[asterisk-bugs] [JIRA] (ASTERISK-29613) chan_sip.c:22332 handle_request_invite: Unable to create/find SIP channel for this INVITE
Joshua C. Colp (JIRA)
noreply at issues.asterisk.org
Wed Aug 25 03:48:33 CDT 2021
[ https://issues.asterisk.org/jira/browse/ASTERISK-29613?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=256052#comment-256052 ]
Joshua C. Colp commented on ASTERISK-29613:
-------------------------------------------
We appreciate the difficulties you are facing, however this does not appear to be a bug report and your request or comments would be better served in a different forum.
The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.
Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.
Thanks!
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
> chan_sip.c:22332 handle_request_invite: Unable to create/find SIP channel for this INVITE
> -----------------------------------------------------------------------------------------
>
> Key: ASTERISK-29613
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-29613
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: . I did not set the category correctly.
> Affects Versions: 18.5.0
> Environment: CENTOS7
> Reporter: Bui Huu Quang
>
> Hi all,
> I see this err when call in.
> This is content of file sip.confg
> [general]
> externip = 171.244.50.xxx
> localnet = 171.244.50.xxx/255.255.255.0
> Context = mobitechs
> Port = 5060
> Srvlookup = yes
> Bindaddr = 0.0.0.0
> disallow=all
> Diallow = all
> allow = g729
> allow = g723
> allow = h261
> allow = h263
> allow = h263p
> Allow = alaw
> Allow = ulaw
> Allow = ilbc
> Nat = 1
> qualify = yes
> externrefresh = 1
> notifyringing = yes
> notifyhold = yes
> limitonpeers = yes
> videosupport = no
> callerid = Unknown
> tos = 0x68
> subscribecontext = device-hints
> subscribecontext = device-hints
> subscribecontext = device-hints
> subscribecontext = device-hints
> allowguest=no
> [trunk_GMSC22]
> type=peer
> host=10.226.2.2
> context=from_trunk_GMSC
> qualify=yes
> ;nat=no
> ;keepalive=45
> dtmfmode=rfc2833
> ;disallow=all
> ;allow=gsm
> ;allow=alaw
> ;allow=ulaw
> Canreinvite = no
> insecure=port,invite
> session-timers=refuse
> session-expires=1800
> session-minse=90
> session-refresher=uac
> [trunk_GMSC210]
> type=peer
> host=10.226.2.10
> context=from_trunk_GMSC
> qualify=yes
> ;nat=no
> ;keepalive=45
> dtmfmode=rfc2833
> ;disallow=all
> ;allow=gsm
> ;allow=alaw
> ;allow=ulaw
> Canreinvite = no
> insecure=port,invite
> session-timers=refuse
> session-expires=1800
> session-minse=90
> session-refresher=uac
> -----------------------------------------
> This is content of result of command : sip show peers
> localhost*CLI> sip show peers
> Name/username Host Dyn Forcerport ACL Port Status
> 2000/2000 (Unspecified) D N 0 UNKNOWN
> 2001/2001 (Unspecified) D N 0 UNKNOWN
> 2002/2002 (Unspecified) D N 0 UNKNOWN
> trunk_GMSC210 10.226.2.10 5060 UNREACHABLE
> trunk_GMSC22 10.226.2.2 N 5060 UNREACHABLE
> 5 sip peers [Monitored: 0 online, 5 offline Unmonitored: 0 online, 0 offline]
> When the call in i see log debug
> ---
> <--- SIP read from UDP:10.226.2.2:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK7078ab1e
> Call-ID: 5f618a412fe41f8d08abd036614d6608 at 10.226.39.49:5060
> From: "199"<sip:199 at 10.226.39.49>;tag=as5911872b
> To: <sip:10.226.2.2>
> CSeq: 102 OPTIONS
> Content-Length: 0
> <------------->
> --- (7 headers 0 lines) ---
> Retransmitting #4 (no NAT) to 10.226.2.10:5060:
> OPTIONS sip:10.226.2.10 SIP/2.0
> Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK2be5085a
> Max-Forwards: 70
> From: "199" <sip:199 at 10.226.39.49>;tag=as2f7f7b41
