kanim Posted March 24, 2011 Posted March 24, 2011 Нужна помощь. Есть железка Addpac IP200 работает как SIP телефон, подключен к SIP серверу в качестве которого выступает SIP плата на АТС Меридиан. Так вот, проблема в том, что при исходящем звонке с Addpac IP200 на любой SIP номер через 20-30 секунд разговора рвется соединение. Входящие звонки на Addpac IP200 проходят нормально. Если вместо Addpac IP200 поставить копм с софтовым клиентом Xlite то все звонки проходят без проблем и входящие и исходящие. Из дебага видно что сессия рвется по ошибке 504. Вот конфиг Addpac IP200(версия :ip200_g2_v8_41_028) hostname IP200 clock timezone EET 2 ! username хххх password ххххх administrator ! ! interface Loopback0 ip address 127.0.0.1 255.0.0.0 ! interface FastEthernet0/0 ip address dhcp speed auto ! interface FastEthernet0/1 no ip address speed auto ! ! ip route 0.0.0.0 0.0.0.0 x.y.64.1 via dhcp ! ! ! snmp name IP200_G2 snmp enable-cputrap ! ip udp blackhole http server ! ! ! IP PHONE OSD configuration. ! osd language english network signaling sip network sscp disable network lan-dhcp none phone lcd-type graphic phone ring-type 1 phone volume ring 2 phone volume input 7 phone volume output 5 phone volume micbooster disable phone auto-hook-on disable phone display-name Odessa phone voice-codec 3 phone dnd-mode silence ! ! SSCP configuration.! ! ! ! SSCP Static CM List sscp ! ! SSCP Dynamic CM List sscp ! ! sscp call-manager broadcast port 8855 logger disable logger level info ! ! ! ! ! VoIP configuration. ! ! ! Voice service voip configuration. ! voice service voip fax protocol t38 redundancy 0 fax rate disable h323 call start fast h323 call tunnel enable announcement language english ! ! ! Voice port configuration. ! ! SPEECH voice-port 0/0 ! ! ! FXO voice-port 0/1 ! ! ! ! ! Pots peer configuration. ! dial-peer voice 0 pots destination-pattern 6010 port 0/0 ! ! ! ! Voip peer configuration. ! dial-peer voice 1001 voip destination-pattern T session target sip-server session protocol sip voice-class codec 0 dtmf-relay rtp-2833 huntstop ! dial-peer voice 1002 voip destination-pattern T session target ras voice-class codec 0 dtmf-relay h245-alphanumeric preference 1 huntstop ! ! ! ! ! ! ! Gateway configuration. ! gateway h323-id voip.x.y.64.210 ! ! ! Codec classes configuration. ! voice class codec 0 codec preference 1 g729 codec preference 2 g7231r63 codec preference 3 g711alaw codec preference 4 g711ulaw ! ! ! ! SIP UA configuration. ! sip-ua user-register sip-username 6010 sip-password xxxxxxxx sip-server x.y.69.1 srv enable rport enable media-channel early register e164 ! ! ! MGCP configuration. ! mgcp codec g729 vad ! ! ! Tones ! ! ! ! line console ! line vty ! ! ! APOS File System ! ! mount mem 512 /tmp ! ! sms quata 30 ! Буду рад любой помощи в этом вопросе.
Kristoff_Vampire Posted March 24, 2011 Posted March 24, 2011 Стандартная лажа Addpac ---- уберите VAD.
kanim Posted March 28, 2011 Author Posted March 28, 2011 Стандартная лажа Addpac ---- уберите VAD. Кроме VAD еще может, что-то мешать? Я уже пробовал отключать VAD и результата это не дало.
