asterisk σαν server

[atux_null]

Το εσωτερικό 500 καλεί 450 και κλήση επιστρέφει στον ίδιο. Κατόπιν το 500 καλεί τον 400 που είναι ένα εσωτερικό του Asterisk_Server

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
       > 0x561a951fff70 -- Strict RTP learning after remote address set to: 192.168.1.116:5004
    -- Executing [450@AllCalls:1] Set("SIP/500-00000000", "CALLERID(name)=") in new stack
    -- Executing [450@AllCalls:2] Answer("SIP/500-00000000", "") in new stack
       > 0x561a951fff70 -- Strict RTP switching to RTP target address 192.168.1.116:5004 as source
    -- Executing [450@AllCalls:3] Wait("SIP/500-00000000", "1") in new stack
    -- Executing [450@AllCalls:4] Dial("SIP/500-00000000", "SIP/500,20") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/500
    -- SIP/500-00000001 is ringing
       > 0x561a951fff70 -- Strict RTP learning complete - Locking on source address 192.168.1.116:5004
  == Spawn extension (AllCalls, 450, 4) exited non-zero on 'SIP/500-00000000'
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
       > 0x561a95204bf0 -- Strict RTP learning after remote address set to: 192.168.1.116:5004
[Jul  3 14:36:51] NOTICE[730][C-00000002]: chan_sip.c:26690 handle_request_invite: Call from '500' (192.168.1.116:5060) to extension '400' rejected because extension not found in context 'AllCalls'.
client_A*CLI>

[kronos911]

Δοκίμασε το παρακάτω στον server B
Κάλεσε εσωτερικό του Α
και ανέβασε log B και Α

client

[AllCalls]
exten => _4[01-4]X,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) 
exten => _4[01-4]X,n,Dial(SIP/450/${EXTEN},20) 
exten => _4[01-4]X,n,Busy(3)

[atux_null]

Δοκίμασε το παρακάτω στον server B
Κάλεσε εσωτερικό του Α
και ανέβασε log B και Α

client

[AllCalls]
exten => _4[01-4]X,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) 
exten => _4[01-4]X,n,Dial(SIP/450/${EXTEN},20) 
exten => _4[01-4]X,n,Busy(3)

στο [AllCalls] πρόσθεσα τις 3 γραμμές που αναφέρεις παραπάνω.
logs στον Server A δεν έχει τίποτα, ενώ στο Β έχει

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [400@AllCalls:1] Set("SIP/500-00000002", "CALLERID(name)=") in new stack
    -- Executing [400@AllCalls:2] Dial("SIP/500-00000002", "SIP/450/400,20") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/450/400
[Jul  3 15:11:19] WARNING[730][C-00000003]: channel.c:5589 set_format: Unable to find a codec translation path: (g729) -> (alaw)
[Jul  3 15:11:19] WARNING[730][C-00000003]: channel.c:5589 set_format: Unable to find a codec translation path: (alaw) -> (g729)
    -- SIP/450-00000003 answered SIP/500-00000002
[Jul  3 15:11:19] WARNING[1851][C-00000003]: channel.c:6549 ast_channel_make_compatible_helper: No path to translate from SIP/450-00000003 to SIP/500-00000002
[Jul  3 15:11:19] WARNING[1851][C-00000003]: app_dial.c:3244 dial_exec_full: Had to drop call because I couldn't make SIP/500-00000002 compatible with SIP/450-00000003
  == Spawn extension (AllCalls, 400, 2) exited non-zero on 'SIP/500-00000002'
client_A*CLI>

Ο 500 προσπάθησε να καλέσει τον 400

[kronos911]

στο [AllCalls] πρόσθεσα τις 3 γραμμές που αναφέρεις παραπάνω.
logs στον Server A δεν έχει τίποτα, ενώ στο Β έχει

