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Fixed issue when wild cards or regex is used in phone number for multiple carriers (#202)

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* Fixed issue when wild cards or regex is used in phone number for multiple carriers
https://github.com/jambonz/sbc-inbound/issues/201

* Implemanted the regex/pattern based logic when url is a sip_realm

* Refactored getApplicationForDidAndCarrier as it is failing to parse other regex if first regex is invalid
In gateway is defined at "all accounts" using the phone number's account_sid

* Performance improvements

* Exported getApplicationForDidAndCarriers method

* Added additional unit test cases
This commit is contained in:
rammohan-y
2025-06-19 00:41:05 +05:30
committed by GitHub
parent 680d32e9a2
commit 5f267906b1
5 changed files with 397 additions and 5 deletions
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29
test/db/populate-test-data.sql
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@@ -92,3 +92,32 @@ insert into sip_gateways (sip_gateway_sid, voip_carrier_sid, ipv4, inbound, outb
values ('664a5339-c62c-4075-9e19-f4de70a96597', '731abdc7-0220-4964-bc66-32b5c70cd9ab', '172.38.0.40', true, false);
insert into sip_gateways (sip_gateway_sid, voip_carrier_sid, ipv4, inbound, outbound)
values ('554a5339-c62c-4075-9e19-f4de70a96597', '987abdc7-0220-4964-bc66-32b5c70cd9ab', '172.38.0.40', true, false);
-- voip carrier belonging to all accounts
insert into voip_carriers (voip_carrier_sid, name, service_provider_sid)
values ('voip100', 'test-voip-carrier', '3f35518f-5a0d-4c2e-90a5-2407bb3b36f0');
insert into sip_gateways (sip_gateway_sid, voip_carrier_sid, ipv4, inbound, outbound)
values ('sip100', 'voip100', '172.38.0.50', true, false);
insert into voip_carriers (voip_carrier_sid, name, service_provider_sid)
values ('voip101', 'test-voip-carrier-101', '3f35518f-5a0d-4c2e-90a5-2407bb3b36f0');
insert into sip_gateways (sip_gateway_sid, voip_carrier_sid, ipv4, inbound, outbound)
values ('sip101', 'voip101', '172.38.0.50', true, false);
insert into sip_gateways (sip_gateway_sid, voip_carrier_sid, ipv4, inbound, outbound)
values ('sip102', 'voip101', '172.38.0.51', true, false);
insert into applications (application_sid, name, account_sid, call_hook_sid, call_status_hook_sid)
values ('app100', 'app100', 'ee9d7d49-b3e4-4fdb-9d66-661149f717e8', '90dda62e-0ea2-47d1-8164-5bd49003476c', '90dda62e-0ea2-47d1-8164-5bd49003476c');
insert into applications (application_sid, name, account_sid, call_hook_sid, call_status_hook_sid)
values ('app101', 'app101', 'ee9d7d49-b3e4-4fdb-9d66-661149f717e8', '90dda62e-0ea2-47d1-8164-5bd49003476c', '90dda62e-0ea2-47d1-8164-5bd49003476c');
insert into phone_numbers (phone_number_sid, number, voip_carrier_sid, account_sid, application_sid)
values ('phone100', '^100', 'voip101', 'ee9d7d49-b3e4-4fdb-9d66-661149f717e8', 'app100');
insert into phone_numbers (phone_number_sid, number, voip_carrier_sid, account_sid, application_sid)
values ('phone101', '^10012', 'voip100', 'ee9d7d49-b3e4-4fdb-9d66-661149f717e8', 'app101');
-- insert an invalid regex pattern, the below pattern should be ignored during pattern
insert into phone_numbers (phone_number_sid, number, voip_carrier_sid, account_sid, application_sid)
values ('phone102', '\\dkjfhmdf\\', 'voip100', 'ee9d7d49-b3e4-4fdb-9d66-661149f717e8', 'app101');
-- account with a sip realm that is not associated with any voip carriers
insert into accounts (account_sid, name, service_provider_sid, webhook_secret, sip_realm)
values ('acct-100', 'Account 100', '3f35518f-5a0d-4c2e-90a5-2407bb3b36f0', 'foobar', 'ram.sip.jambonz.org');

118
test/scenarios/uac-did-regex-match-vc-all-accts-nosiprealm.xml Normal file
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@@ -0,0 +1,118 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp 'uac' scenario with pcap (rtp) play -->
<!-- -->
<scenario name="UAC with media">
<!-- 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:100678@172.38.0.10:5060 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: <sip:100678@172.38.0.10:5060>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-pcap-carrier-success
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" 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" crlf="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:100678@172.38.0.10:5060 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: <sip:100678@172.38.0.10:5060>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Max-Forwards: 70
Subject: uac-pcap-carrier-success
Content-Length: 0
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<!-- Pause briefly -->
<pause milliseconds="3000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:100678@172.38.0.10:5060 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: <sip:100678@172.38.0.10:5060>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Subject: uac-pcap-carrier-success
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>

118
test/scenarios/uac-did-regex-match-vc-all-accts.xml Normal file
View File

@@ -0,0 +1,118 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp 'uac' scenario with pcap (rtp) play -->
<!-- -->
<scenario name="UAC with media">
<!-- 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:+100678@ram.sip.jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: <sip:100678@ram.sip.jambonz.org>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-pcap-carrier-success
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" 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" crlf="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:100678@ram.sip.jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: <sip:100678@ram.sip.jambonz.org>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Max-Forwards: 70
Subject: uac-pcap-carrier-success
Content-Length: 0
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<!-- Pause briefly -->
<pause milliseconds="3000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:100678@ram.sip.jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: <sip:100678@ram.sip.jambonz.org>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Subject: uac-pcap-carrier-success
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>

10
test/sip-tests.js
View File

@@ -64,6 +64,15 @@ test('incoming call tests', async(t) => {
await sippUac('uac-pcap-carrier-max-call-limit.xml', '172.38.0.20');
t.pass('rejects incoming call with 503 when max calls per account reached');
await sippUac('uac-did-regex-match-vc-all-accts.xml', '172.38.0.50');
t.pass('incoming call matched by trailing wildcard *, voice gateway belongs to all accounts, with sip realm');
await sippUac('uac-did-regex-match-vc-all-accts-nosiprealm.xml', '172.38.0.51');
t.pass('incoming call matched by trailing wildcard *, voice gateway belongs to all accounts, without sip realm');
await sippUac('uac-did-regex-match-vc-all-accts-nosiprealm.xml', '172.38.0.50');
t.pass('incoming call matched by trailing wildcard *, voice gateway belongs to all accounts, without sip realm');
/* switch off this env for remaining tests (JAMBONES_HOSTING is for Saas sts) */
delete process.env.JAMBONES_HOSTING;
await sippUac('uac-pcap-carrier-fail-ambiguous.xml', '172.38.0.40');
@@ -71,6 +80,7 @@ test('incoming call tests', async(t) => {
await waitFor(12);
const res = await queryCdrs({account_sid: 'ed649e33-e771-403a-8c99-1780eabbc803'});
console.log(`cdrs res.total: ${res.total}`);
//console.log(`cdrs: ${JSON.stringify(res)}`);
t.ok(7 === res.total, 'successfully wrote 8 cdrs for calls');