Не спрашивайте что я курил или какие грибы ел, но нужно для эксперимента соединить две астериски по E1! Физически поток поднялся, но вызовы не идут - "Все линии заняты". Маршрутизация как при IAX2 не работает... Может есть у кого идеи?
Добавил!
(сервер 1, подключен span2)
pbx*CLI> pri show spans
PRI span 1/0: In Alarm, Down, Active
PRI span 2/0: Up, Active
cat /etc/asterisk/dahdi-channels.conf
; Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 15 09:30:40 2012
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global settings
;
; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1"
group=2
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63
; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group=3
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
context = default
group = 63
cat /etc/dahdi/system.conf
# Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 15 09:30:40 2012
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
span=1,1,0,ccs,hdb3
echocanceller=oslec,1-15,17-31
# termtype: te
bchan=1-15,17-31
dchan=16
# Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
span=2,2,0,ccs,hdb3
echocanceller=oslec,32-46,48-62
# termtype: te
bchan=32-46,48-62
dchan=47
# Global data
loadzone = us
defaultzone = us
(сервер 2, подключен span3)
pbx*CLI> pri show spans
PRI span 2/0: Up, Active
PRI span 3/0: Up, Active
cat /etc/asterisk/dahdi-channels.conf
; Autogenerated by /usr/sbin/dahdi_genconf on Thu Jul 26 17:10:54 2012
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global settings
;
.....................
; Span 2: TE2/0/1 "T2XXP (PCI) Card 0 Span 1"
group=2
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 13-27,29-43
context = default
group = 63
; Span 3: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group=3
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 44-58,60-74
context = default
group = 63
cat /etc/dahdi/system.conf
# Autogenerated by /usr/sbin/dahdi_genconf on Thu Jul 26 17:10:54 2012
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Span 2: TE2/0/1 "T2XXP (PCI) Card 0 Span 1"
span=2,1,0,ccs,hdb3
echocanceller=oslec,13-27,29-43
# termtype: te
bchan=13-27,29-43
dchan=28
# Span 3: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
span=3,2,0,ccs,hdb3
echocanceller=oslec,44-58,60-74
# termtype: te
bchan=44-58,60-74
dchan=59
# Global data
loadzone = us
defaultzone = us
1 Сервер
cat /etc/asterisk/chan_dahdi.conf
[trunkgroups]
[channels]
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
;Uncomment these lines if you have problems with the disconection of your analog lines
busydetect=yes
busycount=1
immediate=no
#include dahdi-channels.conf
#include chan_dahdi_additional.conf
2 сервер
cat /etc/asterisk/chan_dahdi.conf
; Auto-generated by /usr/sbin/hardware_detector
[trunkgroups]
[channels]
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
relaxdtmf=yes
;Uncomment these lines if you have problems with the disconection of your analog lines
busydetect=yes
busycount=1
immediate=no
#include dahdi-channels.conf
#include chan_dahdi_additional.conf
Connected to Asterisk 1.8.11.0 currently running on pbx (pid = 5933)
Verbosity is at least 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [4001@from-internal:1] Macro("SIP/9012-00000079", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/9012-00000079", "AMPUSER=9012") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/9012-00000079", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/9012-00000079", "1?Set(REALCALLERIDNUM=9012)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/9012-00000079", "AMPUSER=9012") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/9012-00000079", "AMPUSERCIDNAME=Trokhina E V") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/9012-00000079", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/9012-00000079", "AMPUSERCID=9012") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/9012-00000079", "CALLERID(all)="Trokhina E V" <9012>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/9012-00000079", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/9012-00000079", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] Set("SIP/9012-00000079", "CALLERID(number)=9012") in new stack
-- Executing [s@macro-user-callerid:20] Set("SIP/9012-00000079", "CALLERID(name)=Trokhina E V") in new stack
-- Executing [s@macro-user-callerid:21] NoOp("SIP/9012-00000079", "Using CallerID "Trokhina E V" <9012>") in new stack
-- Executing [4001@from-internal:2] NoOp("SIP/9012-00000079", "Calling Out Route: to_dss") in new stack
-- Executing [4001@from-internal:3] Set("SIP/9012-00000079", "MOHCLASS=default") in new stack
-- Executing [4001@from-internal:4] Set("SIP/9012-00000079", "_NODEST=") in new stack
-- Executing [4001@from-internal:5] Macro("SIP/9012-00000079", "record-enable,9012,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/9012-00000079", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/9012-00000079", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/9012-00000079", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/9012-00000079", "0?IN") in new stack
-- Executing [s@macro-record-enable:16] ExecIf("SIP/9012-00000079", "1?MacroExit()") in new stack
-- Executing [4001@from-internal:6] Macro("SIP/9012-00000079", "dialout-trunk,3,4001,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/9012-00000079", "DIAL_TRUNK=3") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/9012-00000079", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/9012-00000079", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/9012-00000079", "DIAL_NUMBER=4001") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/9012-00000079", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/9012-00000079", "OUTBOUND_GROUP=OUT_3") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/9012-00000079", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/9012-00000079", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/9012-00000079", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/9012-00000079", "outbound-callerid,3") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/9012-00000079", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/9012-00000079", "0?Set(REALCALLERIDNUM=9012)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/9012-00000079", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/9012-00000079", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/9012-00000079", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/9012-00000079", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/9012-00000079", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/9012-00000079", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/9012-00000079", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/9012-00000079", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/9012-00000079", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/9012-00000079", "0?sub-flp-3,s,1") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/9012-00000079", "OUTNUM=4001") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/9012-00000079", "custom=DAHDI/g3") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/9012-00000079", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/9012-00000079", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/9012-00000079", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/9012-00000079", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/9012-00000079", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/9012-00000079", "DAHDI/g3/4001,300,") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called