[asterisk-bugs] [JIRA] (ASTERISK-29613) chan_sip.c:22332 handle_request_invite: Unable to create/find SIP channel for this INVITE
Bui Huu Quang (JIRA)
noreply at issues.asterisk.org
Wed Aug 25 01:52:33 CDT 2021
Bui Huu Quang created ASTERISK-29613:
----------------------------------------
Summary: 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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