[asterisknow] SIP transaction detruct automaticaly
bkruse
bkruse at digium.com
Mon Dec 15 14:34:50 CST 2008
I really am not sure what you mean, but you will have much better luck
on the mailing list of the Asterisk-users
Thanks!
-bk
rain wrote:
> 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: eibbcnlfaoofljio1 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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