Everyone is busy/congested at this time

[prathibhacdac]

Sometimes the call is getting connected but most of the time I am getting the error in the subject.

Executing [6238900031@Internal:1] NoOp("SIP/201004-00000138", "New Outgoing call from 201004") in new stack
-- Executing [6238900031@Internal:2] Set("SIP/201004-00000138", "CALLERLEN=6") in new stack
-- Executing [6238900031@Internal:3] GotoIf("SIP/201004-00000138", "0?rotate") in new stack
-- Executing [6238900031@Internal:4] Set("SIP/201004-00000138", "CRMSTATE=NOT_INUSE") in new stack
-- Executing [6238900031@Internal:5] GotoIf("SIP/201004-00000138", "0?Kill,crmoffline,1") in new stack
-- Executing [6238900031@Internal:6] Set("SIP/201004-00000138", "GLOBAL(OUTGW)=2") in new stack
== Setting global variable 'OUTGW' to '2'
-- Executing [6238900031@Internal:7] NoOp("SIP/201004-00000138", "2") in new stack
-- Executing [6238900031@Internal:8] GotoIf("SIP/201004-00000138", "0?call") in new stack
-- Executing [6238900031@Internal:9] Set("SIP/201004-00000138", "GLOBAL(OUTGW)=1") in new stack
== Setting global variable 'OUTGW' to '1'
-- Executing [6238900031@Internal:10] NoOp("SIP/201004-00000138", "INVALID") in new stack
-- Executing [6238900031@Internal:11] Dial("SIP/201004-00000138", "SIP/6238900031,20,tT") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [6238900031@Internal:12] Hangup("SIP/201004-00000138", "") in new stack

[InnovateAsterisk]

The Browser Phone is offline at the time to call is sent to the browser. Check the CLI at the time of the failure.

[prathibhacdac]

How to bring the offline phone to online mode on receiving a call? I am using push notification. The browser phone comes to the front but the answer window doesn't appear. Instead the websocket registration page appears and the ring tone is played in the back.

[InnovateAsterisk]

This is the solution we provide via Siperb, and one of the primary reasons for the proxy configuration and service we provide.

The proxy receives the invite, send the push notifications, and waits for the user to register, and on registration it sends the call to the user. It can send the call to multiple devices, and the first user to answer will receive the call.

[prathibhacdac]

How to identify when the user registration is completed?

[prathibhacdac]

Can we identify whether the registration return 200 OK from DialByLine()?

[InnovateAsterisk]

With the register web hooks, you can identify if the user is registered, and then use a script like fetch to signal to asterisk that the user came online. However this is only half the problem. You need to get asterisk to take the call in without answering it yet, but make it wait until it receives this signal that the user is online. The last time is checked this was not possible, unless someone has built a push notifications module.

[prathibhacdac]

which proxy server are you using? Will Siperb be open sourced?

[InnovateAsterisk]

Siperb is a big system, and will eventually have points of presence around the world. It will consist of a Web based Softphone and mobile Softphone. These will use that global cloud network of proxy servers. The web part will be open source, while the mobile part will not be, neither the global network of proxy servers. It will be free to use too, but if you consume more than the free tier of space, or if you use transcoding, or if you use the AI voice tools, you will need to go to a paid account.

[prathibhacdac]

Have you used flexisip in Siprb?

[InnovateAsterisk]

We use various technologies. For the proxy we use OpenSIPS

[prathibhacdac]

Are you using the registrar module in opensips for Push Notification?

[InnovateAsterisk]

It’s based on this, but mostly my own work:
https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/

[prathibhacdac]

Which version of opensips you used - latest one or 3.1?

[InnovateAsterisk]

I believe it’s 3.4