Aleck_K Posted March 28, 2011 Posted March 28, 2011 Кроме VAD еще может, что-то мешать? Я уже пробовал отключать VAD и результата это не дало. Так бывает, когда АСК некорректно долетает. Покажите sip flow
edwin Posted March 28, 2011 Posted March 28, 2011 Очень похоже на то, что Addpac или не шлет ACK на SIP OK сервера, или шлет в некорректном виде ... надо бы дамп обмена глянуть.
kanim Posted April 1, 2011 Author Posted April 1, 2011 Наконец появилась возможность снять дебаг Вот, что он отвечает на debug voip sip IP200# Sending SIP PDU to ( 10.100.69.1:5060 ) from 5060 INVITE sip:6012@10.100.69.1 SIP/2.0 Via: SIP/2.0/UDP 10.115.64.98:5060;branch=z9hG4bK224d7919a425 From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1> Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 25 INVITE Supported: replaces, timer, 100rel, early-session Min-SE: 1800 Date: Fri, 01 Apr 2011 13:12:02 GMT Session-Expires: 1800 User-Agent: AddPac SIP Gateway Contact: <sip:6010@10.115.64.98> Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO Allow-Events: talk, hold, conference Content-Type: application/sdp Content-Length: 305 Max-Forwards: 70 v=0 o=6010 1301663522 1301663522 IN IP4 10.115.64.98 s=AddPac Gateway SDP c=IN IP4 10.115.64.98 t=1301663522 0 m=audio 23002 RTP/AVP 18 4 8 0 101 a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Received SIP PDU from ( 10.100.69.1:5060 ) SIP/2.0 100 Trying From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1> Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 25 INVITE Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a425 Content-Length: 0 Organization: 10.100.69.1 Server: Avaya SIP Enablement Services Received SIP PDU from ( 10.100.69.1:5060 ) SIP/2.0 407 Proxy Authentication Required From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=471CE0B3F3F312BA810FD40EEAC3794C1270120036123744 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 25 INVITE Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a425 Content-Length: 0 Proxy-Authenticate: Digest realm="10.100.69.1",domain="10.100.69.1",nonce="MTI3MDEyMDAzNjpTREZTZXJ2ZXJTZWNyZXRLZXk6MjA2Nzg0NTQ0Ng==",algorithm=MD5 Server: Avaya SIP Enablement Services Organization: 10.100.69.1 Sending SIP PDU to ( 10.100.69.1:5060 ) from 5060 ACK sip:6012@10.100.69.1 SIP/2.0 Via: SIP/2.0/UDP 10.115.64.98:5060;branch=z9hG4bK224d7919a425 From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=471CE0B3F3F312BA810FD40EEAC3794C1270120036123744 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 25 ACK Content-Length: 0 Max-Forwards: 70 Sending SIP PDU to ( 10.100.69.1:5060 ) from 5060 INVITE sip:6012@10.100.69.1 SIP/2.0 Via: SIP/2.0/UDP 10.115.64.98:5060;branch=z9hG4bK224d7919a426 From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1> Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 26 INVITE Supported: replaces, timer, 100rel, early-session Min-SE: 1800 Date: Fri, 01 Apr 2011 13:12:02 GMT Session-Expires: 1800 User-Agent: AddPac SIP Gateway Contact: <sip:6010@10.115.64.98> Accept: application/sdp Proxy-Authorization: Digest username="6010", realm="10.100.69.1", nonce="MTI3MDEyMDAzNjpTREZTZXJ2ZXJTZWNyZXRLZXk6MjA2Nzg0NTQ0Ng==", ri="sip:6012@10.100.69.1", response="e95608718af7e87c5d012a852a23e5d6", algorithm=MD5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO Allow-Events: talk, hold, conference Content-Type: application/sdp Content-Length: 315 Max-Forwards: 70 v=0 o=6010 1301663522 1301663522 IN IP4 10.115.64.98 s=AddPac Gateway SDP c=IN IP4 10.115.64.98 t=1301663522 0 m=audio 23002 RTP/AVP 18 4 8 0 101 a=ptime:20 a=rtpmap:18 G729/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 Received SIP PDU from ( 10.100.69.1:5060 ) SIP/2.0 100 Trying From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1> Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 26 INVITE Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a426 