[asterisk-bugs] [JIRA] (ASTERISK-21300) Asterisk send wrong codec order in the leg B of the call
Jesus Tovar (JIRA)
noreply at issues.asterisk.org
Tue Mar 19 02:21:02 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-21300?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Jesus Tovar updated ASTERISK-21300:
-----------------------------------
Attachment: SIP Debug PBX V11_2_ticket.txt
Tshark and Sip Debug information
> Asterisk send wrong codec order in the leg B of the call
> --------------------------------------------------------
>
> Key: ASTERISK-21300
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-21300
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/CodecHandling
> Affects Versions: 10.12.0, 11.2.0
> Environment: Centos 6.3 Linux rdns.vpdoc.com 2.6.32-279.22.1.el6.i686 #1 SMP Wed Feb 6 00:31:03 UTC 2013 i686 i686 i386 GNU/Linux
> Reporter: Jesus Tovar
> Severity: Critical
> Attachments: SIP Debug PBX V11_2_ticket.txt
>
>
> Basically the issue is focused on the codec order of the second leg of the call (PBX --> callee leg), where the PBX sets ulaw as first choice of the SDP whatever is the order in the configuration files of the peers, this order of codecs in the SDP makes the callee phone to pick up ulaw since it is already among the preference list of codecs. This issue is generating a permanent transcoding inside the server.
> Whatever the codec order is in the Leg A of the call, the codec order in the Leg B is ulaw.
> SDP_Caller:
> Media Format: ITU-T G.711 PCMA
> Media Format: ITU-T G.711 PCMU
> Media Format: ITU-T G.729
> SDP_Callee:
> Media Format: ITU-T G.711 PCMU
> Media Format: ITU-T G.729
> Media Format: ITU-T G.711 PCMA
> Caller (g729/alaw/ulaw) ---> PBX --> (ulaw/alaw/g729) transcoding.
> Caller (alaw/ulaw/g729) ---> PBX --> (ulaw/alaw/g729) transcoding.
> Below there is a SIP Debug that shows clearly the failure:
> Trace Information:
> Number A (Caller): 3135
> IP Address Caller: 10.1.1.237
>
> Number B (Callee): 3088
> IP Address Callee: 10.1.1.240
>
> Sip.conf
>
> disallow=all ; First disallow all codecs
> allow=h264
> allow=g729 ; Allow codecs in order of preference
> allow=alaw
> allow=ulaw
>
> [3135]
> type=peer
> secret=7wq
> canreinvite=no
> host=dynamic
> context=test
> callerid="PBX CLIENT t6" <3135>
> videosupport=yes
> mailbox=3135 at default
> qualify=10000
> [3088]
> type=peer
> secret=HIw
> canreinvite=no
> host=dynamic
> context=test
> callerid="PBX CLIENT t6" <3088>
> videosupport=yes
> mailbox=3088 at default
> qualify=10000
> testing*CLI>
> testing*CLI> sip set debug on
> SIP Debugging enabled
> <--- SIP read from UDP:10.1.1.240:32200 --->
> <------------->
> <--- SIP read from UDP:10.1.1.237:42936 --->
> INVITE sip:3088 at 10.1.1.236:5066;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bK8156bf5778454b05
> From: "VirtualHospital-RDT6" <sip:3135 at 10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
> To: <sip:3088 at 10.1.1.236:5066;user=phone>
> Contact: <sip:3135 at 10.1.1.237:42936;transport=udp;user=phone>
> Supported: replaces, timer, path
> Call-ID: 87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237
> CSeq: 5661 INVITE
> User-Agent: Grandstream GXV3000 1.2.3.7
> Max-Forwards: 70
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Content-Type: application/sdp
> Content-Length: 447
> v=0
> o=3135 8000 8000 IN IP4 10.1.1.237
> s=SIP Call
> c=IN IP4 10.1.1.237
> t=0 0
> m=audio 61432 RTP/AVP 8 0 18 101
> a=sendrecv
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=ptime:40
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
> m=video 61434 RTP/AVP 99
> a=sendrecv
> a=rtpmap:99 H264/90000
