1 | изначальная версия редактировать | |
Изменил, но всё равно ничего не изменилось. Включил дебаг по вашему совету, не могу понять, чего ему не хватает:
1 <cep 000000=""> : Call Received 2 <time 0=""> : SIP_TREGISTER timer timeout. 3 <sip 0=""> : localPeer->registerStat(0) 4 <sip 0=""> : ExistInRegList(localpeer) : TRUE 5 <sip 0=""> : localPeer->registerStat(0) 6 <sip 0=""> : ExistInRegList(localpeer) : TRUE 7 <cep 000000=""> : Disconnected(16) at Busy 8 <cep 000000=""> : Call Received 9 <cep 000000=""> : Disconnected(16) at Busy 10 <cep 000000=""> : Call Received 11 <cep 000000=""> : Disconnected(16) at Busy 12 <cep 000000=""> : Call Received 13 <cep 000000=""> : Disconnected(16) at Busy 14 <cep 000000=""> : Call Received 15 <time 0=""> : SIP_TREGISTER timer timeout. 16 <sip 0=""> : localPeer->registerStat(0) 17 <sip 0=""> : ExistInRegList(localpeer) : TRUE 18 <sip 0=""> : localPeer->registerStat(0) 19 <sip 0=""> : ExistInRegList(localpeer) : TRUE 20 <cep 000000=""> : Disconnected(16) at Busy 21 <cep 000000=""> : Call Received 22 <cep 000000=""> : Disconnected(16) at Busy 23 <cep 000000=""> : Call Received
2 | No.2 Revision редактировать |
Изменил, но всё равно ничего не изменилось. Включил дебаг по вашему совету, не могу понять, чего ему не хватает:
1 <cep 000000=""> : Call Received 2 <time 0=""> : SIP_TREGISTER timer timeout. 3 <sip 0=""> : localPeer->registerStat(0) 4 <sip 0=""> : ExistInRegList(localpeer) : TRUE 5 <sip 0=""> : localPeer->registerStat(0) 6 <sip 0=""> : ExistInRegList(localpeer) : TRUE 7 <cep 000000=""> : Disconnected(16) at Busy 8 <cep 000000=""> : Call Received 9 <cep 000000=""> : Disconnected(16) at Busy 10 <cep 000000=""> : Call Received 11 <cep 000000=""> : Disconnected(16) at Busy 12 <cep 000000=""> : Call Received 13 <cep 000000=""> : Disconnected(16) at Busy 14 <cep 000000=""> : Call Received 15 <time 0=""> : SIP_TREGISTER timer timeout. 16 <sip 0=""> : localPeer->registerStat(0) 17 <sip 0=""> : ExistInRegList(localpeer) : TRUE 18 <sip 0=""> : localPeer->registerStat(0) 19 <sip 0=""> : ExistInRegList(localpeer) : TRUE 20 <cep 000000=""> : Disconnected(16) at Busy 21 <cep 000000=""> : Call Received 22 <cep 000000=""> : Disconnected(16) at Busy 23 <cep 000000=""> : Call Received
А вот что говорит debug voip sip
Received SIP PDU from ( 10.5.5.253:5060 ) OPTIONS sip:10.5.5.252 SIP/2.0 Via: SIP/2.0/UDP 10.5.5.253:5060;branch=z9hG4bK0a12e07e;rport Max-Forwards: 70 From: "Unknown" <sip:unknown@10.5.5.253>;tag=as76083eff To: <sip:10.5.5.252> Contact: <sip:unknown@10.5.5.253:5060> Call-ID: 6045d9c174ca808e77227d861da4b442@10.5.5.253:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.11.0 Date: Wed, 18 Apr 2012 10:23:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 Sending SIP PDU to ( 10.5.5.253:5060 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.5.253:5060;branch=z9hG4bK0a12e07e;rport From: "Unknown" <sip:unknown@10.5.5.253>;tag=as76083eff To: <sip:10.5.5.252> Call-ID: 6045d9c174ca808e77227d861da4b442@10.5.5.253:5060 CSeq: 102 OPTIONS User-Agent: AddPac SIP Gateway Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY Content-Length: 0
Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании
Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией
GNU GPL.