[asterisk-users] wrong from URI in options message
I recently configured a SIP peer which i must specify my fromuser as my ten digit DID. I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is asterisk. Is this a bug? Or is there some other config I must make ? register = 211941:123456@10.0.138.226/211941~600 [peer](!) type=peer context=inbound qualify=yes qualifyfreq=300 insecure=port,invite nat=yes outgoinglimit=4 incominglimit=4 [mypeer](peer) host=10.0.138.226 defaultuser=211941 fromuser=211941 md5secret=023f30a320a5781e8ffd1af9888012af incominglimit=10 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 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrong from URI in options message
On 3/29/2011 12:52 PM, Jeremy Kister wrote: I recently configured a SIP peer which i must specify my fromuser as my ten digit DID. I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is asterisk. Is this a bug? Or is there some other config I must make ? register = 211941:123456@10.0.138.226/211941~600 [peer](!) type=peer context=inbound qualify=yes qualifyfreq=300 insecure=port,invite nat=yes outgoinglimit=4 incominglimit=4 [mypeer](peer) host=10.0.138.226 defaultuser=211941 fromuser=211941 md5secret=023f30a320a5781e8ffd1af9888012af incominglimit=10 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 IIRC, you need to define the fromuser in the peer in order for the qualify checks (options packets) to contain the user you want -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrong from URI in options message
On 3/29/2011 1:56 PM, Sherwood McGowan wrote: [mypeer](peer) host=10.0.138.226 defaultuser=211941 fromuser=211941 md5secret=023f30a320a5781e8ffd1af9888012af incominglimit=10 IIRC, you need to define the fromuser in the peer in order for the qualify checks (options packets) to contain the user you want uhm, didn't I ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrong from URI in options message
On Tue, Mar 29, 2011 at 1:25 PM, Jeremy Kister asterisk...@jeremykister.com wrote: On 3/29/2011 1:56 PM, Sherwood McGowan wrote: [mypeer](peer) host=10.0.138.226 defaultuser=211941 fromuser=211941 md5secret=023f30a320a5781e8ffd1af9888012af incominglimit=10 IIRC, you need to define the fromuser in the peer in order for the qualify checks (options packets) to contain the user you want uhm, didn't I ? It looks like you did to me. Is it just OPTIONS packets that are showing the wrong fromuser field? In other words, when you send call traffic over this peer, does it properly create the SIP packets? For some reason, I'm thinking this is just the way it is, but someone closer to the the actual sip development may be able to better tell you. Perhaps open a ticket on the bug tracker? -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrong from URI in options message
On 3/29/2011 2:29 PM, Warren Selby wrote: It looks like you did to me. Is it just OPTIONS packets that are showing the wrong fromuser field? In other words, when you send call traffic over this peer, does it properly create the SIP packets? For some reason, I'm correct - when i actually invite a call or do the register, the from uri is correct. it's just the options packet that is broken. sip development may be able to better tell you. Perhaps open a ticket on the bug tracker? yep, that was the next step - just wanted to run it by a few more eyes before i bothered the devs. https://issues.asterisk.org/view.php?id=19036 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrong from URI in options message
Oh, damn, my bad, I've apparently read too many sip.conf entries today -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
