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2020 Mar 18
2
congested/busy on trunk?
...'core show license' for details. ========================================================================= Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 602055) dunkel*CLI> <--- Received SIP request (940 bytes) from TCP:[2605:e000:130a:fb:de1:71fc:e257:6f4e]:37879 ---> INVITE sip:13107950860 at dunkel.dev1ce.com;transport=tcp SIP/2.0 Call-ID: 26b4820ac727b0f23fea131b7c5cd450 at 2605:e000:130a:fb:de1:71fc:e257:6f4e CSeq: 8612 INVITE From: "demo-alice" <sip:demo-alice at dunkel.dev1ce.com;transport=tcp>;tag=3166828162 To: <sip:1310...
2020 Mar 14
2
congested/busy on trunk?
greetings asterisk users :) ive just deployed version 17 and migrated as best I can to pjsip. I can receive calls, and get to my mailbox prompt, however placing calls seems impossible with the following error on dial: Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 517890) dunkel*CLI> dunkel*CLI> == Setting global variable 'SIPDOMAIN' to
2003 May 23
12
Unable to create channel of type 'Zap'
I've just installed an X100P, built the kernel module, and tried to use it to make an outgoing call (via a phone connected to an ATA-186). However, I just get a reorder tone and see this on the console: -- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type