call manager и asterisk

[drap_hap]

настроил транк между станциями по этому мануалу
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Cisco CUCM Side:

1.) First and foremost go to System à Security à Sip Trunk Security Profile
2.) Click on Non Secure Sip Trunk Profile
3.) Click Copy
4.) In the Name Text Box add, “ – AsteriskServerName
5.) Make sure the Outgoing Transport type is UDP



6.) Should look like the above, afterwards click Save
7.) Go to Device à Trunk
8.) Click Add New
9.) Under Device Information enter a device name, I name mine by Office_Asterisk
10.) Device Pool should be set to Default
11.) Check the Media Termination Point Required Box
12.) Go to the Sip information section and under destination address enter the IP of the Asterisk Server
13.) MTP Preferred Originating Codec set to 711ulaw
14.) Sip Trunk Security Profile should be set to the profile we created above
15.) Sip Profile set as Standard Sip Profile
16.) DTMF Signaling Method should be RFC2833
17.) Click Save
18.) Go to Call Routing à Route/Hunt à Route Pattern
19.) Click Add New
20.) In Route Pattern set the dial pattern, for example if you want to send all local calls based on seven digit dials put XXXXXXX, this makes it so any seven digit dials get past to the Asterisk server
21.) Gateway/Route List should be set to the trunk we created above.
22.) Save it

We are pretty much done on the Cisco side, we have the Trunk, and the route pattern, so now the CUCM Server will send any seven digit dials to the asterisk server.

Now Depending on the Asterisk server build you have, some have the configuration will change so on the asterisk side I will just put whats needed.

1.) Create a new trunk, I called mine Sip/CUCM8
2.) Outgoing settings should be set as such
Trunk Name CUCM8
allow=ulaw
canreinvite=yes
context=from-internal
disallow=all
host=(IP OF CUCM Server)
insecure=very
nat=yes (IF YOUR ASTERISK IS BEHIND A NAT)
type=peer
qualify=yes
3.) Incoming settings
User Context = IP OF CUCM
context=from-internal
host=(IP OF ASTERISK SERVER)
type=user

