[asterisk-dev] BUG? Asterisk V10 SIP Message To: non numeric IP (mobile1.xyz.com) fails
Hi, I’m using SIP MESSAGE to asterisk V10 and it fails to be received. I’m not super sure of the reason but I’m making this guess: Due to I’m using non ipaddress in the to field, which contains sip:mobil1.xyz.com, Asterisk makes the mistake to try matching this name ”mobil1.testserver.com” in extensions.conf and no extension/peer is found in the sip-message context I’ve configured. It works when the TO: field contains an numeric ipadress. Is this a bug or an intentional limitation? /Johan LOG [Feb 12 15:13:59] VERBOSE[25824] chan_sip.c: --- SIP read from UDP:83.186.238.111:5060 --- MESSAGE sip:mobil1.xyz.com SIP/2.0 Via: SIP/2.0/UDP 83.186.238.111:5060;branch=z9hG4bK-3f138a53 To: sip:mobil1.xyz.com From: sip:83.186.238.111;tag=7a82b127 Call-ID: 857d4ed8@83.186.238.111mailto:857d4ed8@83.186.238.111 CSeq: 245 MESSAGE Max-Forwards: 70 User-Agent: CareIP 7813409 v1.2.4.0 Content-Type: application/scaip+xml Content-Length: 138 My message - [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 0 [ 49]: MESSAGE sip:mobil1.xyz.com SIP/2.0 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 83.186.238.111:5060;branch=z9hG4bK-3f138a53 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 2 [ 39]: To: sip:mobil1.xyz.com [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 3 [ 39]: From: sip:83.186.238.111;tag=7a82b127 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 4 [ 32]: Call-ID: 857d4ed8@83.186.238.111mailto:857d4ed8@83.186.238.111 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 5 [ 17]: CSeq: 245 MESSAGE [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 7 [ 35]: User-Agent: CareIP 7813409 v1.2.4.0 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 8 [ 35]: Content-Type: application/scaip+xml [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 9 [ 19]: Content-Length: 138 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Header 10 [ 0]: [Feb 12 15:13:59] DEBUG[25824] chan_sip.c:Body 0 [138]: My message [Feb 12 15:13:59] VERBOSE[25824] chan_sip.c: --- (10 headers 1 lines) --- [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: = Looking for Call ID: 857d4ed8@83.186.238.111mailto:857d4ed8@83.186.238.111 (Checking From) --From tag 7a82b127 --To-tag [Feb 12 15:13:59] DEBUG[25824] acl.c: For destination '83.186.238.111', our source address is '172.26.19.13'. [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Target address 83.186.238.111:5060 is not local, substituting externaddr [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.105.99.108:5060 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Allocating new SIP dialog for 857d4ed8@83.186.238.111mailto:857d4ed8@83.186.238.111 - MESSAGE (No RTP) [Feb 12 15:13:59] VERBOSE[25824] chan_sip.c: No matching peer for '83.186.238.111' from '83.186.238.111:5060' [Feb 12 15:13:59] VERBOSE[25824] chan_sip.c: Looking for s in sipmessage (domain mobil1.xyz.com) [Feb 12 15:13:59] WARNING[25812] pbx.c: Channel 'Message/ast_msg_queue' sent into invalid extension 'mobil1.xyz.com' in context 'sipmessage', but no invalid handler Johan Sandgren Software Engineer Svep Design Center AB S:t Lars väg 42A 222 70 Lund, Sweden Phone +46 46 192 722 E-mail j...@svep.semailto:j...@svep.se Website www.svep.sehttp://www.svep.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Make asterisk V10 require SIP authentication for unregistered Message / Invite (calls) ?
Did I catch you all at a bad time with the code reviews and all? I'm still hoping for some responses or ideas, or from someone with knowledge. Are there anyone out there? :) /Johan Från: asterisk-dev-boun...@lists.digium.com [mailto:asterisk-dev-boun...@lists.digium.com] För Johan Sandgren Skickat: den 22 november 2013 18:09 Till: asterisk-dev@lists.digium.com Ämne: [asterisk-dev] Make asterisk V10 require SIP authentication for unregistered Message / Invite (calls) ? Hi everyone, Is it possible to make asterisk REQUIRE authentication for unregistered incoming SIP MESSAGE och SIP INVITE (all related to incoming calls)? With registered sip clients, asterisk successfully asks for authorization for each message. I also need to support unregistered clients (it would be a global user + password in this case). Any ideas of which global settings does this? I haven't found anything yet. Or suggestions of where I possibly could edit the source code to enable this feature. I have a bit of knowledge of the sourcecode, and have compiled it before. Johan Sandgren Software Engineer Svep Design Center AB S:t Lars väg 42A 222 70 Lund, Sweden Phone +46 46 192 722 E-mail j...@svep.semailto:j...@svep.se Website www.svep.sehttp://www.svep.se/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Make asterisk V10 require SIP authentication for unregistered Message / Invite (calls) ?
Oh thanks Olle, I'm on the wrong list. Sorry about that. Från: asterisk-dev-boun...@lists.digium.com [mailto:asterisk-dev-boun...@lists.digium.com] För Olle E. Johansson Skickat: den 25 november 2013 11:32 Till: Asterisk Developers Mailing List Kopia: Olle E Johansson Ämne: Re: [asterisk-dev] Make asterisk V10 require SIP authentication for unregistered Message / Invite (calls) ? Hi! The development mailing list is not a good list for asking general questions - please use the user's mailing list for that. There's currently no way of authenticating devices that are not peers or users of asterisk. Registration is unrelated to other requests. You could do this with Kamailio of course. Hälsningar /Olle On 25 Nov 2013, at 11:17, Johan Sandgren j...@svep.semailto:j...@svep.se wrote: Did I catch you all at a bad time with the code reviews and all? I'm still hoping for some responses or ideas, or from someone with knowledge. Are there anyone out there? :) /Johan Från: asterisk-dev-boun...@lists.digium.commailto:asterisk-dev-boun...@lists.digium.com [mailto:asterisk-dev-boun...@lists.digium.com] För Johan Sandgren Skickat: den 22 november 2013 18:09 Till: asterisk-dev@lists.digium.commailto:asterisk-dev@lists.digium.com Ämne: [asterisk-dev] Make asterisk V10 require SIP authentication for unregistered Message / Invite (calls) ? Hi everyone, Is it possible to make asterisk REQUIRE authentication for unregistered incoming SIP MESSAGE och SIP INVITE (all related to incoming calls)? With registered sip clients, asterisk successfully asks for authorization for each message. I also need to support unregistered clients (it would be a global user + password in this case). Any ideas of which global settings does this? I haven't found anything yet. Or suggestions of where I possibly could edit the source code to enable this feature. I have a bit of knowledge of the sourcecode, and have compiled it before. Johan Sandgren Software Engineer Svep Design Center AB S:t Lars väg 42A 222 70 Lund, Sweden Phone +46 46 192 722 E-mail j...@svep.semailto:j...@svep.se Website www.svep.sehttp://www.svep.se/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://www.api-digital.com/ -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Make asterisk V10 require SIP authentication for unregistered Message / Invite (calls) ?
Hi everyone, Is it possible to make asterisk REQUIRE authentication for unregistered incoming SIP MESSAGE och SIP INVITE (all related to incoming calls)? With registered sip clients, asterisk successfully asks for authorization for each message. I also need to support unregistered clients (it would be a global user + password in this case). Any ideas of which global settings does this? I haven't found anything yet. Or suggestions of where I possibly could edit the source code to enable this feature. I have a bit of knowledge of the sourcecode, and have compiled it before. Johan Sandgren Software Engineer Svep Design Center AB S:t Lars väg 42A 222 70 Lund, Sweden Phone +46 46 192 722 E-mail j...@svep.semailto:j...@svep.se Website www.svep.sehttp://www.svep.se/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