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [400@AllCalls:1] Set("SIP/500-00000002", "CALLERID(name)=") in new stack
    -- Executing [400@AllCalls:2] Dial("SIP/500-00000002", "SIP/450/400,20") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/450/400
[Jul  3 15:11:19] WARNING[730][C-00000003]: channel.c:5589 set_format: Unable to find a codec translation path: (g729) -> (alaw)
[Jul  3 15:11:19] WARNING[730][C-00000003]: channel.c:5589 set_format: Unable to find a codec translation path: (alaw) -> (g729)
    -- SIP/450-00000003 answered SIP/500-00000002
[Jul  3 15:11:19] WARNING[1851][C-00000003]: channel.c:6549 ast_channel_make_compatible_helper: No path to translate from SIP/450-00000003 to SIP/500-00000002
[Jul  3 15:11:19] WARNING[1851][C-00000003]: app_dial.c:3244 dial_exec_full: Had to drop call because I couldn't make SIP/500-00000002 compatible with SIP/450-00000003
  == Spawn extension (AllCalls, 400, 2) exited non-zero on 'SIP/500-00000002'
client_A*CLI>

Ο 500 προσπάθησε να καλέσει τον 400

codec error
βάλε μόνο alaw παντού.

[atux_null]

Περίεργο για το codec error. όλα τα άκρα υποστηρίζουν όλους τους codec και θα έπρεπε να κάνουν renegotiate! Παρόλα αυτά έβαλα μόνο alaw και τώρα το 500 μπορεί να καλέσει το 400. Στην οθόνη του 400 φαίνεται το 450, που είναι το trunk. Σωστό.
Το αντίθετο δεν γίνεται, όπου ο 400 προσπαθεί να καλέσει το 450, server logs:

server*CLI>
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [450@AllCalls:1] Goto("SIP/400-00000016", "DefaultPlan,450,1") in new stack
    -- Goto (DefaultPlan,450,1)
    -- Executing [450@DefaultPlan:1] Answer("SIP/400-00000016", "") in new stack
    -- Executing [450@DefaultPlan:2] VoiceMail("SIP/400-00000016", "450@Office,u") in new stack
[Jul  3 15:27:37] WARNING[3844][C-00000012]: app_voicemail.c:6549 leave_voicemail: No entry in voicemail config file for '450'
    -- Executing [450@DefaultPlan:3] Hangup("SIP/400-00000016", "") in new stack
  == Spawn extension (DefaultPlan, 450, 3) exited non-zero on 'SIP/400-00000016'
server*CLI>

ο κώδικας

[AllCalls]
exten => _4[01-4]X,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) 
exten => _4[01-4]X,n,Dial(SIP/450/${EXTEN},20) 
exten => _4[01-4]X,n,Busy(3)

Χρειάζεται?

[kronos911]

Στον server A πρόσθεσε τα παρακάτω στο [DefaultPlan]

exten => _5XX,1,Set(CALLERID(num)=${EXTEN}) 
exten => _5XX,n,Dial(SIP/450/${EXTEN},20) 
exten => _5XX,n,Busy(3)

Στον server B άλλαξε τη γραμμή

exten => _4[01-4]X,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})

σε

exten => _4[01-4]X,1,Set(CALLERID(num)=${EXTEN})

logs από server A και Β

[atux_null]

To 400 καλεί το 450 και αποτυγχάνει. Κατόπιν το 500 καλεί το 400 και η κλήση βγαίνει. logs:
Server:

