[asterisknow] SIP transaction detruct automaticaly
rain
test_sir at 126.com
Sat Dec 13 08:40:40 CST 2008
Dear all,
I got a problem like the following.Tansaction will destruct by asterisk in 32000ms.May I know hot to set 32000ms to 60000ms or much longer?
Thank you all.
Best Regards.
####Log from sip set debug########
<--- SIP read from 251.2.1.75:5070 --->
INVITE sip:008613699401263 at 232.25.1.221:5060 SIP/2.0
Via: SIP/2.0/UDP 251.2.1.75:5070;branch=z9hG4bK-Koncept4de8c0752098bc57-1--K
Max-Forwards: 70
Contact: <sip:4563200460213 at 251.2.1.75:5070>
To: 008613699401263<sip:008613699401263 at 232.25.1.221>
From: 4563200460213<sip:4563200460213 at sip99.darenjp.com>;tag=c78c4f76
Call-ID: eibbcnlfaoofljio.1 at 122917849930879
CSeq: 1 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, REFER, NOTIFY, MESSAGE
Content-Type: application/sdp
User-Agent: CityMoon SIP/1.8.0.013
Content-Length: 244
v=0
o=root 1229178501 1229178501 IN IP4 251.2.1.76
s=Koncept Session
c=IN IP4 251.2.1.76
t=0 0
m=audio 38180 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
<------------->
--- (12 headers 10 lines) ---
Sending to 251.2.1.75 : 5070 (NAT)
Using INVITE request as basis request - eibbcnlfaoofljio.1 at 122917849930879
Found peer 'daren75'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 251.2.1.76:38180
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8010f (g723|gsm|ulaw|alaw|g729|h263), peer - audio=0x10c (u
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone
Peer audio RTP is at port 251.2.1.76:38180
Looking for 008613699401263 in from-daren (domain 232.25.1.221)
list_route: hop: <sip:4563200460213 at 251.2.1.75:5070>
s2p4*CLI>
<--- Transmitting (no NAT) to 251.2.1.75:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 251.2.1.75:5070;branch=z9hG4bK-Koncept4de8c0752098bc57-1--K
From: 4563200460213<sip:4563200460213 at sip99.darenjp.com>;tag=c78c4f76
To: 008613699401263<sip:008613699401263 at 232.25.1.221>
Call-ID: eibbcnlfaoofljio.1 at 122917849930879
CSeq: 1 INVITE
User-Agent: MVTS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:008613699401263 at 232.25.1.221>
Content-Length: 0
<------------>
Audio is at 127.0.1.1 port 12054
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 127.0.1.1:5061:
INVITE sip:008613699401263 at 127.0.1.1:5061 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK06b799de;rport
From: "4563200460213" <sip:4563200460213 at 127.0.1.1>;tag=as59b6d969
To: <sip:008613699401263 at 127.0.1.1:5061>
Contact: <sip:4563200460213 at 127.0.1.1>
Call-ID: 26d66cf93a7e517d4b1f750b5e2be72f at 127.0.1.1
CSeq: 102 INVITE
User-Agent: MVTS
Max-Forwards: 70
Date: Sat, 13 Dec 2008 14:29:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 279
v=0
o=root 4429 4429 IN IP4 127.0.1.1
s=session
c=IN IP4 127.0.1.1
t=0 0
m=audio 12054 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- Transmitting (no NAT) to 251.2.1.75:5070 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 251.2.1.75:5070;branch=z9hG4bK-Koncept4de8c0752098bc57-1--K
From: 4563200460213<sip:4563200460213 at sip99.darenjp.com>;tag=c78c4f76
To: 008613699401263<sip:008613699401263 at 232.25.1.221>;tag=as658b048b
Call-ID: eibbcnlfaoofljio.1 at 122917849930879
CSeq: 1 INVITE
User-Agent: MVTS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:008613699401263 at 232.25.1.221>
Content-Length: 0
<------------>
s2p4*CLI>
<--- SIP read from 127.0.1.1:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK06b799de;rport=5060
From: "4563200460213" <sip:4563200460213 at 127.0.1.1>;tag=as59b6d969
To: <sip:008613699401263 at 127.0.1.1:5061>
Call-ID: 26d66cf93a7e517d4b1f750b5e2be72f at 127.0.1.1
CSeq: 102 INVITE
User-Agent: s2p4
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
s2p4*CLI>
Really destroying SIP dialog '3A841799F37347DF9D2E5B309DCC35AF0xc0a80166' Method
Really destroying SIP dialog 'eibbcmllaonpbfch.1 at 12291819819481' Method: ACK
Scheduling destruction of SIP dialog '26d66cf93a7e517d4b1f750b5e2be72f at 127.0.1.1
Reliably Transmitting (no NAT) to 127.0.1.1:5061:
CANCEL sip:008613699401263 at 127.0.1.1:5061 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK06b799de;rport
From: "4563200460213" <sip:4563200460213 at 127.0.1.1>;tag=as59b6d969
To: <sip:008613699401263 at 127.0.1.1:5061>
Call-ID: 26d66cf93a7e517d4b1f750b5e2be72f at 127.0.1.1
CSeq: 102 CANCEL
User-Agent: MVTS
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of SIP dialog '26d66cf93a7e517d4b1f750b5e2be72f at 127.0.1.1
Scheduling destruction of SIP dialog 'eibbcnlfaoofljio.1 at 122917849930879' in 320
<--- Reliably Transmitting (no NAT) to 251.2.1.75:5070 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 251.2.1.75:5070;branch=z9hG4bK-Koncept4de8c0752098bc57-1--K
From: 4563200460213<sip:4563200460213 at sip99.darenjp.com>;tag=c78c4f76
To: 008613699401263<sip:008613699401263 at 232.25.1.221>;tag=as658b048b
Call-ID: eibbcnlfaoofljio.1 at 122917849930879
CSeq: 1 INVITE
User-Agent: MVTS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:008613699401263 at 232.25.1.221>
Content-Length: 0
<------------>
s2p4*CLI>
<--- SIP read from 127.0.1.1:5061 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK06b799de;rport=5060
From: "4563200460213" <sip:4563200460213 at 127.0.1.1>;tag=as59b6d969
To: <sip:008613699401263 at 127.0.1.1:5061>;tag=2131591594
Call-ID: 26d66cf93a7e517d4b1f750b5e2be72f at 127.0.1.1
CSeq: 102 INVITE
User-Agent: s2p4
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 127.0.1.1:5061:
ACK sip:008613699401263 at 127.0.1.1:5061 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK06b799de;rport
From: "4563200460213" <sip:4563200460213 at 127.0.1.1>;tag=as59b6d969
To: <sip:008613699401263 at 127.0.1.1:5061>;tag=2131591594
Contact: <sip:4563200460213 at 127.0.1.1>
Call-ID: 26d66cf93a7e517d4b1f750b5e2be72f at 127.0.1.1
CSeq: 102 ACK
User-Agent: MVTS
Max-Forwards: 70
Content-Length: 0
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