Content-Length: 0 Organization: 10.100.69.1 Server: Avaya SIP Enablement Services Received SIP PDU from ( 10.100.69.1:5060 ) SIP/2.0 422 Session Interval Too Small From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=471CE0B3F3F312BA810FD40EEAC3794C1270120037123747 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 26 INVITE Via: SIP/2.0/UDP 10.115.64.98:5060;psrrposn=1;received=10.115.64.98;branch=z9hG4bK224d7919a426 Server: Avaya CM/R015x.02.1.016.4 Min-SE: 3600 Content-Length: 0 Sending SIP PDU to ( 10.100.69.1:5060 ) from 5060 ACK sip:6012@10.100.69.1 SIP/2.0 Via: SIP/2.0/UDP 10.115.64.98:5060;branch=z9hG4bK224d7919a426 From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=471CE0B3F3F312BA810FD40EEAC3794C1270120037123747 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 26 ACK Proxy-Authorization: Digest username="6010", realm="10.100.69.1", nonce="MTI3MDEyMDAzNjpTREZTZXJ2ZXJTZWNyZXRLZXk6MjA2Nzg0NTQ0Ng==", ri="sip:6012@10.100.69.1", response="7991b5da1f3102c5f03a9a71c41aab3e", algorithm=MD5 Content-Length: 0 Max-Forwards: 70 Sending SIP PDU to ( 10.100.69.1:5060 ) from 5060 INVITE sip:6012@10.100.69.1 SIP/2.0 Via: SIP/2.0/UDP 10.115.64.98:5060;branch=z9hG4bK224d7919a427 From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1> Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 27 INVITE Supported: replaces, timer, 100rel, early-session Min-SE: 3600 Date: Fri, 01 Apr 2011 13:12:02 GMT Session-Expires: 3600 User-Agent: AddPac SIP Gateway Contact: <sip:6010@10.115.64.98> Accept: application/sdp Proxy-Authorization: Digest username="6010", realm="10.100.69.1", nonce="MTI3MDEyMDAzNjpTREZTZXJ2ZXJTZWNyZXRLZXk6MjA2Nzg0NTQ0Ng==", uri="sip:6012@10.100.69.1", response="e95608718af7e87c5d012a852a23e5d6", algorithm=MD5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO Allow-Events: talk, hold, conference Content-Type: application/sdp Content-Length: 315 Max-Forwards: 70 v=0 o=6010 1301663522 1301663522 IN IP4 10.115.64.98 s=AddPac Gateway SDP c=IN IP4 10.115.64.98 t=1301663522 0 m=audio 23002 RTP/AVP 18 4 8 0 101 a=ptime:20 a=rtpmap:18 G729/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 Received SIP PDU from ( 10.100.69.1:5060 ) SIP/2.0 100 Trying From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1> Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 27 INVITE Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427 Content-Length: 0 Organization: 10.100.69.1 Server: Avaya SIP Enablement Services Received SIP PDU from ( 10.100.69.1:5060 ) SIP/2.0 180 Ringing From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 27 INVITE Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427 Record-Route: <sip:10.100.69.1:6001;lr;transport=tls> Record-Route: <sip:10.100.69.1:5060;lr> Server: Avaya CM/R015x.02.1.016.4 Contact: "Polycom" <sip:6012@10.100.69.1:6001;user=phone;transport=tls> P-Asserted-Identity: "Polycom" <sip:6012@10.100.69.1:6001;user=phone> Supported: timer,replaces,join,histinfo,100rel Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH RSeq: 1 Require: 100rel Content-Type: application/sdp Content-Length: 184 v=0 o=- 1 2 IN IP4 10.100.69.1 s=- c=IN IP4 10.100.69.2 b=AS:64 t=0 0 m=audio 2050 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 Sending SIP PDU to ( 10.100.69.1:5060 ) from 5060 PRACK sip:6012@10.100.69.1:6001;transport=tls;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.115.64.98:5060;branch=z9hG4bK224d7919a428 From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 28 PRACK Route: <sip:10.100.69.1:5060;lr>,<sip:10.100.69.1:6001;lr;transport=tls> Date: Fri, 01 Apr 2011 13:12:03 GMT User-Agent: AddPac SIP Gateway RAck: 1 27 INVITE Contact: <sip:6010@10.115.64.98> Content-Length: 0 Max-Forwards: 70 Received SIP PDU from ( 10.100.69.1:5060 ) SIP/2.0 407 Proxy Authentication Required From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 