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Summary:ASTERISK-29613: chan_sip.c:22332 handle_request_invite: Unable to create/find SIP channel for this INVITE
Reporter:Bui Huu Quang (quangbh)Labels:
Date Opened:2021-08-25 01:51:22Date Closed:2021-08-25 03:48:14
Priority:MinorRegression?
Status:Closed/CompleteComponents:. I did not set the category correctly.
Versions:18.5.0 Frequency of
Occurrence
Related
Issues:
Environment:CENTOS7Attachments:
Description: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@10.226.39.49:5060
From: "199"<sip:199@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@10.226.39.49>;tag=as2f7f7b41
To: <sip:10.226.2.10>
Contact: <sip:199@10.226.39.49:5060>
Call-ID: 0091b01e7074ba8c3cde57bc6c1b5d00@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@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@10.226.39.49:5060
From: "199"<sip:199@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@10.226.39.49>;tag=as5911872b
To: <sip:10.226.2.2>
Contact: <sip:199@10.226.39.49:5060>
Call-ID: 5f618a412fe41f8d08abd036614d6608@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@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@10.226.39.49:5060
From: "199"<sip:199@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@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@10.18.5.64
From: "75666668"<sip:75666668@10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
To: "199"<sip:199@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@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@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@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@10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
To: "199"<sip:199@10.226.39.49;transport=udp;user=phone>
Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub@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@10.226.39.49:5060>
Content-Length: 0


<------------>
   -- Executing [199@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@10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
To: "199"<sip:199@10.226.39.49;transport=udp;user=phone>;tag=as0e4697cc
Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub@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@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@10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
To: "199"<sip:199@10.226.39.49;transport=udp;user=phone>;tag=as0e4697cc
Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub@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@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@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@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@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@10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
To: "199"<sip:199@10.226.39.49;transport=udp;user=phone>;tag=as0e4697cc
Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub@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@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@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@10.18.5.64
From: "75666668"<sip:75666668@10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
To: "199"<sip:199@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@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@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@10.226.39.49;transport=udp;user=phone>;tag=as0e4697cc
To: "75666668"<sip:75666668@10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub@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@10.18.5.64' in 6400 ms (Method: ACK)

<--- SIP read from UDP:10.226.2.2:5066 --->
INVITE sip:199@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@10.18.5.64
From: "75666668"<sip:75666668@10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
To: "199"<sip:199@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@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@10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
To: "199"<sip:199@10.226.39.49;transport=udp;user=phone>;tag=as0e4697cc
Call-ID: 0x0v2yxvybqxy4xxjbwwaujvxv2quyub@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@10.18.5.64' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:10.226.2.2:5066 --->
ACK sip:199@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@10.18.5.64
From: "75666668"<sip:75666668@10.226.2.2;transport=udp;user=phone>;tag=kbkxjy42-CC-1013-OFC-11
To: "199"<sip:199@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@10.18.5.64
From: "199"<sip:199@10.226.27.44:65476;transport=udp;user=phone>;tag=as0e4697cc
To: "75666668"<sip:75666668@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@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@10.226.39.49>;tag=as59c44cb0
To: <sip:10.226.2.10>
Contact: <sip:199@10.226.39.49:5060>
Call-ID: 0c78cfcf0a27badb28fc9f901272b01a@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@10.226.39.49>;tag=as275c8aa4
To: <sip:10.226.2.2>
Contact: <sip:199@10.226.39.49:5060>
Call-ID: 31097ba07c0dbb06175ced730b069255@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



Comments:By: Asterisk Team (asteriskteam) 2021-08-25 01:51:26.706-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.

Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/].
By: Joshua C. Colp (jcolp) 2021-08-25 03:48:01.466-0500

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