> a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg==
> a=framerate:12
> <------------->
> --- (13 headers 18 lines) ---
> Sending to 10.1.1.237:42936 (no NAT)
> Using INVITE request as basis request - 87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237
> Found peer '3135' for '3135' from 10.1.1.237:42936
> <--- Reliably Transmitting (no NAT) to 10.1.1.237:42936 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bK8156bf5778454b05;received=10.1.1.237
> From: "VirtualHospital-RDT6" <sip:3135 at 10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
> To: <sip:3088 at 10.1.1.236:5066;user=phone>;tag=as1fa61e2e
> Call-ID: 87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237
> CSeq: 5661 INVITE
> Server: PBX PBX 11.2.1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="PBX", nonce="4b547a92"
> Content-Length: 0
> <------------>
> Scheduling destruction of SIP dialog '87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237' in 6400 ms (Method: INVITE)
> <--- SIP read from UDP:10.1.1.237:42936 --->
> ACK sip:3088 at 10.1.1.236:5066;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bK8156bf5778454b05
> From: "VirtualHospital-RDT6" <sip:3135 at 10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
> To: <sip:3088 at 10.1.1.236:5066;user=phone>;tag=as1fa61e2e
> Contact: <sip:3135 at 10.1.1.237:42936;transport=udp;user=phone>
> Call-ID: 87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237
> CSeq: 5661 ACK
> User-Agent: Grandstream GXV3000 1.2.3.7
> Max-Forwards: 70
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Content-Length: 0
> <------------->
> --- (11 headers 0 lines) ---
> <--- SIP read from UDP:10.1.1.237:42936 --->
> INVITE sip:3088 at 10.1.1.236:5066;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKec101f6561e07d30
> From: "VirtualHospital-RDT6" <sip:3135 at 10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
> To: <sip:3088 at 10.1.1.236:5066;user=phone>
> Contact: <sip:3135 at 10.1.1.237:42936;transport=udp;user=phone>
> Supported: replaces, timer, path
> Authorization: Digest username="3135", realm="PBX", algorithm=MD5, uri="sip:3088 at 10.1.1.236:5066;user=phone", nonce="4b547a92", response="199a546451f4d26bce7f8b623be1a32d"
> Call-ID: 87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237
> CSeq: 5662 INVITE
> User-Agent: Grandstream GXV3000 1.2.3.7
> Max-Forwards: 70
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Content-Type: application/sdp
> Content-Length: 447
> v=0
> o=3135 8000 8001 IN IP4 10.1.1.237
> s=SIP Call
> c=IN IP4 10.1.1.237
> t=0 0
> m=audio 61432 RTP/AVP 8 0 18 101
> a=sendrecv
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=ptime:40
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
> m=video 61434 RTP/AVP 99
> a=sendrecv
> a=rtpmap:99 H264/90000
> a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg==
> a=framerate:12
> <------------->
> --- (14 headers 18 lines) ---
> Sending to 10.1.1.237:42936 (no NAT)
> Using INVITE request as basis request - 87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237
> Found peer '3135' for '3135' from 10.1.1.237:42936
> Found RTP audio format 8
> Found RTP audio format 0
> Found RTP audio format 18
> Found RTP audio format 101
> Found audio description format PCMA for ID 8
> Found audio description format PCMU for ID 0
> Found audio description format G729 for ID 18
> Found audio description format telephone-event for ID 101
> Found RTP video format 99
> Found video description format H264 for ID 99
> Capabilities: us - (ulaw|alaw|g729|h264), peer - audio=(ulaw|alaw|g729)/video=(h264)/text=(nothing), combined - (ulaw|alaw|g729|h264)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> Peer audio RTP is at port 10.1.1.237:61432