28 PRACK Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a428 Content-Length: 0 Proxy-Authenticate: Digest realm="10.100.69.1",domain="10.100.69.1",nonce="MTI3MDEyMDAzNzpTREZTZXJ2ZXJTZWNyZXRLZXk6Mzk3MTA1NjA0",algorithm=MD5 Server: Avaya SIP Enablement Services Organization: 10.100.69.1 Received SIP PDU from ( 10.100.69.1:5060 ) SIP/2.0 180 Ringing From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 27 INVITE Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427 Record-Route: <sip:10.100.69.1:6001;lr;transport=tls> Record-Route: <sip:10.100.69.1:5060;lr> Server: Avaya CM/R015x.02.1.016.4 Contact: "Polycom" <sip:6012@10.100.69.1:6001;user=phone;transport=tls> P-Asserted-Identity: "Polycom" <sip:6012@10.100.69.1:6001;user=phone> Supported: timer,replaces,join,histinfo,100rel Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH RSeq: 1 Require: 100rel Content-Type: application/sdp Content-Length: 184 v=0 o=- 1 2 IN IP4 10.100.69.1 s=- c=IN IP4 10.100.69.2 b=AS:64 t=0 0 m=audio 2050 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 Received SIP PDU from ( 10.100.69.1:5060 ) SIP/2.0 180 Ringing From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 27 INVITE Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427 Record-Route: <sip:10.100.69.1:6001;lr;transport=tls> Record-Route: <sip:10.100.69.1:5060;lr> Server: Avaya CM/R015x.02.1.016.4 Contact: "Polycom" <sip:6012@10.100.69.1:6001;user=phone;transport=tls> P-Asserted-Identity: "Polycom" <sip:6012@10.100.69.1:6001;user=phone> Supported: timer,replaces,join,histinfo,100rel Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH RSeq: 1 Require: 100rel Content-Type: application/sdp Content-Length: 184 v=0 o=- 1 2 IN IP4 10.100.69.1 s=- c=IN IP4 10.100.69.2 b=AS:64 t=0 0 m=audio 2050 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 Received SIP PDU from ( 10.100.69.1:5060 ) SIP/2.0 180 Ringing From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 27 INVITE Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427 Record-Route: <sip:10.100.69.1:6001;lr;transport=tls> Record-Route: <sip:10.100.69.1:5060;lr> Server: Avaya CM/R015x.02.1.016.4 Contact: "Polycom" <sip:6012@10.100.69.1:6001;user=phone;transport=tls> P-Asserted-Identity: "Polycom" <sip:6012@10.100.69.1:6001;user=phone> Supported: timer,replaces,join,histinfo,100rel Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH RSeq: 1 Require: 100rel Content-Type: application/sdp Content-Length: 184 v=0 o=- 1 2 IN IP4 10.100.69.1 s=- c=IN IP4 10.100.69.2 b=AS:64 t=0 0 m=audio 2050 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 Received SIP PDU from ( 10.100.69.1:5060 ) SIP/2.0 180 Ringing From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 27 INVITE Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427 Record-Route: <sip:10.100.69.1:6001;lr;transport=tls> Record-Route: <sip:10.100.69.1:5060;lr> Server: Avaya CM/R015x.02.1.016.4 Contact: "Polycom" <sip:6012@10.100.69.1:6001;user=phone;transport=tls> P-Asserted-Identity: "Polycom" <sip:6012@10.100.69.1:6001;user=phone> Supported: timer,replaces,join,histinfo,100rel Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH RSeq: 1 Require: 100rel Content-Type: application/sdp Content-Length: 184 v=0 o=- 1 2 IN IP4 10.100.69.1 s=- c=IN IP4 10.100.69.2 b=AS:64 t=0 0 m=audio 2050 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 Received SIP PDU from ( 10.100.69.1:5060 ) SIP/2.0 180 Ringing From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 27 INVITE Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427 Record-Route: <sip:10.100.69.1:6001;lr;transport=tls> Record-Route: <sip:10.100.69.1:5060;lr> Server: Avaya CM/R015x.02.1.016.4 Contact: "Polycom" <sip:6012@10.100.69.1:6001;user=phone;transport=tls> P-Asserted-Identity: "Polycom" <sip:6012@10.100.69.1:6001;user=phone> Supported: