --- (8 headers 0 lines)---<br>
> localhost*CLI><br>> <-- SIP read from <a href="http://160.80.40.201:5060">160.80.40.201:5060</a>:<br>> SIP/2.0 486 Busy Here<br>> To: <sip:1004@160.80.40.201:5060>;tag=e434eff616a11501i0<br>> From: "NPI-Rich" <
<a href="mailto:sip:3000@160.80.40.4">sip:3000@160.80.40.4</a>>;tag=as0e37bb22<br>> Call-ID: <a href="mailto:176eea4944e5fd1f63179a042ba51c06@160.80.40.4">176eea4944e5fd1f63179a042ba51c06@160.80.40.4</a><br>> CSeq: 102 INVITE
<br>> Via: SIP/2.0/UDP <a href="http://160.80.40.4:5060">160.80.40.4:5060</a>;branch=z9hG4bK544dbabe<br>> Server: Sipura/SPA942-4.1.10(e)<br>> Content-Length: 0<br>><br>><br>> --- (8 headers 0 lines)---<br>
> -- Got SIP response 486 "Busy Here" back from <a href="http://160.80.40.201">160.80.40.201</a><br>> Transmitting (no NAT) to <a href="http://160.80.40.201:5060">160.80.40.201:5060</a>:<br>> ACK sip:1004@160.80.40.201
:5060 SIP/2.0<br>> Via: SIP/2.0/UDP <a href="http://160.80.40.4:5060">160.80.40.4:5060</a>;branch=z9hG4bK544dbabe;rport<br>> From: "NPI-Rich" <<a href="mailto:sip:3000@160.80.40.4">sip:3000@160.80.40.4</a>
>;tag=as0e37bb22<br>> To: <sip:1004@160.80.40.201:5060>;tag=e434eff616a11501i0<br>> Contact: <<a href="mailto:sip:3000@160.80.40.4">sip:3000@160.80.40.4</a>><br>> Call-ID: <a href="mailto:176eea4944e5fd1f63179a042ba51c06@160.80.40.4">
176eea4944e5fd1f63179a042ba51c06@160.80.40.4</a><br>> CSeq: 102 ACK<br>> User-Agent: Asterisk PBX<br>> Max-Forwards: 70<br>> Content-Length: 0<br>><br>><br>> ---<br>> -- SIP/1004-081e9c08 is busy
<br>> == Everyone is busy/congested at this time (1:1/0/0)<br>> -- Executing VoiceMail("IAX2/to-npi-3", "1004|ug(6)") in new stack<br>> -- Playing 'vm-theperson' (language 'en')<br>
> Destroying call '<a href="mailto:176eea4944e5fd1f63179a042ba51c06@160.80.40.4">176eea4944e5fd1f63179a042ba51c06@160.80.40.4</a>'<br>> -- Playing 'digits/1' (language 'en')<br>> -- Playing 'digits/0' (language 'en')
<br>> -- Playing 'digits/0' (language 'en')<br>> == Spawn extension (from-sip, 1004, 2) exited non-zero on 'IAX2/to-npi-3'<br>> -- Executing Hangup("IAX2/to-npi-3", "") in new stack
<br>> == Spawn extension (from-sip, h, 1) exited non-zero on 'IAX2/to-npi-3'<br>> -- Hungup 'IAX2/to-npi-3'<br>><br>> In addition, if I access the spa942 via a web browser, all lines/extns<br>> are idle. Does not seem to be any reason for the 'busy here' message
<br>> that I can see. Placing a call to another spa942 on the same Asterisk-B<br>> and on the same wire works fine. Yesterday the first spa942 was working<br>> fine as well.<br>><br>> Can anyone see anything strange in the sip debug that would cause this?
<br>><br>> R.<br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:
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