Hello!
I'm trying to setup a basic script to initiate a basic call to voicemail and
I'm having some trouble doing that. I'm getting the error message:
"Authentication keyword without dialog_authentication!" after the first ack.
Adding 'auth="true" to the second invite's send doesn't seem to have any
effect. What's going on?
Thank you in advance,
Robert
#sipp -sf test1.xml [ip of sipx] -m 1 -au [usr] -ap [passwd] -s [voicemail
phone #]
Messages Retrans Timeout Unexpected-Msg
INVITE ----------> 1 0
ACK ----------> 1 0
INVITE ----------> 0 0
ACK ----------> 0 0
BYE ----------> 0 0
--==-- My Scenario --==--
<scenario name="Default scenario">
<send auth="true">
<![CDATA[
INVITE sip:[service]@[remote_ip];user=phone;transport=tcp SIP/2.0
Via: SIP/2.0/[transport] [local_ip];branch=[branch]
From: "Robert Remsik" <sip:[email protected]>;tag=[call_number]
To: <sip:[service]@[remote_ip];user=phone>
CSeq: [cseq] INVITE
Call-ID: [call_id]
Contact: <sip:17120@[local_ip];transport=[transport]>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomVVX-VVX_410-UA/4.1.8.0628
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 1470166919 1470166919 IN IP[local_ip_type] [local_ip]
s=Polycom IP Phone
c=IN IP[media_ip_type] [media_ip]
t=0 0
a=sendrecv
m=audio [media_port] RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
]]>
</send>
<send>
<![CDATA[
ACK sip:[service]@[remote_ip];user=phone;transport=tcp SIP/2.0
Via: SIP/2.0/[transport] [local_ip];branch=[branch]
From: "Robert Remsik" <sip:[email protected]>;tag=[call_number]
To: <sip:[service]@[remote_ip];user=phone>;[peer_tag_param]
CSeq: [cseq] ACK
Call-ID: [call_id]
Contact: <sip:17120@[local_ip];transport=[transport]>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomVVX-VVX_410-UA/4.1.8.0628
Accept-Language: en
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<send>
<![CDATA[
INVITE sip:[service]@[remote_ip];user=phone;transport=tcp SIP/2.0
Via: SIP/2.0/[transport] [local_ip];branch=[branch]
From: "Robert Remsik" <sip:[email protected]>;tag=[call_number]
To: <sip:[service]@[remote_ip];user=phone>
CSeq: [cseq] INVITE
Call-ID: [call_id]
Contact: <sip:17120@[local_ip];transport=[transport]>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomVVX-VVX_410-UA/4.1.8.0628
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
[authentication username="abc" password="abc"]
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 1470166919 1470166919 IN IP[local_ip_type] [local_ip]
s=Polycom IP Phone
c=IN IP[media_ip_type] [media_ip]
t=0 0
a=sendrecv
m=audio [media_port] RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
]]>
</send>
<send>
<![CDATA[
ACK sip:e4PGFkEVw6krpgtHyWyvFNr4mi60DsxLQcG4Err5hfzg.@[remote_ip];transport=tcp
SIP/2.0
Via: SIP/2.0/[transport] [local_ip];branch=[branch]
From: "Robert Remsik" <sip:[email protected]>;tag=[call_number]
To: <sip:[service]@[remote_ip];user=phone>;[peer_tag_param]
Route:
<sip:129.82.254.250:5060;lr;sipXecs-CallDest=VM;sipXecs-rs=%2Aauth%7E.%2Afrom%7EODEyRjdFOTMtNzY4ODg1Rjk%60.900_ntap%2Aid%7ENDE4OS04Mjg0%21ea26a124f1f3352fdb0bcb790663f8a6;x-sipX-done>
CSeq: [cseq] ACK
Call-ID: [call_id]
Contact: <sip:17120@[local_ip];transport=[transport]>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomVVX-VVX_410-UA/4.1.8.0628
Accept-Language: en
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<send>
<![CDATA[
BYE sip:e4PGFkEVw6krpgtHyWyvFNr4mi60DsxLQcG4Err5hfzg.@[remote_ip];transport=tcp
SIP/2.0
Via: SIP/2.0/[transport] [local_ip];branch=[branch]
From: "Robert Remsik" <sip:[email protected]>;tag=[call_number]
To: <sip:[service]@[remote_ip];user=phone>;[peer_tag_param]
Route:
<sip:129.82.254.250:5060;lr;sipXecs-CallDest=VM;sipXecs-rs=%2Aauth%7E.%2Afrom%7EODEyRjdFOTMtNzY4ODg1Rjk%60.900_ntap%2Aid%7ENDE4OS04Mjg0%21ea26a124f1f3352fdb0bcb790663f8a6;x-sipX-done>
CSeq: [cseq] BYE
Call-ID: [call_id]
Contact: <sip:17120@[local_ip]>
User-Agent: PolycomVVX-VVX_410-UA/4.1.8.0628
Accept-Language: en
Max-Forwards: 70
Content-Length: 0
]]>
Robert Remsik
ACNS
Desk Phone: 970 491 7120
[email protected]
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