Кажется проблема в этом
10:19:15.085 1 ROUTER SYSTEM: '192.168.1.230' rejected. Error Code=unknown user account
//
// We ask all users in our domain to authenticate
// If a user is in our domain, but the E-mail is listed
// in the "ExternalGateways" group, we treat it as
// a call from outside our domain, and we do not ask
// for authentication
//
fromWhom = RouteLocalURI(RemoteURI());
if fromWhom != null and then
EmailDomainPart(fromWhom) == MyDomain() and then
RemoteAuthentication() == null and then
Find(ReadGroupMembers("ExternalGateways"),EmailUserPart(fromWhom)) < 0
then
//rejectCall(401); stop;
end if;
-----Original Message-----
From: CommuniGate Pro Russian Discussions [mailto:CGatePro@mx.ru]
Sent: Tuesday, October 28, 2008 11:33 AM
To: CommuniGate Pro Russian Discussions
Subject: [CGP] Не могу победить SIP
Добрый день.
Есть такое устройство:
#sh ver
Cisco IOS Software, 2800 Software (C2800NM-ADVIPSERVICESK9-M), Version
12.4(3g), RELEASE SOFTWARE (fc2)
[...]
Cisco 2811 (revision 53.51) with 243712K/18432K bytes of memory.
Processor board ID FHK1119F187
2 FastEthernet interfaces
4 Serial(sync/async) interfaces
8 Low-speed serial(sync/async) interfaces
2 Channelized (E1 or T1)/PRI ports
2 Virtual Private Network (VPN) Modules
4 Voice FXO interfaces
DRAM configuration is 64 bits wide with parity enabled.
239K bytes of non-volatile configuration memory.
62720K bytes of ATA CompactFlash (Read/Write)
Конфигурация:
!
voice-card 0
no dspfarm
!
voice-card 1
no dspfarm
! ! !
allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g726r32
codec preference 4 g726r24
codec preference 5 g726r16
codec preference 6 g723r53
codec preference 7 g723ar53
codec preference 8 g723ar63
codec preference 9 g711alaw
codec preference 10 g711ulaw
!
!
voice-port 0/1/0
connection plar 200
!
!
dial-peer voice 1 pots
destination-pattern 7495.......
port 0/1/0
!
dial-peer voice 2 voip
destination-pattern 200
session protocol sipv2
session target sip-server
session transport udp
!
gateway
timer receive-rtp 1200
!
sip-ua
retry invite 10
retry response 5
retry cancel 5
timers trying 1000
timers connect 1000
timers disconnect 1000
sip-server dns:mail.mydomen.ru
no suspend-resume
!
CGP с лицензией на *.mail.mydomen.ru установлен на 192.168.0.251 и 192.168.0.252. Вот, как это выглядит в сети:
$ host mail.mydomen.ru
mail.sirena-travel.ru has address 192.168.0.251
mail.mydomain.ru has address 192.168.0.252
$ host mx1.mail.mydomain.ru
mx1.mail.mydomain.ru has address 192.168.0.251
$ host mx2.mail.mydomain.ru
mx2.mail.mydomain.ru has address 192.168.0.252
Все адреса внутренние, NATа пока нет. К voice-port 0/1/0 на cisco-2811 подключена обычная городская телефонная линия.