timer,replaces,join,histinfo,100rel Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH RSeq: 1 Require: 100rel Content-Type: application/sdp Content-Length: 184 v=0 o=- 1 2 IN IP4 10.100.69.1 s=- c=IN IP4 10.100.69.2 b=AS:64 t=0 0 m=audio 2050 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 Received SIP PDU from ( 10.100.69.1:5060 ) SIP/2.0 180 Ringing From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 27 INVITE Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427 Record-Route: <sip:10.100.69.1:6001;lr;transport=tls> Record-Route: <sip:10.100.69.1:5060;lr> Server: Avaya CM/R015x.02.1.016.4 Contact: "Polycom" <sip:6012@10.100.69.1:6001;user=phone;transport=tls> P-Asserted-Identity: "Polycom" <sip:6012@10.100.69.1:6001;user=phone> Supported: timer,replaces,join,histinfo,100rel Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH RSeq: 1 Require: 100rel Content-Type: application/sdp Content-Length: 184 v=0 o=- 1 2 IN IP4 10.100.69.1 s=- c=IN IP4 10.100.69.2 b=AS:64 t=0 0 m=audio 2050 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 Received SIP PDU from ( 10.100.69.1:5060 ) SIP/2.0 504 Server Time-out From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=471CE0B3F3F312BA810FD40EEAC3794C1270120037123753 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 27 INVITE Via: SIP/2.0/UDP 10.115.64.98:5060;psrrposn=1;received=10.115.64.98;branch=z9hG4bK224d7919a427 Server: Avaya CM/R015x.02.1.016.4 Content-Length: 0 Sending SIP PDU to ( 10.100.69.1:5060 ) from 5060 ACK sip:6012@10.100.69.1 SIP/2.0 Via: SIP/2.0/UDP 10.115.64.98:5060;branch=z9hG4bK224d7919a427 From: <sip:6010@10.100.69.1>;tag=224d7919a4 To: <sip:6012@10.100.69.1>;tag=471CE0B3F3F312BA810FD40EEAC3794C1270120037123753 Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98 CSeq: 27 ACK Proxy-Authorization: Digest username="6010", realm="10.100.69.1", nonce="MTI3MDEyMDAzNjpTREZTZXJ2ZXJTZWNyZXRLZXk6MjA2Nzg0NTQ0Ng==", uri="sip:6012@10.100.69.1", response="7991b5da1f3102c5f03a9a71c41aab3e", algorithm=MD5 Content-Length: 0 Max-Forwards: 70
ram_scan Posted April 5, 2011 Posted April 5, 2011 Во первых отключите на аддпаке поддержку 100rel чтобы она prack-ами не бросалась. Во вторых у вас сессия вообще не стартует и весь голос идет (если идет) в предответе. Таймаут выходит (там по дефолту не то 20 не то 30 секунд), сессия рвется.
kanim Posted April 5, 2011 Author Posted April 5, 2011 Во вторых у вас сессия вообще не стартует и весь голос идет (если идет) в предответе. Таймаут выходит (там по дефолту не то 20 не то 30 секунд), сессия рвется. Вы, бы не могли подсказать возможные пути решения данной проблемы?
Voicemaster Posted April 5, 2011 Posted April 5, 2011 У вас 2 проблемы: SIP/2.0 422 Session Interval Too Small и PRACK'и. Звонок у вас вообще не состоялся.
Kristoff_Vampire Posted April 5, 2011 Posted April 5, 2011 да чёто не видно что трубку то подняли и пошёл разговор.
kanim Posted April 5, 2011 Author Posted April 5, 2011 да чёто не видно что трубку то подняли и пошёл разговор. Разговор там точно был, но не длительный. Т.е. на сколько я понял, то нужно отключить 100rel и увеличить время сессии Min-SE до 3600? Дело в том, что у меня ограниченный доступ к оборудованию и хотелось бы определиться с багами.
Voicemaster Posted April 5, 2011 Posted April 5, 2011 да чёто не видно что трубку то подняли и пошёл разговор. Разговор там точно был, но не длительный. Т.е. на сколько я понял, то нужно отключить 100rel и увеличить время сессии Min-SE до 3600? Дело в том, что у меня ограниченный доступ к оборудованию и хотелось бы определиться с багами. Да. После этого надо обязательно увидеть 200 OK & ACK в ответ на INVITE. Без этого доступ к оборудованию придется получать еще раз. .
kanim Posted April 8, 2011 Author Posted April 8, 2011 Отключил 100rel, увеличил Min-SE до 3600 и все заработало! Всем спасибо за помощь.
.png.5d2afa2996cc6a85d0f2c09b92dd0a28.png)
Recommended Posts