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:[email protected]: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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