[asterisk-bugs] [JIRA] (ASTERISK-21299) Asterisk send wrong codec order in the leg B of the call
Jesus Tovar (JIRA)
noreply at issues.asterisk.org
Tue Mar 19 02:21:01 CDT 2013
Jesus Tovar created ASTERISK-21299:
--------------------------------------
Summary: Asterisk send wrong codec order in the leg B of the call
Key: ASTERISK-21299
URL: https://issues.asterisk.org/jira/browse/ASTERISK-21299
Project: Asterisk
Issue Type: Bug
Security Level: None
Components: Channels/chan_sip/CodecHandling
Affects Versions: 11.2.0, 10.12.0
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
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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