I recently migrated from my DAHDI system to a VM on UnRaid. Everything runs smooth with asterisk 18.X and almost perfect expect one weird oddity.
I can dial out every local US number in the world except one particular number *****8840 as you can see in the logs. Is the weirdest thing. I can dial *****7000 from the same extension but it will not work for the *****8840. I tried from all my local extensions and they all behave the same. I can dial every US number except that one?
Any ideas what I am doing wrong here? I attached the debug log for a the working version and not working version.
I would really appreciate the help and guidance as my fluency in asterisk is not where it needs to be :)
VM-HOME-PBX*CLI> VM-HOME-PBX*CLI> VM-HOME-PBX*CLI> VM-HOME-PBX*CLI> sip set debug on SIP Debugging enabled
<--- SIP read from UDP:192.168.0.33:51640 --->
INVITE sip:7137767000@192.168.0.15;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bKb3187b91
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a312900a5594dae49-e836d22c
To: <sip:7137767000@192.168.0.15>
Call-ID: d0574c6a-31290018-c5c71753-afacaace@192.168.0.33
Max-Forwards: 70
Date: Thu, 12 Sep 2024 23:15:34 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7965G/9.3.1
Contact: <sip:200@192.168.0.33:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported:
replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 352
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 27458 0 IN IP4 192.168.0.33
s=SIP Call
t=0 0
m=audio 16388 RTP/AVP 0 8 18 102 116 101
c=IN IP4 192.168.0.33
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (18 headers 16 lines) ---
Sending to 192.168.0.33:5060 (no NAT)
Sending to 192.168.0.33:5060 (no NAT)
Using INVITE request as basis request -
d0574c6a-31290018-c5c71753-afacaace@192.168.0.33
Found peer '200' for '200' from 192.168.0.33:51640
<--- Reliably Transmitting (no NAT) to 192.168.0.33:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bKb3187b91;received=192.168.0.33
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a312900a5594dae49-e836d22c
To: <sip:7137767000@192.168.0.15>;tag=as6d6a722e
Call-ID: d0574c6a-31290018-c5c71753-afacaace@192.168.0.33
CSeq: 101 INVITE
Server: Asterisk PBX 18.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7c348cd1"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'd0574c6a-31290018-c5c71753-afacaace@192.168.0.33' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:192.168.0.33:50062 --->
ACK sip:7137767000@192.168.0.15;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bKb3187b91
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a312900a5594dae49-e836d22c
To: <sip:7137767000@192.168.0.15>;tag=as6d6a722e
Call-ID: d0574c6a-31290018-c5c71753-afacaace@192.168.0.33
Max-Forwards: 70
Date: Thu, 12 Sep 2024 23:15:34 GMT
CSeq: 101 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:192.168.0.33:51640 --->
INVITE sip:7137767000@192.168.0.15;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bK0e1c3c1d
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a312900a5594dae49-e836d22c
To: <sip:7137767000@192.168.0.15>
Call-ID: d0574c6a-31290018-c5c71753-afacaace@192.168.0.33
Max-Forwards: 70
Date: Thu, 12 Sep 2024 23:15:34 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7965G/9.3.1
Contact: <sip:200@192.168.0.33:5060;transport=udp>
Authorization: Digest
username="200",realm="asterisk",uri="sip:7137767000@192.168.0.15;user=phone",response="f169192c259509d8a04fff6d44e1550d",nonce="7c348cd1",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported:
replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 352
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 27458 0 IN IP4 192.168.0.33
s=SIP Call
t=0 0
m=audio 16388 RTP/AVP 0 8 18 102 116 101
c=IN IP4 192.168.0.33
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (19 headers 16 lines) ---
Sending to 192.168.0.33:5060 (no NAT)
Using INVITE request as basis request -
d0574c6a-31290018-c5c71753-afacaace@192.168.0.33
Found peer '200' for '200' from 192.168.0.33:51640
[Sep 12 18:15:35] ERROR[1119][C-00000015]: netsock2.c:303 ast_sockaddr_resolve:
getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname
[Sep 12 18:15:35] WARNING[1119][C-00000015]: acl.c:890 resolve_first: Unable to
lookup 'VM-HOME-PBX'
== Using SIP RTP CoS mark 5
Got SDP version 0 and unique parts [Cisco-SIPUA 27458 IN IP4 192.168.0.33]
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 102
Found RTP audio format 116
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format L16 for ID 102
Found audio description format iLBC for ID 116
