Has anyone been successful at dialing into a Webex meeting?
I have in extensions.conf:
exten = 2222,1,Dial(SIPfirstname.lastname@example.org)
When I dialed 2222, the CLI shows the call being made:
== Using SIP RTP CoS mark 5 >0xb410ead8 -- Strict RTP learning after remote address set to: 192.168.7.11:16466 -- Executing [2223@from-internal:1] Dial("SIP/8001-00000010", "SIPemail@example.com") in new stack == Using SIP RTP CoS mark 5 -- Called SIPfirstname.lastname@example.org [May 27 20:04:11]WARNING: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission email@example.com:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [May 27 20:04:11] WARNING: chan_sip.c:4143 retrans_pkt: Hanging up call firstname.lastname@example.org:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). -- SIP/meetingsapac24.webex.com-00000011 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/8001-00000010' status is 'CONGESTION'
That status on the last line alternates between
CHANUNAVAIL each time I tried.
I tried putting my asterisk server in the DMZ, meaning that it has direct access to the Internet without going through NAT, but the responses were the same.
What are some of the prerequisites or gotchas when dialing to join a Webex meeting?