> To: <sip:10.226.2.10>
> Contact: <sip:199 at 10.226.39.49:5060>
> Call-ID: 0091b01e7074ba8c3cde57bc6c1b5d00 at 10.226.39.49:5060
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX 1.8.5.0
> Date: Wed, 25 Aug 2021 03:06:18 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces
> Content-Length: 0
> ---
> Really destroying SIP dialog '0091b01e7074ba8c3cde57bc6c1b5d00 at 10.226.39.49:5060' Method: OPTIONS
> <--- SIP read from UDP:10.226.2.10:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK2be5085a
> Call-ID: 0091b01e7074ba8c3cde57bc6c1b5d00 at 10.226.39.49:5060
> From: "199"<sip:199 at 10.226.39.49>;tag=as2f7f7b41
> To: <sip:10.226.2.10>
> CSeq: 102 OPTIONS
> Content-Length: 0
> <------------->
> --- (7 headers 0 lines) ---
> Retransmitting #4 (no NAT) to 10.226.2.2:5060:
> OPTIONS sip:10.226.2.2 SIP/2.0
> Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK7078ab1e
> Max-Forwards: 70
> From: "199" <sip:199 at 10.226.39.49>;tag=as5911872b
> To: <sip:10.226.2.2>
> Contact: <sip:199 at 10.226.39.49:5060>
> Call-ID: 5f618a412fe41f8d08abd036614d6608 at 10.226.39.49:5060
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX 1.8.5.0
> Date: Wed, 25 Aug 2021 03:06:18 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces
> Content-Length: 0
> ---
> Really destroying SIP dialog '5f618a412fe41f8d08abd036614d6608 at 10.226.39.49:5060' Method: OPTIONS
> <--- SIP read from UDP:10.226.2.2:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK7078ab1e
> Call-ID: 5f618a412fe41f8d08abd036614d6608 at 10.226.39.49:5060
> From: "199"<sip:199 at 10.226.39.49>;tag=as5911872b
> To: <sip:10.226.2.2>
> CSeq: 102 OPTIONS
> Content-Length: 0
> <------------->
> --- (7 headers 0 lines) ---
> <--- SIP read from UDP:10.226.2.2:5066 --->
> INVITE sip:199 at 10.226.39.49;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.226.2.2:5066;branch=z9hG4bKy0a4q104kkjyv3rbywwyqwqbu;X-DispMsg=1401
> Route: <sip:10.226.39.49:5060;transport=udp;lr>
> Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64
> From: "75666668"<sip:75666668 at 10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
> To: "199"<sip:199 at 10.226.39.49;transport=udp;user=phone>
> CSeq: 1 INVITE
> P-Access-Network-Info: GEN-ACCESS;"area-number=+6707"
> Max-Forwards: 70
> Contact: <sip:75666668 at 10.226.2.2:5060;user=phone>
> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE
> P-Asserted-Identity: <tel:75666668>
> P-Early-Media: supported
> Supported: 100rel,timer,histinfo,precondition
> Min-SE: 90
> Session-Expires: 1800;refresher=uac
> Content-Length: 682
> Content-Type: application/sdp
> v=0
> o=HuaweiSoftx3000 1076671184 1076671185 IN IP4 10.226.2.2
> s=SipCall
> c=IN IP4 10.226.1.132
> t=0 0
> m=audio 24920 RTP/AVP 108 8 18 116 100 107 105 3
> a=rtpmap:108 AMR/8000
> a=fmtp:108 mode-set=7
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:116 telephone-event/8000
> a=rtpmap:100 AMR/8000
> a=fmtp:100 mode-set=0,2,5,7;mode-change-neighbor=1;mode-change-period=2
> a=rtpmap:107 AMR/8000
> a=fmtp:107 mode-set=0,1,2,3,4,5;mode-change-neighbor=1;mode-change-period=2
> a=rtpmap:105 GSM-EFR/8000
> a=rtpmap:3 GSM/8000
> a=ptime:20
> a=maxptime:20
> a=curr:qos local none
> a=curr:qos remote none
> a=des:qos mandatory local sendrecv
> a=des:qos optional remote sendrecv
> a=3gOoBTC
> <------------->
> --- (18 headers 24 lines) ---
> Sending to 10.226.2.2:5066 (no NAT)
> Using INVITE request as basis request - 0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64
> Found peer 'trunk_GMSC22' for '75666668' from 10.226.2.2:5066
> == Using SIP RTP CoS mark 5
> Found RTP audio format 108
> Found RTP audio format 8