> Peer video RTP is at port 10.1.1.237:61434
> Looking for 3088 in test (domain 10.1.1.236)
> list_route: hop: <sip:3135 at 10.1.1.237:42936;transport=udp;user=phone>
> <--- Transmitting (no NAT) to 10.1.1.237:42936 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKec101f6561e07d30;received=10.1.1.237
> From: "VirtualHospital-RDT6" <sip:3135 at 10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
> To: <sip:3088 at 10.1.1.236:5066;user=phone>
> Call-ID: 87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237
> CSeq: 5662 INVITE
> Server: PBX PBX 11.2.1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Contact: <sip:3088 at 10.1.1.236:5066>
> Content-Length: 0
> <------------>
> Audio is at 14418
> Video is at 10.1.1.236:14292
> Adding codec 100003 (ulaw) to SDP
> Adding video codec 200004 (h264) to SDP
> Adding codec 100008 (g729) to SDP
> Adding codec 100004 (alaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 10.1.1.240:32200:
> INVITE sip:3088 at 10.1.1.240:32200;transport=udp;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK19b27029
> Max-Forwards: 70
> From: "PBX CLIENT t6" <sip:3135 at 10.1.1.236:5066>;tag=as54d904ac
> To: <sip:3088 at 10.1.1.240:32200;transport=udp;user=phone>
> Contact: <sip:3135 at 10.1.1.236:5066>
> Call-ID: 20f762cf712c4a85090df4402cc71d56 at 10.1.1.236:5066
> CSeq: 102 INVITE
> User-Agent: PBX PBX 11.2.1
> Date: Mon, 18 Mar 2013 06:05:11 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 446
> v=0
> o=root 1112394555 1112394555 IN IP4 10.1.1.236
> s=PBX PBX 11.2.1
> c=IN IP4 10.1.1.236
> b=CT:384
> t=0 0
> m=audio 14418 RTP/AVP 0 18 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> m=video 14292 RTP/AVP 99
> a=rtpmap:99 H264/90000
> a=fmtp:99 profile-level-id=428014
> a=sendrecv
> ---
> <--- SIP read from UDP:10.1.1.240:32200 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK19b27029
> From: "PBX CLIENT t6" <sip:3135 at 10.1.1.236:5066>;tag=as54d904ac
> To: <sip:3088 at 10.1.1.240:32200;transport=udp;user=phone>
> Call-ID: 20f762cf712c4a85090df4402cc71d56 at 10.1.1.236:5066
> CSeq: 102 INVITE
> User-Agent: Grandstream GXV3000 1.2.3.7
> Content-Length: 0
> <------------->
> --- (8 headers 0 lines) ---
> <--- SIP read from UDP:10.1.1.240:32200 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK19b27029
> From: "PBX CLIENT t6" <sip:3135 at 10.1.1.236:5066>;tag=as54d904ac
> To: <sip:3088 at 10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4
> Call-ID: 20f762cf712c4a85090df4402cc71d56 at 10.1.1.236:5066
> CSeq: 102 INVITE
> User-Agent: Grandstream GXV3000 1.2.3.7
> Contact: <sip:3088 at 10.1.1.240:32200;transport=udp;user=phone>
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Content-Length: 0
> <------------->
> --- (10 headers 0 lines) ---
> list_route: hop: <sip:3088 at 10.1.1.240:32200;transport=udp;user=phone>
> <--- Transmitting (no NAT) to 10.1.1.237:42936 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKec101f6561e07d30;received=10.1.1.237
> From: "VirtualHospital-RDT6" <sip:3135 at 10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
> To: <sip:3088 at 10.1.1.236:5066;user=phone>;tag=as4b2e34fd
> Call-ID: 87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237
> CSeq: 5662 INVITE
> Server: PBX PBX 11.2.1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Contact: <sip:3088 at 10.1.1.236:5066>
> Content-Length: 0
> <------------>
> Really destroying SIP dialog 'dc32e644caf312f6ab938814b3914792 at 10.1.1.240' Method: REGISTER