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [450@AllCalls:1] Goto("SIP/400-0000001b", "DefaultPlan,450,1") in new stack
    -- Goto (DefaultPlan,450,1)
    -- Executing [450@DefaultPlan:1] Answer("SIP/400-0000001b", "") in new stack
    -- Executing [450@DefaultPlan:2] VoiceMail("SIP/400-0000001b", "450@Office,u") in new stack
[Jul  3 15:56:08] WARNING[3936][C-00000015]: app_voicemail.c:6549 leave_voicemail: No entry in voicemail config file for '450'
    -- Executing [450@DefaultPlan:3] Hangup("SIP/400-0000001b", "") in new stack
  == Spawn extension (DefaultPlan, 450, 3) exited non-zero on 'SIP/400-0000001b'
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [450@AllCalls:1] Goto("SIP/400-0000001c", "DefaultPlan,450,1") in new stack
    -- Goto (DefaultPlan,450,1)
    -- Executing [450@DefaultPlan:1] Answer("SIP/400-0000001c", "") in new stack
    -- Executing [450@DefaultPlan:2] VoiceMail("SIP/400-0000001c", "450@Office,u") in new stack
[Jul  3 15:56:16] WARNING[3937][C-00000016]: app_voicemail.c:6549 leave_voicemail: No entry in voicemail config file for '450'
    -- Executing [450@DefaultPlan:3] Hangup("SIP/400-0000001c", "") in new stack
  == Spawn extension (DefaultPlan, 450, 3) exited non-zero on 'SIP/400-0000001c'
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [400@AllCalls:1] Answer("SIP/450-0000001d", "") in new stack
    -- Executing [400@AllCalls:2] Dial("SIP/450-0000001d", "SIP/400,60") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/400
    -- SIP/400-0000001e is ringing
    -- SIP/400-0000001e answered SIP/450-0000001d
    -- Channel SIP/400-0000001e joined 'simple_bridge' basic-bridge <d84381d3-c1b2-43ec-a656-c68284f29925>
    -- Channel SIP/450-0000001d joined 'simple_bridge' basic-bridge <d84381d3-c1b2-43ec-a656-c68284f29925>
    -- Channel SIP/400-0000001e left 'native_rtp' basic-bridge <d84381d3-c1b2-43ec-a656-c68284f29925>
    -- Channel SIP/450-0000001d left 'native_rtp' basic-bridge <d84381d3-c1b2-43ec-a656-c68284f29925>
  == Spawn extension (AllCalls, 400, 2) exited non-zero on 'SIP/450-0000001d'
server*CLI>

client

[Jul  3 15:55:32] WARNING[730][C-00000005]: chan_sip.c:17379 check_auth: username mismatch, have <500>, digest has <s>
[Jul  3 15:55:32] NOTICE[730][C-00000005]: chan_sip.c:26596 handle_request_invite: Failed to authenticate device "400" <sip:500@192.168.1.46>;tag=as7b56b474
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [400@AllCalls:1] Set("SIP/500-00000006", "CALLERID(num)=400") in new stack
    -- Executing [400@AllCalls:2] Dial("SIP/500-00000006", "SIP/450/400,20") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/450/400
    -- SIP/450-00000007 answered SIP/500-00000006
    -- Channel SIP/450-00000007 joined 'simple_bridge' basic-bridge <be681db2-2bbf-4e6e-b10b-8aa401584fae>
    -- Channel SIP/500-00000006 joined 'simple_bridge' basic-bridge <be681db2-2bbf-4e6e-b10b-8aa401584fae>
    -- Channel SIP/500-00000006 left 'native_rtp' basic-bridge <be681db2-2bbf-4e6e-b10b-8aa401584fae>
    -- Channel SIP/450-00000007 left 'native_rtp' basic-bridge <be681db2-2bbf-4e6e-b10b-8aa401584fae>
  == Spawn extension (AllCalls, 400, 2) exited non-zero on 'SIP/500-00000006'
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [400@AllCalls:1] Set("SIP/500-00000008", "CALLERID(num)=400") in new stack
    -- Executing [400@AllCalls:2] Dial("SIP/500-00000008", "SIP/450/400,20") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/450/400
    -- SIP/450-00000009 answered SIP/500-00000008
    -- Channel SIP/450-00000009 joined 'simple_bridge' basic-bridge <38a83a00-b16f-4b69-ba87-87e43beb9e99>
    -- Channel SIP/500-00000008 joined 'simple_bridge' basic-bridge <38a83a00-b16f-4b69-ba87-87e43beb9e99>
    -- Channel SIP/450-00000009 left 'native_rtp' basic-bridge <38a83a00-b16f-4b69-ba87-87e43beb9e99>
    -- Channel SIP/500-00000008 left 'native_rtp' basic-bridge <38a83a00-b16f-4b69-ba87-87e43beb9e99>
  == Spawn extension (AllCalls, 400, 2) exited non-zero on 'SIP/500-00000008'
client_A*CLI>