> Found RTP audio format 18
> Found RTP audio format 116
> Found RTP audio format 100
> Found RTP audio format 107
> Found RTP audio format 105
> Found RTP audio format 3
> Found unknown media description format AMR for ID 108
> Found audio description format PCMA for ID 8
> Found audio description format G729 for ID 18
> Found audio description format telephone-event for ID 116
> Found unknown media description format AMR for ID 100
> Found unknown media description format AMR for ID 107
> Found unknown media description format GSM-EFR for ID 105
> Found audio description format GSM for ID 3
> Capabilities: us - 0x1c050d (g723|ulaw|alaw|g729|ilbc|h261|h263|h263p), peer - audio=0x10a (gsm|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x108 (alaw|g729)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> Peer audio RTP is at port 10.226.1.132:24920
> Looking for 199 in from_trunk_GMSC (domain 10.226.39.49)
> list_route: hop: <sip:75666668 at 10.226.2.2:5060;user=phone>
> <--- Transmitting (no NAT) to 10.226.2.2:5066 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.226.2.2:5066;branch=z9hG4bKy0a4q104kkjyv3rbywwyqwqbu;X-DispMsg=1401;received=10.226.2.2
> From: "75666668"<sip:75666668 at 10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
> To: "199"<sip:199 at 10.226.39.49;transport=udp;user=phone>
> Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64
> CSeq: 1 INVITE
> Server: Asterisk PBX 1.8.5.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces
> Contact: <sip:199 at 10.226.39.49:5060>
> Content-Length: 0
> <------------>
> -- Executing [199 at from_trunk_GMSC:1] Answer("SIP/trunk_GMSC22-00000000", "") in new stack
> Audio is at 5060
> Adding codec 0x100 (g729) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> <--- Reliably Transmitting (no NAT) to 10.226.2.2:5066 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.226.2.2:5066;branch=z9hG4bKy0a4q104kkjyv3rbywwyqwqbu;X-DispMsg=1401;received=10.226.2.2
> From: "75666668"<sip:75666668 at 10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
> To: "199"<sip:199 at 10.226.39.49;transport=udp;user=phone>;tag=as0e4697cc
> Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64
> CSeq: 1 INVITE
> Server: Asterisk PBX 1.8.5.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces
> Contact: <sip:199 at 10.226.39.49:5060>
> Content-Type: application/sdp
> Content-Length: 310
> v=0
> o=root 1915328570 1915328570 IN IP4 10.226.39.49
> s=Asterisk PBX 1.8.5.0
> c=IN IP4 10.226.39.49
> t=0 0
> m=audio 27970 RTP/AVP 18 8 116
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:8 PCMA/8000
> a=rtpmap:116 telephone-event/8000
> a=fmtp:116 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> <------------>
> Retransmitting #1 (no NAT) to 10.226.2.2:5066:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.226.2.2:5066;branch=z9hG4bKy0a4q104kkjyv3rbywwyqwqbu;X-DispMsg=1401;received=10.226.2.2
> From: "75666668"<sip:75666668 at 10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
> To: "199"<sip:199 at 10.226.39.49;transport=udp;user=phone>;tag=as0e4697cc
> Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64
> CSeq: 1 INVITE
> Server: Asterisk PBX 1.8.5.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces
> Contact: <sip:199 at 10.226.39.49:5060>
> Content-Type: application/sdp
> Content-Length: 310
> v=0
> o=root 1915328570 1915328570 IN IP4 10.226.39.49
> s=Asterisk PBX 1.8.5.0
> c=IN IP4 10.226.39.49
> t=0 0
> m=audio 27970 RTP/AVP 18 8 116
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:8 PCMA/8000
> a=rtpmap:116 telephone-event/8000
> a=fmtp:116 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> ---