> <--- SIP read from UDP:10.1.1.240:32200 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK19b27029
> From: "PBX CLIENT t6" <sip:3135 at 10.1.1.236:5066>;tag=as54d904ac
> To: <sip:3088 at 10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4
> Call-ID: 20f762cf712c4a85090df4402cc71d56 at 10.1.1.236:5066
> CSeq: 102 INVITE
> User-Agent: Grandstream GXV3000 1.2.3.7
> Contact: <sip:3088 at 10.1.1.240:32200;transport=udp;user=phone>
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Content-Type: application/sdp
> Supported: replaces, timer, 100rel, path
> Content-Length: 395
> v=0
> o=3088 8000 8000 IN IP4 10.1.1.240
> s=SIP Call
> c=IN IP4 10.1.1.240
> t=0 0
> m=audio 3084 RTP/AVP 0 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
> m=video 3086 RTP/AVP 99
> a=sendrecv
> a=rtpmap:99 H264/90000
> a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg==
> a=framerate:12
> <------------->
> --- (12 headers 16 lines) ---
> Found RTP audio format 0
> Found RTP audio format 101
> Found audio description format PCMU for ID 0
> Found audio description format telephone-event for ID 101
> Found RTP video format 99
> Found video description format H264 for ID 99
> Capabilities: us - (ulaw|alaw|g729|h264), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> Peer audio RTP is at port 10.1.1.240:3084
> Peer video RTP is at port 10.1.1.240:3086
> list_route: hop: <sip:3088 at 10.1.1.240:32200;transport=udp;user=phone>
> set_destination: Parsing <sip:3088 at 10.1.1.240:32200;transport=udp;user=phone> for address/port to send to
> set_destination: set destination to 10.1.1.240:32200
> Transmitting (no NAT) to 10.1.1.240:32200:
> ACK sip:3088 at 10.1.1.240:32200;transport=udp;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK45dec5cb
> Max-Forwards: 70
> From: "PBX CLIENT t6" <sip:3135 at 10.1.1.236:5066>;tag=as54d904ac
> To: <sip:3088 at 10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4
> Contact: <sip:3135 at 10.1.1.236:5066>
> Call-ID: 20f762cf712c4a85090df4402cc71d56 at 10.1.1.236:5066
> CSeq: 102 ACK
> User-Agent: PBX PBX 11.2.1
> Content-Length: 0
> ---
> Audio is at 13140
> Video is at 10.1.1.236:16842
> Adding video codec 200004 (h264) to SDP
> Adding codec 100008 (g729) to SDP
> Adding codec 100004 (alaw) to SDP
> Adding codec 100003 (ulaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> <--- Reliably Transmitting (no NAT) to 10.1.1.237:42936 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKec101f6561e07d30;received=10.1.1.237
> From: "VirtualHospital-RDT6" <sip:3135 at 10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
> To: <sip:3088 at 10.1.1.236:5066;user=phone>;tag=as4b2e34fd
> Call-ID: 87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237
> CSeq: 5662 INVITE
> Server: PBX PBX 11.2.1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Contact: <sip:3088 at 10.1.1.236:5066>
> Content-Type: application/sdp
> Content-Length: 446
> v=0
> o=root 1232695348 1232695348 IN IP4 10.1.1.236
> s=PBX PBX 11.2.1
> c=IN IP4 10.1.1.236
> b=CT:384
> t=0 0
> m=audio 13140 RTP/AVP 18 8 0 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> m=video 16842 RTP/AVP 99
> a=rtpmap:99 H264/90000
> a=fmtp:99 profile-level-id=428014
> a=sendrecv
> <------------>
> <--- SIP read from UDP:10.1.1.237:42936 --->
> ACK sip:3088 at 10.1.1.236:5066 SIP/2.0
> Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKcb05b5a399021ff0
> From: "VirtualHospital-RDT6" <sip:3135 at 10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
> To: <sip:3088 at 10.1.1.236:5066;user=phone>;tag=as4b2e34fd
> Contact: <sip:3135 at 10.1.1.237:42936;transport=udp;user=phone>
