Number error when transferring a call?

Question

Asterisk

Greetings, Comrades!
I ran into a problem when transferring a call using Asterisk, tobish through # They
called, talked, you need to transfer. I press # and a pleasant female voice says: Transferring the call, and the interlocutor starts playing music, then as soon as I dial the first digit of the internal number, the answer is that the number does not exist. Although I didn’t even dial the number to the end, it instantly cuts off. Here are the logs through -rvvvvv when calling and trying to transfer:

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 [[email protected]_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:[email protected]>;tag=as1e528952'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [[email protected]_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'

Answer 1

By default, call transfer looks for extensions in the same context from which the Dial was called, in your case, the trunk_check context. If you need to use extensions of another context, then first set it in the global variable TRANSFER_CONTEXT
Set(__TRANSFER_CONTEXT=...)