asterisk.ru

Здраствуйте

При использовании в качестве платформы IP02 http://www.atcom.cn/IP02.html#~tab-smal ... nterprises
Asterisk/1.4.21.2
GUI version: 2.0.2-ce
получаю ворнинг:
try_calling: The device state of this queue member, Agent/7019, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings.

Добавил в sip.conf
[general]
canreinvite =
compactheaders = no
context = default
defaultexpiry = 1800
domain =
dtmfmode = inband
dumphistory = no
externrefresh = 10
fromdomain =
limitonpeer = yes
maxexpiry = 300
minexpiry = 60
nat =
notifyringing = no
pedantic = no
progressinband = yes
promiscredir = no
realm = asterisk
recordhistory = no
relaxdtmf = yes
rtpholdtimeout =
rtptimeout =
sendrpid = no
sipdebug = no
srvlookup = yes
subscribecontext =
t38pt_udptl = yes
trustrpid = no
useragent = Asterisk PBX
disallow = all
videosupport = yes
allow = alaw,ulaw,h263


В users.conf
[7019]
username = 7019
transfer = yes
mailbox = 7019
registersip = no
host = dynamic
callgroup = 1
context = DLPN_DialPlan1
cid_number = 7019
hasvoicemail = no
vmsecret = ******
email = **************
threewaycalling = yes
hasdirectory = yes
callwaiting = yes
hasmanager = no
hasagent = yes
enablecallrecord = no
hassip = yes
hasiax = no
secret = *********
nat = yes
canreinvite = no
dtmfmode = inband
relaxdtmf = yes
insecure = port,invite
pickupgroup = 1
autoprov = no
label =
macaddress =
linenumber = 1
flash = 0
rxflash = 0
type = friend
call-limit = 10
disallow = all
allow = alaw,ulaw,g729,g726,gsm

В чем может быть проблема?

Did you check UPGRADE.txt for correct configuration settings?

Опять услуги переводчика нужны? Я тут! Перевожу банальные фразы - дорого.

callcounter еще не забывайте.

Делал как описано в UPGRADE.txt

* Queues depend on the channel driver reporting the proper state
for each member of the queue. To get proper signalling on
queue members that use the SIP channel driver, you need to
enable a call limit (could be set to a high value so it
is not put into action) and also make sure that both inbound
and outbound calls are accounted for.

Example:

[general]
limitonpeer = yes

[peername]
type=friend
call-limit=10

Вероятно я что-то пропустил, подскажите пожалуйста.

Samael28 писал(а):callcounter еще не забывайте.

callcounter разве не в 1.6 появился?

Где в тексте UPGRADE.txt Вы увидели упоминание users.conf?

Цитируете одно,

Example:
[general]
limitonpeer = yes

[peername]
type=friend
call-limit=10

а делаете другое

Добавил в sip.conf
[general]
........
В users.conf
[7019]
........