> -- Executing [199 at from_trunk_GMSC:2] Playback("SIP/trunk_GMSC22-00000000", "/ivrshared/voice/ivr/COMMON/thanks") in new stack
> [Aug 25 12:06:28] WARNING[27002]: channel.c:5064 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)
> [Aug 25 12:06:28] WARNING[27002]: file.c:950 ast_streamfile: Unable to open /ivrshared/voice/ivr/COMMON/thanks (format 0x100 (g729)): No such file or directory
> [Aug 25 12:06:28] WARNING[27002]: app_playback.c:471 playback_exec: ast_streamfile failed on SIP/trunk_GMSC22-00000000 for /ivrshared/voice/ivr/COMMON/thanks
> -- Executing [199 at from_trunk_GMSC:3] Hangup("SIP/trunk_GMSC22-00000000", "") in new stack
> == Spawn extension (from_trunk_GMSC, 199, 3) exited non-zero on 'SIP/trunk_GMSC22-00000000'
> Scheduling destruction of SIP dialog '0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64' in 6400 ms (Method: INVITE)
> Retransmitting #2 (no NAT) to 10.226.2.2:5066:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.226.2.2:5066;branch=z9hG4bKy0a4q104kkjyv3rbywwyqwqbu;X-DispMsg=1401;received=10.226.2.2
> From: "75666668"<sip:75666668 at 10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
> To: "199"<sip:199 at 10.226.39.49;transport=udp;user=phone>;tag=as0e4697cc
> Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64
> CSeq: 1 INVITE
> Server: Asterisk PBX 1.8.5.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces
> Contact: <sip:199 at 10.226.39.49:5060>
> Content-Type: application/sdp
> Content-Length: 310
> v=0
> o=root 1915328570 1915328570 IN IP4 10.226.39.49
> s=Asterisk PBX 1.8.5.0
> c=IN IP4 10.226.39.49
> t=0 0
> m=audio 27970 RTP/AVP 18 8 116
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:8 PCMA/8000
> a=rtpmap:116 telephone-event/8000
> a=fmtp:116 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> ---
> <--- SIP read from UDP:10.226.2.2:5066 --->
> ACK sip:199 at 10.226.39.49:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.226.2.2:5066;branch=z9hG4bKy240au3j4ayxvjx4xvv304y3k;X-DispMsg=1401
> Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64
> From: "75666668"<sip:75666668 at 10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
> To: "199"<sip:199 at 10.226.39.49;transport=udp;user=phone>;tag=as0e4697cc
> CSeq: 1 ACK
> Max-Forwards: 70
> Content-Length: 0
> <------------->
> --- (8 headers 0 lines) ---
> set_destination: Parsing <sip:75666668 at 10.226.2.2:5060;user=phone> for address/port to send to
> set_destination: set destination to 10.226.2.2:5060
> Reliably Transmitting (no NAT) to 10.226.2.2:5060:
> BYE sip:75666668 at 10.226.2.2:5060;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK1ab9b6db
> Max-Forwards: 70
> From: "199"<sip:199 at 10.226.39.49;transport=udp;user=phone>;tag=as0e4697cc
> To: "75666668"<sip:75666668 at 10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
> Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64
> CSeq: 102 BYE
> User-Agent: Asterisk PBX 1.8.5.0
> X-Asterisk-HangupCause: Normal Clearing
> X-Asterisk-HangupCauseCode: 16
> Content-Length: 0
> ---
> Scheduling destruction of SIP dialog '0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64' in 6400 ms (Method: ACK)
> <--- SIP read from UDP:10.226.2.2:5066 --->
> INVITE sip:199 at 10.226.39.49:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.226.2.2:5066;branch=z9hG4bKb2j31xwk4bjy2u4aaqrx1qbj1;X-DispMsg=1401
> Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64
> From: "75666668"<sip:75666668 at 10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
> To: "199"<sip:199 at 10.226.39.49;transport=udp;user=phone>;tag=as0e4697cc
> CSeq: 2 INVITE
