Jul
03
2017
0

How to connect Asterisk SIP Clients over Duel Layers of NAT

A common problem with SIP is it’s difficulty handling NAT(especially over multiple layers).  A common symptom is no audio stream.  The Asterisk CLI  shows a successful call connection succeeding followed by an error in /var/log/asterisk/messages:

[Jun 14 17:16:26] WARNING[2967] chan_sip.c: Retransmission timeout reached on transmission 4e2a8586-3a91c1cf-e73f6b64@10.1.10.115 for seqno 2 (Critical Response) — See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response

(more…)

Apr
22
2009
1

queues.conf – agent logoff example

[general]

monitor-type=MixMonitor
eventmemberstatus=no
eventwhencalled=no

;[default]
;
; Default settings for queues (currently unused)
;

[savelono-support]

context = savelono-queue-out
musiconhold = default
strategy = leastrecent
timeout = 10
retry = 5
wrapuptime=20
announce-frequency = 90
announce-holdtime = no
announce-round-seconds = 30
queue-youarenext = queue-youarenext
queue-thereare = queue-thereare
queue-callswaiting = queue-callswaiting
queue-thankyou = queue-thankyou
monitor-format = wav
monitor-join = yes
joinempty = yes
eventmemberstatus=no
autofill=yes
;memberdelay = 1

member => Agent/8888