ben.carrasco/jambonz-feature-server
1
0
Fork 0
mirror of https://github.com/jambonz/jambonz-feature-server.git synced 2025-12-20 16:50:39 +00:00
Code Issues Packages Projects Releases Wiki Activity

feat: play verb url support single or array list url (#158)

Browse Source 
* feat: update time-series 0.11.12

* feat: support play verb url in plain text or array

* fix: review comment

Co-authored-by: Dave Horton <daveh@beachdognet.com>
This commit is contained in:
xquanluu
2022-08-25 15:09:48 +07:00
committed by GitHub
parent 4f0439dad9
commit 127432f2ec
14 changed files with 323 additions and 20 deletions
Download Patch File Download Diff File Expand all files Collapse all files

92
test/scenarios/uac-success-send-bye.xml Normal file
View File

@@ -0,0 +1,92 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[to]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [from] <sip:[from]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[to]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:[from]@[local_ip]:[local_port]
Max-Forwards: 70
X-Account-Sid: bb845d4b-83a9-4cde-a6e9-50f3743bab3f
Subject: uac-say
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[to]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [from] <sip:[from]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [to] <sip:[to]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:[from]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-say
Content-Length: 0
]]>
</send>
<pause milliseconds="3000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:sip:[to]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [from] <sip:[from]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [to] <sip:[to]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
</scenario>