Hello, I'm trying to send multiple calls to my Asterisk Instance(Asterisk 19.2.1). I have created the UAC XML from the requests logged by Asterisk when communicating with a softphone. However when I trigger this, I see Asterisk responding to all of these requests with 401 status.
What should I change here to successfully trigger calls to Asterisk? I have not masked any IP or numbers here since these computers are only accessible from my local network. SIPp command, ``` sipp -sf Basic/uac.xml asterisk-dev ``` uac.xml ```xml <?xml version="1.0" encoding="UTF-8" ?> <scenario name="Basic UAC scenario"> <send> <![CDATA[ INVITE sip:100@asterisk-dev SIP/2.0 Via: SIP/2.0/UDP 100.103.250.60:50906 ;rport;branch=z9hG4bKPjb0kGNfULZnPtVspABKIkmekflf3zPKk1 Max-Forwards: 70 From: "shai" <sip:6001@asterisk-dev >;tag=6ukVZBwnfdB7Ae95nE.EtAF6MaA9I43m To: sip:100@asterisk-dev Contact: <sip:6001@100.103.250.60:50906;ob> Call-ID: i1XKt0o6WzopIEc4oTwEnz3zxjemmI8s CSeq: 6001 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Telephone 1.5.2 Authorization: Digest username="6001", realm="asterisk", nonce="1650456215/33cf144cc5c0370a2d355d070f987efd", uri="sip:100@asterisk-dev", response="bf3337f5c8edd5c6506e42225ed4bc35", algorithm=md5, cnonce="LvE3CsEAXJMJlFszvDDEVaLHwfKhseD4", opaque="4a7e2ea9085ffb42", qop=auth, nc=00000001 Content-Type: application/sdp Content-Length: 483 v=0 o=- 3859445015 3859445015 IN IP4 100.103.250.60 s=pjmedia b=AS:117 t=0 0 a=X-nat:0 m=audio 4006 RTP/AVP 96 9 8 0 101 102 c=IN IP4 100.103.250.60 b=TIAS:96000 a=rtcp:4007 IN IP4 100.103.250.60 a=sendrecv a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/48000 a=fmtp:101 0-16 a=rtpmap:102 telephone-event/8000 a=fmtp:102 0-16 a=ssrc:1512283992 cname:785442ca63ed197d ]]> </send> <recv response="100" optional="true"> </recv> <recv response="180" optional="true"> </recv> <recv response="200"> </recv> <send> <![CDATA[ ACK sip:100.69.169.87:5060 SIP/2.0 Via: SIP/2.0/UDP 100.103.250.60:50906 ;rport;branch=z9hG4bKPjB20rSWAebAkr8Vr9lHsKTkGr84MrCahP Max-Forwards: 70 From: "shai" <sip:6001@asterisk-dev >;tag=6EVN12veK30DOeBFZ9mB7a9o9HCaI1X6 To: sip:100@asterisk-dev;tag=33b584f6-5849-4020-bf16-6ea0296e38ae Call-ID: 8LT92XH8Qf8ZZSebCulkuOwdngQTpbF- CSeq: 14242 ACK Content-Length: 0 ]]> </send> <pause milliseconds="5000"/> <send retrans="500"> <![CDATA[ BYE sip:100.69.169.87:5060 SIP/2.0 Via: SIP/2.0/UDP 100.103.250.60:50906 ;rport;branch=z9hG4bKPjtpJP7zMUrJixyy-QHRk79pmcKQlFTVzL Max-Forwards: 70 From: "shai" <sip:6001@asterisk-dev >;tag=6EVN12veK30DOeBFZ9mB7a9o9HCaI1X6 To: sip:100@asterisk-dev;tag=33b584f6-5849-4020-bf16-6ea0296e38ae Call-ID: 8LT92XH8Qf8ZZSebCulkuOwdngQTpbF- CSeq: 14243 BYE User-Agent: Telephone 1.5.2 Content-Length: 0 ]]> </send> <recv response="200"> </recv> </scenario> ``` Asterisk response, ``` <--- Transmitting SIP response (568 bytes) to UDP:100.103.250.60:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 100.103.250.60:50906 ;rport=5060;received=100.103.250.60;branch=z9hG4bKPjb0kGNfULZnPtVspABKIkmekflf3zPKk1 Call-ID: i1XKt0o6WzopIEc4oTwEnz3zxjemmI8s From: "shai" <sip:6001@asterisk-dev>;tag=6ukVZBwnfdB7Ae95nE.EtAF6MaA9I43m To: <sip:100@asterisk-dev>;tag=z9hG4bKPjb0kGNfULZnPtVspABKIkmekflf3zPKk1 CSeq: 6001 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1650456365/6f4af806234d2a4690cdc945b0903a31",opaque="2683948a0d25bcd3",stale=true,algorithm=md5,qop="auth" Server: Asterisk PBX 19.2.1 Content-Length: 0 ``` Thanks
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