> Max-Forwards: 70
> Contact: <sip:10.226.2.2:5060>
> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER
> Supported: timer
> Content-Length: 226
> Content-Type: application/sdp
> v=0
> o=HuaweiSoftx3000 1076671184 1076671186 IN IP4 10.226.2.2
> s=SipCall
> c=IN IP4 10.226.1.132
> t=0 0
> m=audio 24920 RTP/AVP 18 116
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:116 telephone-event/8000
> a=ptime:20
> <------------->
> --- (12 headers 10 lines) ---
> Sending to 10.226.2.2:5066 (no NAT)
> Using INVITE request as basis request - 0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64
> [Aug 25 12:06:29] NOTICE[23080]: chan_sip.c:22332 handle_request_invite: Unable to create/find SIP channel for this INVITE
> <--- Reliably Transmitting (no NAT) to 10.226.2.2:5066 --->
> SIP/2.0 503 Unavailable
> Via: SIP/2.0/UDP 10.226.2.2:5066;branch=z9hG4bKb2j31xwk4bjy2u4aaqrx1qbj1;X-DispMsg=1401;received=10.226.2.2
> From: "75666668"<sip:75666668 at 10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
> To: "199"<sip:199 at 10.226.39.49;transport=udp;user=phone>;tag=as0e4697cc
> Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64
> CSeq: 2 INVITE
> Server: Asterisk PBX 1.8.5.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces
> Content-Length: 0
> <------------>
> Scheduling destruction of SIP dialog '0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64' in 6400 ms (Method: INVITE)
> <--- SIP read from UDP:10.226.2.2:5066 --->
> ACK sip:199 at 10.226.39.49:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.226.2.2:5066;branch=z9hG4bKb2j31xwk4bjy2u4aaqrx1qbj1;X-DispMsg=1401
> Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64
> From: "75666668"<sip:75666668 at 10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
> To: "199"<sip:199 at 10.226.39.49;transport=udp;user=phone>;tag=as0e4697cc
> CSeq: 2 ACK
> Max-Forwards: 70
> Content-Length: 0
> <------------->
> --- (8 headers 0 lines) ---
> <--- SIP read from UDP:10.226.2.2:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK1ab9b6db
> Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64
> From: "199"<sip:199 at 10.226.27.44:65476;transport=udp;user=phone>;tag=as0e4697cc
> To: "75666668"<sip:75666668 at 10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
> CSeq: 102 BYE
> Content-Length: 0
> <------------->
> --- (7 headers 0 lines) ---
> SIP Response message for INCOMING dialog BYE arrived
> Really destroying SIP dialog '0x0v2yxvybqxy4xxjbwwaujvxv2quyub at 10.18.5.64' Method: ACK
> Reliably Transmitting (no NAT) to 10.226.2.10:5060:
> OPTIONS sip:10.226.2.10 SIP/2.0
> Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK4e5d45c2
> Max-Forwards: 70
> From: "199" <sip:199 at 10.226.39.49>;tag=as59c44cb0
> To: <sip:10.226.2.10>
> Contact: <sip:199 at 10.226.39.49:5060>
> Call-ID: 0c78cfcf0a27badb28fc9f901272b01a at 10.226.39.49:5060
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX 1.8.5.0
> Date: Wed, 25 Aug 2021 03:06:32 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces
> Content-Length: 0
> ---
> Reliably Transmitting (no NAT) to 10.226.2.2:5060:
> OPTIONS sip:10.226.2.2 SIP/2.0
> Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK67c44e44
> Max-Forwards: 70
> From: "199" <sip:199 at 10.226.39.49>;tag=as275c8aa4
> To: <sip:10.226.2.2>
> Contact: <sip:199 at 10.226.39.49:5060>
> Call-ID: 31097ba07c0dbb06175ced730b069255 at 10.226.39.49:5060
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX 1.8.5.0
> Date: Wed, 25 Aug 2021 03:06:32 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces
> Content-Length: 0
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