А вот, что получается при попытке позвонить на эту линию:
10:19:15.059 2 SIPDATA-002015 inp: req [0.0.0.0]:5060 <- udp[10.1.8.230]:58829 INVITE(952 bytes) sip:200@mail.mydomain.ru:5060 10:19:15.059 4 SIPDATA-002015 Hash=916534957 10:19:15.059 4 SIPS-000192 enqueued 10:19:15.059 2 SIPDATA-002015 created SIPS-000192 10:19:15.059 2 SIPS-000192 [002015] INVITE sip:200@mail.mydomain.ru:5060from udp[192.168.1.230]:58829
10:19:15.059 4 SIGNAL-002734 enqueued 10:19:15.059 2 SIPS-000192 created SIGNAL-002734 10:19:15.059 2 SIGNAL-002734 SIPS-000192: INVITEsip:200@mail.mydomain.ru:5060
10:19:15.059 4 SIGNAL-002734 AOR added: sip:200@mail.mydomain.ru:5060 10:19:15.059 4 SIGNAL-002734 applying server rules 10:19:15.059 2 SIGNAL-002734 INVITE sip:200@mail.mydomain.ru:5060 viasip:200@mail.mydomain.ru:5060
10:19:15.084 2 DIALOG-000127 created(authInp) 10:19:15.084 2 DIALOG-000127 callee set: pbx@mx1.mail.mydomain.ru 10:19:15.084 4 SIGNAL-002734 applying Account rules 10:19:15.084 4 SIGNALRULE-002734 rule(PBX Center starter) conditions met 10:19:15.084 2 SIGNALRULE-002734 rule(PBX Center starter): redirected 10:19:15.084 4 SIGNAL-002734 AOR added: sip:pbx#pbx@mx1.mail.mydomain.ru 10:19:15.084 2 SIGNAL-002734 redirected by Rules 10:19:15.084 2 SIGNAL-002734 INVITE sip:pbx#pbx@mx1.mail.mydomain.ru viasip:pbx#pbx@mx1.mail.mydomain.ru
10:19:15.084 4 PBXLEG-001816 enqueued
10:19:15.084 2 PBXLEG-001816 'pbx' created for pbx@mx1.mail.mydomain.ru
10:19:15.085 4 PBXLEG-001816 INVITE request received
10:19:15.085 2 SIGNAL-002734 {2} sent to NODE-001816: INVITE
sip:pbx#pbx@mx1.mail.mydomain.ru
10:19:15.085 2 DIALOG-000128 created as DIALOG-000127 copy(copy) 10:19:15.085 2 DIALOG-000128 enqueued 10:19:15.085 2 PBXLEG-001816 DIALOG-000128(inp) started with192.168.1.230(sip:192.168.1.230:5060)(canUpdate)(canTransfer)
10:19:15.085 2 PBXLEG-001816 session refresh=300(active) 10:19:15.085 4 PBXLEG-001816 remote SDP set: peer 10:19:15.085 4 PBXLEG-001816 signalling completed(init) 10:19:15.085 2 PBXLEG-001816 started(Main) 10:19:15.085 1 ROUTER SYSTEM: '192.168.1.230' rejected. Error Code=unknownuser account
10:19:15.085 4 PBXLEG-001816 MEDIA-000064 released 10:19:15.085 4 MEDIA-000064 closing 10:19:15.085 2 MEDIA-000064 processor started 10:19:15.187 2 MEDIA-000064 processor finished 10:19:15.289 2 MEDIA-000064 released 10:19:15.289 4 PBXLEG-001816 session timer refreshed 10:19:15.289 2 PBXLEG-001816 killing DIALOG-000128 10:19:15.289 2 SIGNAL-002734 488 received from NODE-001816 10:19:15.289 2 DIALOG-000128 dequeued(kill) 10:19:15.289 4 SIGNAL-002734 collected code=488 10:19:15.289 2 DIALOG-000128 released 10:19:15.289 2 SIGNAL-002734 updating DIALOG-000127 on 488-response 10:19:15.289 2 PBXLEG-001816 program stopped 10:19:15.289 4 PBXLEG-001816 leg finishing 10:19:15.289 4 PBXLEG-001816 dequeued 10:19:15.289 4 PBXLEG-001816 closing 10:19:15.290 2 ACCOUNT(pbx) inp call failed. Error Code=No supported audiocodec found
10:19:15.327 4 SIPDATA-002018 Hash=916534957 10:19:15.327 2 SIPDATA-002018 sent to SIPS-000192 10:19:15.327 2 SIPS-000192 [002018] confirmed: ACK received 10:19:20.003 4 SIPS-000192 I-timer over 10:19:20.003 4 SIPS-000192 dequeued 10:19:20.003 2 DIALOG-000127 released
Перебор параметра "codec" для "dial-peer voice 2 voip" ни к чему не привел.
Подскажите, пожалуйста, как заставить CISCO передавать правильно звонок, а CGP его принимать на PBX?
-- С уважением, Алексей Подмарев ################################################################## Вы получили это сообщение потому, что подписаны на список рассылки <CGatePro@mx.ru>. Чтобы отписаться, отправьте сообщение на адрес <CGatePro-off@mx.ru> Чтобы переключиться в режим дайджеста - mailto:<CGatePro-digest@mx.ru> Чтобы переключиться в индексный режим - mailto:<CGatePro-index@mx.ru> Для административных запросов адрес <CGatePro-request@mx.ru> Архив списка: http://mx.demos.su/lists/cgp-russian/Получено Tue Oct 28 08:49:56 2008
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