Hello: I'm newbie in asterisk, please help me.
My context is as follows: 192.168.4.2 --> Asterisk 11.13.1 complied from source 192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway When I call from a GSM cell phone, my TG100 GSM gateway answers and dials extension 7777 (configured as a hotline on TG100) to asterisk server, but asterisk server sends me "SIP/2.0 401 Unauthorized" response, I think it's a matter of contexts but I don't find the problem. Attached are sip.conf, extensions.conf and debug from 192.168.4.4 (TG100 GSM gateway). Thanks in advance.
<--- SIP read from UDP:192.168.4.4:5060 ---> INVITE sip:7777@192.168.4.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;rport Max-Forwards: 70 From: "999999999" <sip:999999999@192.168.4.4>;tag=as67354416 To: <sip:7777@192.168.4.2:5060> Contact: <sip:999999999@192.168.4.4> Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4 CSeq: 102 INVITE User-Agent: TG100 Date: Wed, 12 Nov 2014 10:13:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 281 v=0 o=root 1426707418 1426707418 IN IP4 192.168.4.4 s=Asterisk PBX 1.6.2.6 c=IN IP4 192.168.4.4 t=0 0 m=audio 10048 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (14 headers 13 lines) --- Sending to 192.168.4.4:5060 (no NAT) Sending to 192.168.4.4:5060 (no NAT) Using INVITE request as basis request - 12663beb04ae514f10c4b3a145368d5c@192.168.4.4 Found peer '555555555' for '999999999' from 192.168.4.4:5060 <--- Reliably Transmitting (no NAT) to 192.168.4.4:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;received=192.168.4.4;rport=5060 From: "999999999" <sip:999999999@192.168.4.4>;tag=as67354416 To: <sip:7777@192.168.4.2:5060>;tag=as16de6e5c Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4 CSeq: 102 INVITE Server: Asterisk PBX 11.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72011a6b" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '12663beb04ae514f10c4b3a145368d5c@192.168.4.4' in 6400 ms (Method: INVITE) <--- SIP read from UDP:192.168.4.4:5060 ---> ACK sip:7777@192.168.4.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;rport Max-Forwards: 70 From: "999999999" <sip:999999999@192.168.4.4>;tag=as67354416 To: <sip:7777@192.168.4.2:5060>;tag=as16de6e5c Contact: <sip:999999999@192.168.4.4> Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4 CSeq: 102 ACK User-Agent: TG100 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '6e9eab843b74d0860b108ed13a5d22c9@192.168.4.4' Method: OPTIONS Really destroying SIP dialog '12663beb04ae514f10c4b3a145368d5c@192.168.4.4' Method: ACK uc*CLI>
extensions.conf
Description: Binary data
sip.conf
Description: Binary data
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