Paul Duffy
2006-Jul-20 07:50 UTC
[asterisk-users] Fast busy after one digit dialled? - 7970 SIP 8.0.3
Hi All I'm trying to get a 7970 working with SIP 8.0.3S and the latest build of Asterisk (doing this as a new build before a replacement of my existing system). So far I've managed to get the phone upgraded successfully. I can dial the phone from the console successfully. However whenever I dial a number from the phone the phone goes into fast busy mode after the first digit. The SIP debug shows (in this instance the digit dialled was a 7):- <-- SIP read from 10.131.111.51:49226: INVITE sip:7@10.131.111.10;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.131.111.51:5060;branch=z9hG4bK8f6323d8 From: "301" <sip:301@10.131.111.10>;tag=000f3487e566000ed76e9557-dcf4e93a To: <sip:7@10.131.111.10;user=phone> Call-ID: 000f3487-e5660009-e2e36f89-af6ced7c@10.131.111.51 Max-Forwards: 70 Date: Thu, 20 Jul 2006 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7970G/8.0 Contact: <sip:301@10.131.111.51:5060;transport=udp> Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "301" <sip:301@10.131.111.10>;party=calling;id-type=subscriber;privacy=off;screenyes Supported: replaces,join,norefersub Allow-Events: kpml,dialog Content-Length: 275 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 21184 0 IN IP4 10.131.111.51 s=SIP Call t=0 0 m=audio 19796 RTP/AVP 0 8 18 101 c=IN IP4 10.131.111.51 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/0 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (19 headers 13 lines)--- Using INVITE request as basis request - 000f3487-e5660009-e2e36f89-af6ced7c@10.131.111.51 Sending to 10.131.111.51 : 5060 (non-NAT) Found user '301' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.131.111.51:19796 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 7 in 7970SIP (domain 10.131.111.10) Reliably Transmitting (no NAT) to 10.131.111.51:5060: SIP/2.0 484 Address Incomplete ******************************************** Via: SIP/2.0/UDP 10.131.111.51:5060;branch=z9hG4bK8f6323d8;received=10.131.111.51 From: "301" <sip:301@10.131.111.10>;tag=000f3487e566000ed76e9557-dcf4e93a To: <sip:7@10.131.111.10;user=phone>;tag=as17095de3 Call-ID: 000f3487-e5660009-e2e36f89-af6ced7c@10.131.111.51 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:7@10.131.111.10> Content-Length: 0 --- <-- SIP read from 10.131.111.51:49227: ACK sip:7@10.131.111.10;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.131.111.51:5060;branch=z9hG4bK8f6323d8 From: "301" <sip:301@10.131.111.10>;tag=000f3487e566000ed76e9557-dcf4e93a To: <sip:7@10.131.111.10;user=phone>;tag=as17095de3 Call-ID: 000f3487-e5660009-e2e36f89-af6ced7c@10.131.111.51 Date: Thu, 20 Jul 2006 GMT CSeq: 101 ACK Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '000f3487-e5660009-e2e36f89-af6ced7c@10.131.111.51' The "Address Incomplete" that I've highlighted use ************ above is interesting but having read the wiki and googled extensively I can't see a reason for this problem. As I'm now starting to get brain block from this can anyone make any suggestions (it's probably something incredibly simple I'm missing, I just can't see it). Thanks
Tomislav Parčina
2006-Jul-30 23:16 UTC
[asterisk-users] Re: Fast busy after one digit dialled? - 7970 SIP 8.0.3
In article <BAY116-DAV74D4500908EDF12A3E8F4E2610@phx.gbl>, paul_lonny@hotmail.com says...> However whenever I dial a number from the phone the phone goes into fast > busy mode after the first digit.Check phone's dialplan.xml file on your tftp server. -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: tomo@pbx.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr