[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] BUG? Asterisk V10 SIP Message To: non numeric IP (mobile1.xyz.com) fails
On Thu, Feb 13, 2014 at 3:28 AM, Johan Sandgren j...@svep.se wrote: Hi, snip [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@83.186.238.111 CSeq: 245 MESSAGE Max-Forwards: 70 User-Agent: CareIP 7813409 v1.2.4.0 Content-Type: application/scaip+xml Content-Length: 138 snip [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 It isn't a bug. It is telling you how it will attempt to match the inbound request: (1) It looks for a peer that matches what sent the MESSAGE request, in this case, sip:83.186.238.111. That fails. (2) Since the request URI is simply a domain and not a destination, it falls back to looking for an 's' extension in context 'sipmessage'. That fails. (3) Now, truly panicking, it looks for the 'i' extension in the same context. Since you don't have an invalid extension handler, that fails too. Despondent, it throws in the towel. I'm not sure where you thought it would end up, but it certainly tried lots of different places. And the 'rules' for it doing so are (for chan_sip, at any rate) relatively consistent with how inbound requests are matched for INVITE requests as well. chan_sip tries to figure out who sent the request to Asterisk, and then use that peer definition. If chan_sip can't find that, it falls back to using a general entry point. Also: please don't stay on Asterisk 10. That version is no longer supported and is no longer receiving security fixes. You should move to Asterisk 11, which is an LTS release, as soon as possible. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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
