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 01:51:22 | Date Closed: | 2021-08-25 03:48:14 |
Priority: | Minor | Regression? | |
Status: | Closed/Complete | Components: | . I did not set the category correctly. |
Versions: | 18.5.0 | Frequency of Occurrence | |
Related Issues: | |||
Environment: | CENTOS7 | Attachments: | |
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 |