форумы  помощь  поиск  регистрация  майллист  вход/выход  слежка

Вариант для распечатки		Пред. тема | След. тема
Форум Маршрутизаторы CISCO и др. оборудование. (VoIP)
Изначальное сообщение		[ Отслеживать ]

"Обрывается исходящий звонок на Addpac IP200"	+1 +/–
Сообщение от kanim (ok) on 28-Мрт-11, 10:22
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.
Ответить | Правка | Cообщить модератору

Сообщения по теме

[Сортировка по времени | RSS]

1. "Обрывается исходящий звонок на Addpac IP200"	+1 +/–
Сообщение от Николай (??) on 28-Мрт-11, 11:54
Был такой случай у меня когда не совпадали значения   Sess-Expires : 1800 secs   Min-Sess     : 90 secs на стороне клиента и сервера. На сервере стоял Sess-Expires : 3600 secs поменял и все стало отлично.
Ответить | Правка | ^ к родителю #0 | Наверх | Cообщить модератору

2. "Обрывается исходящий звонок на Addpac IP200"	+/–
Сообщение от kanim (ok) on 01-Апр-11, 15:25
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
Ответить | Правка | ^ к родителю #0 | Наверх | Cообщить модератору

	3. "Обрывается исходящий звонок на Addpac IP200"	+/–
	Сообщение от kanim (ok) on 08-Апр-11, 15:24
	Отключил 100rel, увеличил Min-SE до 3600 и все заработало! Всем спасибо за помощь.
	Ответить | Правка | ^ к родителю #2 | Наверх | Cообщить модератору

Архив | Удалить

Рекомендовать для помещения в FAQ | Индекс форумов | Темы | Пред. тема | След. тема