mirror of
https://github.com/jambonz/sbc-inbound.git
synced 2025-12-19 04:37:43 +00:00
#32 - allow wildcard matches
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@@ -56,5 +56,12 @@ values ('888a5339-c62c-4075-9e19-f4de70a96597', '999c1452-620d-4195-9f19-c9814ef
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insert into phone_numbers (phone_number_sid, number, voip_carrier_sid, account_sid)
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values ('999a5339-c62c-4075-9e19-f4de70a96597', '16173333456', '287c1452-620d-4195-9f19-c9814ef90d78', 'ed649e33-e771-403a-8c99-1780eabbc803');
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insert into phone_numbers (phone_number_sid, number, voip_carrier_sid, account_sid)
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values ('29543d4e-d959-4a25-836a-cde7161cd7d5', '1508222*', '287c1452-620d-4195-9f19-c9814ef90d78', 'ed649e33-e771-403a-8c99-1780eabbc803');
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insert into phone_numbers (phone_number_sid, number, voip_carrier_sid, account_sid)
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values ('dddd5c34-feae-4d70-98af-bb4d1f8dc965', '1508*', '287c1452-620d-4195-9f19-c9814ef90d78', 'ed649e33-e771-403a-8c99-1780eabbc803');
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insert into phone_numbers (phone_number_sid, number, voip_carrier_sid, account_sid)
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values ('d458bf7a-bcea-47b2-ac96-66dfc9c5c220', '150822233*', '287c1452-620d-4195-9f19-c9814ef90d78', 'ed649e33-e771-403a-8c99-1780eabbc803');
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insert into phone_numbers (phone_number_sid, number, voip_carrier_sid, account_sid)
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values ('f7ad205d-b92f-4363-8160-f8b5216b40d3', '15083871234', '287c1452-620d-4195-9f19-c9814ef90d78', 'd7cc37cb-d152-49ef-a51b-485f6e917089');
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118
test/scenarios/uac-did-regex-match.xml
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118
test/scenarios/uac-did-regex-match.xml
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@@ -0,0 +1,118 @@
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<?xml version="1.0" encoding="ISO-8859-1" ?>
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<!DOCTYPE scenario SYSTEM "sipp.dtd">
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<!-- This program is free software; you can redistribute it and/or -->
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<!-- modify it under the terms of the GNU General Public License as -->
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<!-- published by the Free Software Foundation; either version 2 of the -->
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<!-- License, or (at your option) any later version. -->
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<!-- -->
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<!-- This program is distributed in the hope that it will be useful, -->
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<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
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<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
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<!-- GNU General Public License for more details. -->
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<!-- -->
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<!-- You should have received a copy of the GNU General Public License -->
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<!-- along with this program; if not, write to the -->
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<!-- Free Software Foundation, Inc., -->
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<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
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<!-- -->
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<!-- Sipp 'uac' scenario with pcap (rtp) play -->
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<!-- -->
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<scenario name="UAC with media">
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<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
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<!-- generated by sipp. To do so, use [call_id] keyword. -->
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<send retrans="500">
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<![CDATA[
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INVITE sip:+15082223333@jambonz.org SIP/2.0
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Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
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To: <sip:15082223333@jambonz.org>
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Call-ID: [call_id]
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CSeq: 1 INVITE
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Contact: sip:sipp@[local_ip]:[local_port]
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Max-Forwards: 70
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Subject: uac-pcap-carrier-success
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Content-Type: application/sdp
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Content-Length: [len]
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v=0
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o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
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s=-
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c=IN IP[local_ip_type] [local_ip]
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t=0 0
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m=audio [auto_media_port] RTP/AVP 8 101
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a=rtpmap:8 PCMA/8000
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a=rtpmap:101 telephone-event/8000
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a=fmtp:101 0-11,16
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]]>
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</send>
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<recv response="100" optional="true">
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</recv>
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<recv response="180" optional="true">
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</recv>
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<!-- By adding rrs="true" (Record Route Sets), the route sets -->
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<!-- are saved and used for following messages sent. Useful to test -->
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<!-- against stateful SIP proxies/B2BUAs. -->
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<recv response="200" rtd="true" crlf="true">
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</recv>
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<!-- Packet lost can be simulated in any send/recv message by -->
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<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
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<send>
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<![CDATA[
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ACK sip:15082223333@jambonz.org SIP/2.0
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Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
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To: <sip:15082223333@jambonz.org>[peer_tag_param]
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Call-ID: [call_id]
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CSeq: 1 ACK
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Max-Forwards: 70
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Subject: uac-pcap-carrier-success
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Content-Length: 0
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]]>
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</send>
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<!-- Play a pre-recorded PCAP file (RTP stream) -->
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<nop>
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<action>
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<exec play_pcap_audio="pcap/g711a.pcap"/>
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</action>
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</nop>
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<!-- Pause briefly -->
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<pause milliseconds="3000"/>
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<!-- The 'crlf' option inserts a blank line in the statistics report. -->
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<send retrans="500">
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<![CDATA[
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BYE sip:15082223333@jambonz.org SIP/2.0
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Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
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To: <sip:15082223333@jambonz.org>[peer_tag_param]
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Call-ID: [call_id]
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CSeq: 2 BYE
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Subject: uac-pcap-carrier-success
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Content-Length: 0
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]]>
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</send>
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<recv response="200" crlf="true">
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</recv>
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<!-- definition of the response time repartition table (unit is ms) -->
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<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
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<!-- definition of the call length repartition table (unit is ms) -->
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<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
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</scenario>
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@@ -39,6 +39,9 @@ test('incoming call tests', async(t) => {
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await sippUac('uac-pcap-pbx-success.xml', '172.38.0.21');
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t.pass('incoming call from account-level carrier completed successfully');
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await sippUac('uac-did-regex-match.xml', '172.38.0.20');
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t.pass('incoming call matched by trailing wildcard *');
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await sippUac('uac-pcap-device-success.xml', '172.38.0.30');
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t.pass('incoming call from authenticated device completed successfully');
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@@ -60,7 +63,7 @@ test('incoming call tests', async(t) => {
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await waitFor(10);
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const res = await queryCdrs({account_sid: 'ed649e33-e771-403a-8c99-1780eabbc803'});
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console.log(`cdrs: ${JSON.stringify(res)}`);
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t.ok(6 === res.total, 'successfully wrote 6 cdrs for calls');
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t.ok(7 === res.total, 'successfully wrote 7 cdrs for calls');
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srf.disconnect();
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t.end();
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