DAHDI/g3/4001
-- Span 2: Channel 0/1 got hangup, cause 27
-- DAHDI/i2/4001-f is circuit-busy
-- Hungup 'DAHDI/i2/4001-f'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] NoOp("SIP/9012-00000079", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 27") in new stack
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/9012-00000079", "s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/9012-00000079", "RC=27") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/9012-00000079", "27,1") in new stack
-- Goto (macro-dialout-trunk,27,1)
-- Executing [27@macro-dialout-trunk:1] Goto("SIP/9012-00000079", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/9012-00000079", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/9012-00000079", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 27 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/9012-00000079", "CALLERID(number)=9012") in new stack
-- Executing [4001@from-internal:7] Macro("SIP/9012-00000079", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/9012-00000079", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/9012-00000079", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/9012-00000079", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/9012-00000079", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/9012-00000079> Playing 'all-circuits-busy-now.gsm' (language 'ru')
== Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/9012-00000079' in macro 'outisbusy'
== Spawn extension (from-internal, 4001, 7) exited non-zero on 'SIP/9012-00000079'
-- Executing [h@from-internal:1] Macro("SIP/9012-00000079", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/9012-00000079", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("SIP/9012-00000079", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/9012-00000079", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,15)
-- Executing [s@macro-hangupcall:15] NoOp("SIP/9012-00000079", "MEETME_RECORDINGFILE=") in new stack
-- Executing [s@macro-hangupcall:16] GotoIf("SIP/9012-00000079", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,18)
-- Executing [s@macro-hangupcall:18] NoOp("SIP/9012-00000079", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:19] GotoIf("SIP/9012-00000079", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,25)
-- Executing [s@macro-hangupcall:25] NoOp("SIP/9012-00000079", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:26] GotoIf("SIP/9012-00000079", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,29)
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/9012-00000079", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,32)
-- Executing [s@macro-hangupcall:32] GotoIf("SIP/9012-00000079", "1?theend") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] Hangup("SIP/9012-00000079", "") in new stack
== Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/9012-00000079' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/9012-00000079'
На обоих серверах стоит
signalling = pri_cpe
На одном всёж нужно поставить
signalling = pri_net
И будет счастье =))))))
На мой взгляд очевидно не настроен chan_dahdi.conf сделай так:
[trunkgroups]
[channels]
language=ru
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
group=1
;;;[E1]
context=from-trunk
switchtype=euroisdn
prilocaldialplan=local
pridialplan=national
signalling=pri_cpe
;faxdetect=incoming
channel =>1-15,17-31,32-46,48-62
Соответственно нужно понимать в какой контекст у тебя должен уходить звонок.
скорее всего вы не скомпилировали pri в dahdi либо неправильно настроили.
вообще можно на тот же провод е1 поставить два модема shdsl и по линку даже больше каналов пройдет. смысл соединять по е1?
в этой связи возникает несколько вопросов - е1 прямым проводом между астерами, или через некого провайдера ?
если через провайдера, то нужно смотреть маски по которым провайдер дает звонить.. например ограничение что с 100 по 400 номер пробиваются наружу, остальные болт ! (или отключить определитель номера в потоке)
а какой из астеров у вас главный ? просто в е1 на сколько я понимаю 1 сторона вроде как должна быть главной, а другая из серии user - просто я недавно шлюз с е1 потоками ковырял - там как раз в настройках от вышестоящего астера я выставлял в настройках порта q.931user а в передающем порту на другую атс q931network
romirin (Aug 23 '12)editЗадан: Aug 17 '12
Просмотрен: 1,164 раз
Обновлен: Aug 23 '12
Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании
Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией
GNU GPL.
pri show spans
zzuz (Aug 17 '12)editкак всегда - конфиги и логи в студию..
Zavr2008 (Aug 17 '12)editи pri intense debug span 1. стоит показать SETUP звонки и DISCONNECT.
Zavr2008 (Aug 17 '12)editЧего там показывать если у автора логика не поднята?
zzuz (Aug 17 '12)editА кто chan_dahdi.conf будет рисовать?
zzuz (Aug 22 '12)editДобавил конфиги
zavulon (Aug 22 '12)editи как звоните на E1 ?
zzuz (Aug 22 '12)editвсмысле как звоню? Вам маршрутизация нужна, или лог? У меня эластикс, маршруты могу скринами предоставить. Лог звонка тоже выложу.
zavulon (Aug 22 '12)editлог звонка. Вообще по мне очень странно описаны каналы, на E1 не похоже. Вы джампер на плате переключили с T1 на E1 ?
zzuz (Aug 22 '12)editОбе платы с каждого из серверов тестились на существующем канале Е1 от провайдера, все работает. (сервер 2, span2 как раз на потоке Е1 от прова принимает линию) Добавил лог звонка!
zavulon (Aug 23 '12)editDAHDI/g3/ - где у Вас в конфиге определены группы?
zzuz (Aug 23 '12)editВсе заработало!!!
zavulon (Aug 23 '12)editнеужели всегда так нужно вытягивать информацию.
zzuz (Aug 23 '12)edit