Eureka! First SIP Call Done :)

Asterisk Asterisk Endpoints
1

But I am wondering why mobile to mobile (SIP Calling) is showing always congested. Any limitations?
[Jul 26 17:38:03] WARNING[2513]: res_pjsip_pubsub.c:3396 pubsub_on_rx_publish_request: No registered publish handler for event presence from SOFTPHONE_B
– Executing [102@sets:1] Dial(“PJSIP/SOFTPHONE_B-00000050”, “PJSIP/SOFTPHONE_A”) in new stack
– Called PJSIP/SOFTPHONE_A
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘PJSIP/SOFTPHONE_B-00000050’ status is ‘CONGESTION’

2

You need to enable the full log, then issue “pjsip set logger on”, and provide the resulting log for a call. SOFTPHONE_A has rejected the call.

3

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