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Make SIP Number and SIP Client tied to an account and Unique per Organization #2106

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maria-farooq opened this issue Apr 26, 2017 · 4 comments
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@maria-farooq
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maria-farooq commented Apr 26, 2017

Use Cases:

  • Add new sip number should be possible even if that number exists in a different organization
  • Add new sip client should be possible even if that number exists in a different organization
  • Dialling a sip number means dialling that number of your(dialling client's) organization.
  • Dialling a sip client means dialling that number of your(dialling client's) organization.
  • upon sub account creation, tie the sub account to the same org as the parent.
  • Register clients with same login under their respective organizations.
  • monitoringservice metrics for registered users should distinctly identify users of all organizations #2128
  • verify existing test suit
  • update db changes in test suit
  • add new test in test suit
  • add upgrade scripts ( for numbers and client table changes)

@scottbarstow @deruelle do we want to support inter calls between client-to-number and client-to-client of different organizations?
For Example:
[email protected] dials [email protected]
or
[email protected] dials [email protected]
do you think that it is part of scope of this ticket or we should create a different issue for that?

@deruelle
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@maria-farooq looks good to me.

[email protected] dials [email protected] and [email protected] dials [email protected] only via Dial SIP not via dial client or dial number or dial conference, all of those should stay within an organization

maria-farooq pushed a commit that referenced this issue Apr 26, 2017
@scottbarstow
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@maria-farooq this looks right to me as well. Jean can speak better to the inter-org, but what I'm not sure about is if we'd want to treat inter-org calls in your example as off-net, but I think this is a billing decision not a technical one. @deruelle comment on Dial SIP sounds right, tho I'm not deep in the weeds on that yet.

Either way I think we should treat that as separate ticket so that we can include test cases separately.

maria-farooq pushed a commit that referenced this issue Apr 26, 2017
…, updated getClientByLogin to include organization sid #2106
maria-farooq pushed a commit that referenced this issue May 1, 2017
maria-farooq pushed a commit that referenced this issue May 1, 2017
maria-farooq pushed a commit that referenced this issue May 4, 2017
added organizationSid in registrations dao and xml mapper files #2106
maria-farooq pushed a commit that referenced this issue May 5, 2017
maria-farooq pushed a commit that referenced this issue May 5, 2017
maria-farooq pushed a commit that referenced this issue May 10, 2017
@maria-farooq
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maria-farooq commented May 10, 2017

test suit packages verification

  • org.restcomm.connect.testsuite
  • org.restcomm.connect.testsuite.faultTolerance
  • org.restcomm.connect.testsuite.http
  • org.restcomm.connect.testsuite.provisioning.number.bandwidth
  • org.restcomm.connect.testsuite.provisioning.number.nexmo
  • org.restcomm.connect.testsuite.provisioning.number.vi (3 failing but they are failing in master as well)
  • org.restcomm.connect.testsuite.provisioning.number.voxbone (2 failing but they are failing in master as well)
  • org.restcomm.connect.testsuite.telephony.proxy
  • org.restcomm.connect.testsuite.telephony.ua
  • org.restcomm.connect.testsuite.smpp
  • org.restcomm.connect.testsuite.sms
  • org.restcomm.connect.testsuite.telephony

@maria-farooq
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maria-farooq commented May 17, 2017

Add New test for the feature:

  • Add same number in 2 (pre-configured) organizations @ org (other than 127.0.0.1)
  • Add same clients in 2 (pre-configured) organizations @ org (other than 127.0.0.1)
  • Dial number @ org (other than 127.0.0.1)
  • Dial client @ org (other than 127.0.0.1)
  • Dial sip from org1 to number@org2 (other than 127.0.0.1)
  • Dial sip from org1 to client@org2 (other than 127.0.0.1)
  • p2p messages between 2 clients of same organization
  • p2p messages between 2 clients of different organization (this one should fail)
  • inbound SMS terminating on app via SIP
  • inbound SMS terminating on app via SMPP
  • inbound USSD CALL terminating on app via SIP
  • Add registrations of 2 clients with same login in 2 (pre-configured) organizations
  • Test OPTIONS for registrations of 2 clients with same login in 2 (pre-configured) organizations

maria-farooq pushed a commit that referenced this issue May 17, 2017
* master:
  Minor patch for Restcomm testsuite failing tests
  Patch for CallManager to properly handle numbers that start with + This close #2158
  Patch to fix Outbound call never reach the completed state when inbound call sends BYE This close #2157
  Patch that fixes various Restcomm testsuite failing tests

Conflicts resolved:
	restcomm/restcomm.telephony/src/main/java/org/restcomm/connect/telephony/CallManager.java
#2106
maria-farooq pushed a commit that referenced this issue May 18, 2017
maria-farooq pushed a commit that referenced this issue Jun 11, 2017
maria-farooq pushed a commit that referenced this issue Jun 16, 2017
maria-farooq pushed a commit that referenced this issue Jun 29, 2017
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