Приветствую, Камрады!
Столкнулся с проблемой при переводе звонка средствами Астериска, тобишь через #
Позвонили, пообщались, нужно перевести. нажимаю # и приятный женский голос говорит: Перевод звонка, а у собеседника начинает играть музыка, то как только я набираю первую цифру внутреннего номера, то мне в ответ что номер не существует. Хотя я даже не до конца набрал номер, моментально отсекает. Вот логи через -rvvvvv при звонке и попытке перевода:
Connected to Asterisk 13.14.0 currently running on voip (pid = 1426)
== Using SIP RTP CoS mark 5
-- Executing [<mobile_num>@phones:1] Gosub("SIP/125-00001b0c", "trunk_check,s,1(<mobile_num>)") in new stack
-- Executing [s@trunk_check:1] Dial("SIP/125-00001b0c", "SIP/<mobile_num>@213137,45,t") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/<mobile_num>@213137
[May 2 13:37:45] WARNING[2366][C-00000c1c]: chan_sip.c:23875 handle_response_invite: Received response: "Forbidden" from '"Ryzhkin S.N." <sip:213137@10.9.3.7>;tag=as1e528952'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@trunk_check:2] Dial("SIP/125-00001b0c", "SIP/<mobile_num>@213135,45,t") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/<mobile_num>@213135
> 0x7fd19c013d70 -- Probation passed - setting RTP source address to 10.10.12.199:27584
[May 2 13:37:48] WARNING[2387]: netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null)
-- SIP/213135-00001b0e is ringing
-- SIP/213135-00001b0e is making progress passing it to SIP/125-00001b0c
> 0x7fd19c013d70 -- Probation passed - setting RTP source address to 10.10.12.199:27584
> 0x7fd18807d140 -- Probation passed - setting RTP source address to 192.168.50.21:11940
-- SIP/213135-00001b0e answered SIP/125-00001b0c
-- Channel SIP/213135-00001b0e joined 'simple_bridge' basic-bridge <67ee09ec-a517-4b49-b4eb-818f2bcbe536>
-- Channel SIP/125-00001b0c joined 'simple_bridge' basic-bridge <67ee09ec-a517-4b49-b4eb-818f2bcbe536>
> 0x7fd19c013d70 -- Probation passed - setting RTP source address to 10.10.12.199:27584
-- Started music on hold, class 'default', on channel 'SIP/125-00001b0c'
-- <SIP/213135-00001b0e> Playing 'pbx-transfer.slin' (language 'ru')
-- <SIP/213135-00001b0e> Playing 'pbx-invalid.slin' (language 'ru')
-- Stopped music on hold on SIP/125-00001b0c
-- Channel SIP/213135-00001b0e left 'simple_bridge' basic-bridge <67ee09ec-a517-4b49-b4eb-818f2bcbe536>
-- Channel SIP/125-00001b0c left 'simple_bridge' basic-bridge <67ee09ec-a517-4b49-b4eb-818f2bcbe536>
== Spawn extension (trunk_check, s, 2) exited non-zero on 'SIP/125-00001b0c'
