Testing multiple DID Numbers

[zaryabkhan (Muhammad Zaryab Khan)]

I have about large amount of DID’s that terminate to our system. I need to test these numbers to make sure that they all work properly. Without calling every DID by hand and get the results.please help me out

[tomeko (Tomasz Ostrowski)]

Get a tSIP softphone, configure account for outgoing calls, run from “Tools” menu script like this: tSIP softphone: call numbers from list · GitHub. To determine number status use call history (this would be JSON), call recording or GetCallState() function, writing status to text file, maybe in CSV format.

Note: these numbers should be actually strings, enclosed in quotation marks (this would be important for numbers with 9+ digits). If those are random numbers, use spreadsheet to generate list, if they are subsequent you may use some loop with starting point.

[zaryabkhan (Muhammad Zaryab Khan)]

I’m using the Linux operating system, but it supports only the Windows operating system. Is there any solution?

[dicko]

GitHub - nils-ohlmeier/sipsak: SIP swiss army knife

it’s been around for longer than most

[zaryabkhan (Muhammad Zaryab Khan)]

@tomeko

Get a tSIP softphone, configure account for outgoing calls, run from “Tools” menu script like this: tSIP softphone: call numbers from list · GitHub . To determine number status use call history (this would be JSON), call recording or GetCallState() function, writing status to text file, maybe in CSV format.

Note: these numbers should be actually strings, enclosed in quotation marks (this would be important for numbers with 9+ digits). If those are random numbers, use spreadsheet to generate list, if they are subsequent you may use some loop with starting point.

How do I know if the number is working or not? The log file provides me with this information.
sip:[email protected];transport=udp: session closed: Connection timed out
Callback::CALL_STATE, call uid = 13, state = 0
connecting to ‘sip:[email protected];transport=udp’…
Callback::CALL_STATE, call uid = 14, state = 2
terminate call with sip:[email protected];transport=udp
Callback::CALL_STATE, call uid = 14, state = 0
connecting to ‘sip:[email protected];transport=udp’…
Callback::CALL_STATE, call uid = 15, state = 2
terminate call with sip:[email protected];transport=udp
Callback::CALL_STATE, call uid = 15, state = 0
connecting to ‘sip:[email protected];transport=udp’…
Callback::CALL_STATE, call uid = 16, state = 2
terminate call with sip:[email protected];transport=udp
Callback::CALL_STATE, call uid = 16, state = 0

[tomeko (Tomasz Ostrowski)]

@zaryabkhan

connecting to ‘sip:[email protected];transport=udp’…
Callback::CALL_STATE, call uid = 14, state = 2

That’s CALL_STATE_OUTGOING and there is no transition to TRYING/RINGING/PROGRESS, but before making any calls other than direct IP calls you have to configure account (unless you really have some PABX on 127.0.0.1).

There is also no universal condition allowing to distinguish “working” and not working numbers, this depends on operators and configuration. Sometimes it would be visible through SIP messages, sometimes this might be only specific audio announcement transmitted with early media - that’s when recording (and maybe automated transcribing) might be needed.