The trunk is there, so what you will want to do now is test a call on CUCM with the XXXXXXX dial pattern, after that first call the trunk will be established. On the asterisk side you need to have a dial plan for the XXXXXXX, in my case I send all traffic to the Zap/g0 which sends all calls to the Pots line.
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с call manager на asterisk звонки проходят без каких либо проблем, а наоборот нет. Перепробовал кучу настроек, но результат один и тот же.
Вот логи при звонке с asterisk
-- Executing [504@from-internal:1] Macro("SIP/2001-0000016d", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/2001-0000016d", "AMPUSER=2001") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/2001-0000016d", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/2001-0000016d", "1?Set(REALCALLERIDNUM=2001)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/2001-0000016d", "AMPUSER=2001") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/2001-0000016d", "AMPUSERCIDNAME=Victor Frolov") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/2001-0000016d", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/2001-0000016d", "AMPUSERCID=2001") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/2001-0000016d", "CALLERID(all)="Victor Frolov" <2001>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/2001-0000016d", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/2001-0000016d", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] Set("SIP/2001-0000016d", "CALLERID(number)=2001") in new stack
-- Executing [s@macro-user-callerid:20] Set("SIP/2001-0000016d", "CALLERID(name)=Victor Frolov") in new stack
-- Executing [s@macro-user-callerid:21] NoOp("SIP/2001-0000016d", "Using CallerID "Victor Frolov" <2001>") in new stack
-- Executing [504@from-internal:2] NoOp("SIP/2001-0000016d", "Calling Out Route: tocucm") in new stack
-- Executing [504@from-internal:3] Set("SIP/2001-0000016d", "MOHCLASS=default") in new stack
-- Executing [504@from-internal:4] Set("SIP/2001-0000016d", "_NODEST=") in new stack
-- Executing [504@from-internal:5] Macro("SIP/2001-0000016d", "record-enable,2001,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/2001-0000016d", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/2001-0000016d", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/2001-0000016d", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/2001-0000016d", "0?IN") in new stack
-- Executing [s@macro-record-enable:16] ExecIf("SIP/2001-0000016d", "1?MacroExit()") in new stack
-- Executing [504@from-internal:6] Macro("SIP/2001-0000016d", "dialout-trunk,1,504,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/2001-0000016d", "DIAL_TRUNK=1") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/2001-0000016d", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/2001-0000016d", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/2001-0000016d", "DIAL_NUMBER=504") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/2001-0000016d", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/2001-0000016d", "OUTBOUND_GROUP=OUT_1") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/2001-0000016d", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/2001-0000016d", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/2001-0000016d", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/2001-0000016d", "outbound-callerid,1") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/2001-0000016d", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/2001-0000016d", "0?Set(REALCALLERIDNUM=2001)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/2001-0000016d", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/2001-0000016d", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/2001-0000016d", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/2001-0000016d", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/2001-0000016d", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/2001-0000016d", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/2001-0000016d", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/2001-0000016d", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/2001-0000016d", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/2001-0000016d", "0?sub-flp-1,s,1") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/2001-0000016d", "OUTNUM=504") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/2001-0000016d", "custom=SIP/CUCM8") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/2001-0000016d", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/2001-0000016d", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/2001-0000016d", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/2001-0000016d", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/2001-0000016d", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/2001-0000016d", "SIP/CUCM8/504,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/CUCM8/504
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] NoOp("SIP/2001-0000016d", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/2001-0000016d", "s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/2001-0000016d", "RC=34") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/2001-0000016d", "34,1") in new stack
-- Goto (macro-dialout-trunk,34,1)
-- Executing [34@macro-dialout-trunk:1] Goto("SIP/2001-0000016d", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/2001-0000016d", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/2001-0000016d", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/2001-0000016d", "CALLERID(number)=2001") in new stack
-- Executing [504@from-internal:7] Macro("SIP/2001-0000016d", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/2001-0000016d", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/2001-0000016d", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/2001-0000016d", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/2001-0000016d", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/2001-0000016d> Playing 'all-circuits-busy-now.gsm' (language 'en')
-- <SIP/2001-0000016d> Playing 'pls-try-call-later.gsm' (language 'en')
-- Executing [s@macro-outisbusy:5] Congestion("SIP/2001-0000016d", "20") in new stack
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/2001-0000016d' in macro 'outisbusy'
== Spawn extension (from-internal, 504, 7) exited non-zero on 'SIP/2001-0000016d'
-- Executing [h@from-internal:1] Macro("SIP/2001-0000016d", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/2001-0000016d", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("SIP/2001-0000016d", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/2001-0000016d", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,15)
-- Executing [s@macro-hangupcall:15] NoOp("SIP/2001-0000016d", "MEETME_RECORDINGFILE=") in new stack
-- Executing [s@macro-hangupcall:16] GotoIf("SIP/2001-0000016d", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,18)
-- Executing [s@macro-hangupcall:18] NoOp("SIP/2001-0000016d", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:19] GotoIf("SIP/2001-0000016d", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,25)
-- Executing [s@macro-hangupcall:25] NoOp("SIP/2001-0000016d", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:26] GotoIf("SIP/2001-0000016d", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,29)
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/2001-0000016d", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,32)
-- Executing [s@macro-hangupcall:32] GotoIf("SIP/2001-0000016d", "1?theend") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] Hangup("SIP/2001-0000016d", "") in new stack
== Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/2001-0000016d' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/2001-0000016d'


подскажите куда копать?

[ded]

Подскажу
- не надо постить инструкции, кто их будет проверять?
- не надо постить бессмысленные портянки вида -

Вот логи при звонке с asterisk
-- Executing [504@from-internal:1] Macro("SIP/2001-0000016d", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/2001-0000016d", "AMPUSER=2001") in new stack

- а надо sip set debug peer CUCM8, смотреть код ошибки которым отбивает CUCM8
и изучать самому, постить сюда не надо, если не знаете на что обратить внимания.
Без 5 минут платный суппорт.

[drap_hap]

Спасибо за помощь, правильно подебажил, полазил по забугорным форумам и понял что проблема в настройках css в транке call managera. Всё заработало.