> Authorization: Digest username="3135", realm="PBX", algorithm=MD5, uri="sip:3088 at 10.1.1.236:5066;user=phone", nonce="4b547a92", response="199a546451f4d26bce7f8b623be1a32d"
> Call-ID: 87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237
> CSeq: 5662 ACK
> User-Agent: Grandstream GXV3000 1.2.3.7
> Max-Forwards: 70
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Content-Length: 0
> <------------->
> --- (12 headers 0 lines) ---
> Really destroying SIP dialog 'bf1e2303-f567eb7e-f1b433c9 at 10.80.7.217' Method: REGISTER
> <--- SIP read from UDP:10.1.1.240:32200 --->
> BYE sip:3135 at 10.1.1.236:5066 SIP/2.0
> Via: SIP/2.0/UDP 10.1.1.240:32200;branch=z9hG4bKa26238e6e0617643
> From: <sip:3088 at 10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4
> To: "PBX CLIENT t6" <sip:3135 at 10.1.1.236:5066>;tag=as54d904ac
> Call-ID: 20f762cf712c4a85090df4402cc71d56 at 10.1.1.236:5066
> CSeq: 2839 BYE
> User-Agent: Grandstream GXV3000 1.2.3.7
> Max-Forwards: 70
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Reason: SIP ;text="Onhook event"
> Content-Length: 0
> <------------->
> --- (11 headers 0 lines) ---
> Sending to 10.1.1.240:32200 (no NAT)
> Scheduling destruction of SIP dialog '20f762cf712c4a85090df4402cc71d56 at 10.1.1.236:5066' in 6400 ms (Method: BYE)
> <--- Transmitting (no NAT) to 10.1.1.240:32200 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.1.240:32200;branch=z9hG4bKa26238e6e0617643;received=10.1.1.240
> From: <sip:3088 at 10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4
> To: "PBX CLIENT t6" <sip:3135 at 10.1.1.236:5066>;tag=as54d904ac
> Call-ID: 20f762cf712c4a85090df4402cc71d56 at 10.1.1.236:5066
> CSeq: 2839 BYE
> Server: PBX PBX 11.2.1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Content-Length: 0
> <------------>
> Scheduling destruction of SIP dialog '87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237' in 6400 ms (Method: ACK)
> set_destination: Parsing <sip:3135 at 10.1.1.237:42936;transport=udp;user=phone> for address/port to send to
> set_destination: set destination to 10.1.1.237:42936
> Reliably Transmitting (no NAT) to 10.1.1.237:42936:
> BYE sip:3135 at 10.1.1.237:42936;transport=udp;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK121773f3
> Max-Forwards: 70
> From: <sip:3088 at 10.1.1.236:5066;user=phone>;tag=as4b2e34fd
> To: "VirtualHospital-RDT6" <sip:3135 at 10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
> Call-ID: 87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237
> CSeq: 102 BYE
> User-Agent: PBX PBX 11.2.1
> Proxy-Authorization: Digest username="3135", realm="PBX", algorithm=MD5, uri="sip:10.1.1.236", nonce="4b547a92", response="d885ef09ee905ad79fb5c45e490084f1"
> X-PBX-HangupCause: Normal Clearing
> X-PBX-HangupCauseCode: 16
> Content-Length: 0
> ---
> <--- SIP read from UDP:10.1.1.237:42936 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK121773f3
> From: <sip:3088 at 10.1.1.236:5066;user=phone>;tag=as4b2e34fd
> To: "VirtualHospital-RDT6" <sip:3135 at 10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
> Call-ID: 87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237
> CSeq: 102 BYE
> User-Agent: Grandstream GXV3000 1.2.3.7
> Contact: <sip:3135 at 10.1.1.237:42936;transport=udp;user=phone>
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Supported: replaces, timer, 100rel, path
> Content-Length: 0
> <------------->
> --- (11 headers 0 lines) ---
> SIP Response message for INCOMING dialog BYE arrived
> Really destroying SIP dialog '87b5f796d6d01167924739f5bcb65c66 at 10.1.1.237' Method: ACK
> testing*CLI> sip set debug off
> SIP Debugging Disabled
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