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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 160.80.40.4:5060;branch=z9hG4bK544dbabe;rport From: "NPI-Rich" <sip:[EMAIL PROTECTED]>;tag=as0e37bb22 To: <sip:[EMAIL PROTECTED]:5060> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060> From: "NPI-Rich" <sip:[EMAIL PROTECTED]>;tag=as0e37bb22 Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060>;tag=e434eff616a11501i0 From: "NPI-Rich" <sip:[EMAIL PROTECTED]>;tag=as0e37bb22 Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 160.80.40.4:5060;branch=z9hG4bK544dbabe;rport From: "NPI-Rich" <sip:[EMAIL PROTECTED]>;tag=as0e37bb22 To: <sip:[EMAIL PROTECTED]:5060>;tag=e434eff616a11501i0 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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 '[EMAIL PROTECTED]' -- 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 _______________________________________________ --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
