Summary: | ASTERISK-17616: [patch] fromuser not respected during OPTIONS message (qualify) | ||
Reporter: | Jeremy Kister (jkister) | Labels: | |
Date Opened: | 2011-03-29 13:43:03 | Date Closed: | 2011-10-24 15:00:15 |
Priority: | Minor | Regression? | |
Status: | Closed/Complete | Components: | Channels/chan_sip/General |
Versions: | 1.8.2 1.8.3 1.8.4 1.8.7.0 1.8.8.0 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | Attachments: | ( 0) chan_sip.c-options-fromuser-fix-v1.patch | |
Description: | {code} [peer](!) type=peer context=inbound qualify=yes qualifyfreq=300 insecure=port,invite [mypeer](peer) host=10.0.138.226 defaultuser=2155551941 fromuser=2155551941 md5secret=023f30a320a5781e8ffd1af9888012af {code} ****** ADDITIONAL INFORMATION ****** IP (tos 0x0, ttl 64, id 9242, offset 0, flags [none], proto UDP (17), length 555) 10.0.1.3.5060 > 10.0.138.226.5060: SIP, length: 527 OPTIONS sip:10.0.138.226 SIP/2.0 Via: SIP/2.0/UDP 10.0.83.61:5060;branch=z9hG4bK6abb74e3;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.0.83.61>;tag=as7444eb08 To: <sip:10.0.138.226> Contact: <sip:asterisk@10.0.83.61:5060> Call-ID: 20afd7e40fb31362355eae245dae1fd6@10.0.83.61:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.3 Date: Tue, 29 Mar 2011 17:43:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 IP (tos 0xb8, ttl 250, id 0, offset 0, flags [none], proto UDP (17), length 411) 10.0.138.226.5060 > 10.0.1.3.5060: SIP, length: 383 SIP/2.0 403 From: URI not recognized Via: SIP/2.0/UDP 10.245.83.61:5060;received=10.0.83.61;branch=z9hG4bK6abb74e3;rport=5060 From: "asterisk" <sip:asterisk@10.0.83.61>;tag=as7444eb08 To: <sip:10.0.138.226>;tag=metaswitch+1+0+e288612a Call-ID: 20afd7e40fb31362355eae245dae1fd6@10.0.83.61:5060 CSeq: 102 OPTIONS Server: DC-SIP/2.0 Organization: Content-Length: 0 | ||
Comments: | By: Jeremy Kister (jkister) 2011-03-29 13:43:31 fromuser works fine on REGISTER and INVITE. By: Leif Madsen (lmadsen) 2011-04-05 15:19:47 Please provide a full SIP trace from the Asterisk console. By: Jeremy Kister (jkister) 2011-04-05 20:48:11 i dont see how that's any different from what i've provided, but sure. pbx1*CLI> sip set debug peer mypeer SIP Debugging Enabled for IP: 10.0.138.226 pbx1*CLI> sip qualify peer mypeer Reliably Transmitting (NAT) to 10.0.138.226:5060: OPTIONS sip:10.0.138.226 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.3:5060;branch=z9hG4bK25e0a3b6;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.0.1.3>;tag=as55e1bb2c To: <sip:10.0.138.226> Contact: <sip:asterisk@10.0.1.3:5060> Call-ID: 016369855b5cf1fd11dad4921666ca14@10.0.1.3:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.3 Date: Wed, 06 Apr 2011 01:44:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:10.0.138.226:5060 ---> SIP/2.0 403 From: URI not recognized Via: SIP/2.0/UDP 10.0.1.3:5060;received=10.0.1.3;branch=z9hG4bK25e0a3b6;rport=5060 From: "asterisk" <sip:asterisk@10.0.1.3>;tag=as55e1bb2c To: <sip:10.0.138.226>;tag=metaswitch+1+0+e02fc5f2 Call-ID: 016369855b5cf1fd11dad4921666ca14@10.0.1.3:5060 CSeq: 102 OPTIONS Server: DC-SIP/2.0 Organization: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '016369855b5cf1fd11dad4921666ca14@10.0.1.3:5060' Method: OPTIONS pbx1*CLI> By: Jeremy Kister (jkister) 2011-09-02 23:48:36.912-0500 same patch verified to work on 1.8.6.0 |