[kronos911]

400 -> 500 please

- - - Updated - - -

Στο server A αφαίρεσε την γραμμή

exten => _5XX,1,Set(CALLERID(num)=${EXTEN})

και άλλαξε την από κάτω σε

exten => _5XX,1,Dial(SIP/450/${EXTEN},20)

[atux_null]

Το 400 καλεί το 500. Server logs:

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [500@AllCalls:1] Goto("SIP/400-0000001f", "DefaultPlan,500,1") in new stack
    -- Goto (DefaultPlan,500,1)
    -- Executing [500@DefaultPlan:1] Set("SIP/400-0000001f", "CALLERID(num)=500") in new stack
    -- Executing [500@DefaultPlan:2] Dial("SIP/400-0000001f", "SIP/450/500,20") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/450/500
[Jul  3 16:22:57] WARNING[734][C-00000018]: chan_sip.c:23875 handle_response_invite: Received response: "Forbidden" from '"400" <sip:500@192.168.1.46>;tag=as6e6a65f9'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [500@DefaultPlan:3] Busy("SIP/400-0000001f", "3") in new stack
  == Spawn extension (DefaultPlan, 500, 3) exited non-zero on 'SIP/400-0000001f'
server*CLI>

[kronos911]

Στο server A αφαίρεσε την γραμμή

exten => _5XX,1,Set(CALLERID(num)=${EXTEN})

και άλλαξε την από κάτω σε

exten => _5XX,1,Dial(SIP/450/${EXTEN},20)

[atux_null]

Στο server A αφαίρεσε την γραμμή

exten => _5XX,1,Set(CALLERID(num)=${EXTEN})

και άλλαξε την από κάτω σε

exten => _5XX,1,Dial(SIP/450/${EXTEN},20)

Το έχω κάνει παραπάνω και είναι τα logs που ανέβασα.

[kronos911]

Στο server A άλλαξε την γραμμή

exten => _5XX,1,Dial(SIP/450/${EXTEN},20)

σε

exten => _5XX,1,Dial(SIP/450,20)

Λόγου ότι το 450 είναι απλό extension στο server A και όχι trunk, δεν μπορεί να γίνει πρόσβαση στα νούμερα του server B.

Άρα αν θέλεις αμφίδρομη επικοινωνία έχεις δυο επιλογές:
Α) φτιάχνεις ένα εσωτερικό στον server B που θα κάνει register ο server A.
Β) το μετατρέπεις σε sip trunk.


Δες Asterisk tips SIP URI Dial

[atux_null]

οπότε δεν βλέπω λύση οι asterisk clients να μπορούν να μιλήσουν μεταξύ τους!

[kronos911]

Λοιπόν scrach ότι έγραψα στο 27

Στον server A πρόσθεσε τα παρακάτω στο [DefaultPlan]

exten => 450,n,Dial(SIP/450,20) 
exten => 450,n,Busy(3)

Στον server B βάλε

/450

στο τέλος του registration string
Ανέβασε log από Α και Β όταν καλείς 450 από τον Α.

[atux_null]

Το 400 καλεί το 450. Server logs:

server*CLI>
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [450@AllCalls:1] Goto("SIP/400-00000000", "DefaultPlan,450,1") in new stack
    -- Goto (DefaultPlan,450,1)
    -- Executing [450@DefaultPlan:1] Answer("SIP/400-00000000", "") in new stack
    -- Executing [450@DefaultPlan:2] VoiceMail("SIP/400-00000000", "450@Office,u") in new stack
[Jul  4 14:24:52] WARNING[2163][C-00000000]: app_voicemail.c:6549 leave_voicemail: No entry in voicemail config file for '450'
    -- Executing [450@DefaultPlan:3] Hangup("SIP/400-00000000", "") in new stack
  == Spawn extension (DefaultPlan, 450, 3) exited non-zero on 'SIP/400-00000000'
server*CLI>