Исходи из принципа, что by default можно всё, и только более правильный вопрос - как?
мне надо какой-то дамми драйвер?
Цитата с главной страницы:
Asterisk needs no additional hardware for Voice over IP. For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsors, Digium™. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks as well as a single port FXO card and a one to four-port modular FXS and FXO card.
так что меня интересует, что и куда писать...
например:
1. есть пользователем с софтфоном с динамическим адресом
2. есть гейт с постоянным адресом. даже 2-3 гейта на то же направление
кодек 723 или 729....
примерно - где и что записать?[/b]
что тебе нужно пример iax.conf, sip.conf и тд вводи прям там в поиск по сайту
но я бы начал с изучения этих файлов локально - там очень хороший синтаксис и коментарии - все можно понять и без документации
что не понятно по какой команде - лезешь на voip-info.org и смотришь описание её
а файлы конфигов лежат в /etc/asterisk/
я бы на твоем месте вообще все там посмотрел/поиграл бы с ними ...
удачи
Однако не получаеться одня вещь.
У меня есть сип гейт. он работает как стандалон и не регестрируеться на *.
Звонок может с него на * пойти а может и с * на него.
Конфиги я брал стандартные.
Однако - не идет. На сип железке кодека г723.1.
Что не так?
Ниже мой конфиг и логи.
гейт -> * (войсмайл)
*CLI> -- Executing Playback("SIP/antek-b465", "transfer|skip") in new stack
-- Executing Macro("SIP/antek-b465", "stdexten|1234|Console/dsp") in new stack
-- Executing Dial("SIP/antek-b465", "Console/dsp|20") in new stack
Feb 27 16:15:36 WARNING[5284]: channel.c:1879 ast_request: No translator path exists for channel type Console (native 64) to 256
Feb 27 16:15:36 NOTICE[5284]: app_dial.c:746 dial_exec: Unable to create channel of type 'Console'
== Everyone is busy/congested at this time
-- Executing Goto("SIP/antek-b465", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-stdexten,s-CHANUNAVAIL,1)
-- Executing Goto("SIP/antek-b465", "s-NOANSWER|1") in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing VoiceMail("SIP/antek-b465", "u1234") in new stack
Feb 27 16:15:36 NOTICE[5284]: channel.c:1691 ast_set_write_format: Unable to find a path from gsm to g729
Feb 27 16:15:36 WARNING[5284]: file.c:779 ast_streamfile: Unable to open voicemail/default/1234/unavail (format g729): No such file or directory
Feb 27 16:15:36 NOTICE[5284]: channel.c:1691 ast_set_write_format: Unable to find a path from gsm to g729
Feb 27 16:15:36 WARNING[5284]: file.c:779 ast_streamfile: Unable to open vm-intro (format g729): No such file or directory
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'SIP/antek-b465' in macro 'stdexten'
== Spawn extension (default, 1234, 2) exited non-zero on 'SIP/antek-b465'
гейт -> * (зарегестрированный клиент)
-- Registered SIP 'xlite-andrius' at IP-моего-компа port 62607 expires 900
-- Saved useragent "kphone/4.0.5" for peer xlite-andrius
-- Executing Playback("SIP/antek-bb99", "transfer|skip") in new stack
-- Executing Macro("SIP/antek-bb99", "stdexten|1234|Console/dsp") in new stack
-- Executing Dial("SIP/antek-bb99", "Console/dsp|20") in new stack
Feb 27 16:16:03 WARNING[5284]: channel.c:1879 ast_request: No translator path exists for channel type Console (native 64) to 256
Feb 27 16:16:03 NOTICE[5284]: app_dial.c:746 dial_exec: Unable to create channel of type 'Console'
== Everyone is busy/congested at this time
-- Executing Goto("SIP/antek-bb99", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-stdexten,s-CHANUNAVAIL,1)
-- Executing Goto("SIP/antek-bb99", "s-NOANSWER|1") in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing VoiceMail("SIP/antek-bb99", "u1234") in new stack
Feb 27 16:16:03 NOTICE[5284]: channel.c:1691 ast_set_write_format: Unable to find a path from gsm to g729
Feb 27 16:16:03 WARNING[5284]: file.c:779 ast_streamfile: Unable to open voicemail/default/1234/unavail (format g729): No such file or directory
Feb 27 16:16:03 NOTICE[5284]: channel.c:1691 ast_set_write_format: Unable to find a path from gsm to g729
Feb 27 16:16:03 WARNING[5284]: file.c:779 ast_streamfile: Unable to open vm-intro (format g729): No such file or directory
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'SIP/antek-bb99' in macro 'stdexten'
== Spawn extension (default, 1234, 2) exited non-zero on 'SIP/antek-bb99'
Звонок с сип софтфона на * и далее на гейт
-- Registered SIP 'xlite-andrius' at IP-моего-компа port 62610 expires 900
-- Saved useragent "kphone/4.0.5" for peer xlite-andrius
-- Executing Dial("SIP/xlite-andrius-a10c", "SIP/antek|номер-моего-мобильного") in newstack
-- Called antek
Feb 27 16:20:14 WARNING[5304]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/antek-1567(256) to SIP/xlite-andrius-a10c(2)
-- Got SIP response 404 "Not Found" back from IP-сип-гейта
-- SIP/antek-1567 is circuit-busy
== Everyone is busy/congested at this time
-- Timeout on SIP/xlite-andrius-a10c
== CDR updated on SIP/xlite-andrius-a10c
-- Executing Goto("SIP/xlite-andrius-a10c", "#|1") in new stack
-- Goto (default,#,1)
-- Executing Playback("SIP/xlite-andrius-a10c", "demo-thanks") in new stack
-- Playing 'demo-thanks' (language 'en')
== Spawn extension (default, #, 1) exited non-zero on 'SIP/xlite-andrius-a10c'
конфиг для обеих
sip.conf
[antek]
type=friend ; either "friend" (peer+user), "peer" or "user"
;context=from-sip
username=antek
;fromuser=ANTEK ; overrides the callerid, e.g. required by FWD
;callerid=ANTEK
host=IP-сип-гейта ; we have a static but private IP address
insecure
nat=no ; there is not NAT between phone and Asterisk
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
incominglimit=1 ; permit only 1 outgoing call at a time
; from the phone to asterisk
regexten=1235
;mailbox=1235@default ; mailbox 1234 in voicemail context "default"
disallow=all ; need to disallow=all before we can use allow=
;allow=ulaw ; Note: In user sections the order of codecs
; listed with allow= does NOT matter!
;allow=alaw
allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
allow=g729 ; Pass-thru only unless g729 license obtained
[xlite-andrius]
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
regexten=1234 ; When they register, create extension 1234
username=andrius
;callerid="Jane Smith"
host=dynamic
nat=yes ; X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
extension.conf
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
;exten => 1235,1,Voicemail(u1234) ; Right to voicemail
exten => 1235,1, Dial(SIP/antek,номер-моего-мобильного)
но все же документация разбросанна и примеров я так и не нашел.. мне помогли