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer -
audio=(ulaw|alaw|g729|slin16|ilbc)/video=(nothing)/text=(nothing), combined -
(ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
> 0x1494e4042370 -- Strict RTP learning after remote address set to:
192.168.0.33:16388
Peer audio RTP is at port 192.168.0.33:16388
Looking for 7137767000 in voip (domain 192.168.0.15)
sip_route_dump: route/path hop: <sip:200@192.168.0.33:5060;transport=udp>
<--- Transmitting (no NAT) to 192.168.0.33:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bK0e1c3c1d;received=192.168.0.33
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a312900a5594dae49-e836d22c
To: <sip:7137767000@192.168.0.15>
Call-ID: d0574c6a-31290018-c5c71753-afacaace@192.168.0.33
CSeq: 102 INVITE
Server: Asterisk PBX 18.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:7137767000@192.168.0.15:5060>
Content-Length: 0
<------------>
-- Executing [7137767000@voip:1] MixMonitor("SIP/200-0000002d", "200-Ionel
Chila-20240912-181535-1726182935.101.wav") in new stack
-- Executing [7137767000@voip:2] Dial("SIP/200-0000002d",
"SIP/vitel-outbound/7137767000,90,tr") in new stack
== Begin MixMonitor Recording SIP/200-0000002d
[Sep 12 18:15:35] ERROR[1929][C-00000015]: netsock2.c:303 ast_sockaddr_resolve:
getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname
[Sep 12 18:15:35] WARNING[1929][C-00000015]: acl.c:890 resolve_first: Unable to
lookup 'VM-HOME-PBX'
== Using SIP RTP CoS mark 5
Audio is at 10466
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 64.2.142.189:5060:
INVITE sip:7137767...@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.15:5060;branch=z9hG4bK676350e4;rport
Max-Forwards: 70
From: "Ionel Chila" <sip:ione_chila@192.168.0.15>;tag=as3457e8a8
To: <sip:7137767...@outbound.vitelity.net>
Contact: <sip:ione_chila@192.168.0.15:5060>
Call-ID: 24ff3957072fe9c34620a683334c0799@192.168.0.15:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.22.0
Date: Thu, 12 Sep 2024 23:15:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Ionel Chila"
<sip:200@192.168.0.15>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 551255825 551255825 IN IP4 192.168.0.15
s=Asterisk PBX 18.22.0
c=IN IP4 192.168.0.15
t=0 0
m=audio 10466 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:140
a=sendrecv
---
-- Called SIP/vitel-outbound/7137767000
<--- Transmitting (no NAT) to 192.168.0.33:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bK0e1c3c1d;received=192.168.0.33
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a312900a5594dae49-e836d22c
To: <sip:7137767000@192.168.0.15>;tag=as18305159
Call-ID: d0574c6a-31290018-c5c71753-afacaace@192.168.0.33
CSeq: 102 INVITE
Server: Asterisk PBX 18.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:7137767000@192.168.0.15:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.15:5060;received=192.168.0.15;branch=z9hG4bK676350e4;rport=5060
From: "Ionel Chila" <sip:ione_chila@192.168.0.15:5060>;tag=as3457e8a8
To: <sip:7137767...@outbound.vitelity.net>
Call-ID: 24ff3957072fe9c34620a683334c0799@192.168.0.15:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.0.15:5060;received=192.168.0.15;branch=z9hG4bK676350e4;rport=5060
From: "Ionel Chila" <sip:ione_chila@192.168.0.15:5060>;tag=as3457e8a8
To: <sip:7137767...@outbound.vitelity.net>;tag=as5b88a39f
Call-ID: 24ff3957072fe9c34620a683334c0799@192.168.0.15:5060
CSeq: 102 INVITE
Server: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="560b2177"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 64.2.142.189:5060:
ACK sip:7137767...@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.15:5060;branch=z9hG4bK676350e4;rport
Max-Forwards: 70
From: "Ionel Chila" <sip:ione_chila@192.168.0.15>;tag=as3457e8a8
To: <sip:7137767...@outbound.vitelity.net>;tag=as5b88a39f
Contact: <sip:ione_chila@192.168.0.15:5060>
Call-ID: 24ff3957072fe9c34620a683334c0799@192.168.0.15:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 18.22.0
Content-Length: 0
---
Audio is at 10466
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 64.2.142.189:5060:
INVITE sip:7137767...@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.15:5060;branch=z9hG4bK1ea0a1b9;rport
Max-Forwards: 70
From: "Ionel Chila" <sip:ione_chila@192.168.0.15>;tag=as3457e8a8
To: <sip:7137767...@outbound.vitelity.net>
Contact: <sip:ione_chila@192.168.0.15:5060>
Call-ID: 24ff3957072fe9c34620a683334c0799@192.168.0.15:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 18.22.0
Authorization: Digest username="ione_chila", realm="asterisk", algorithm=MD5,
uri="sip:7137767...@outbound.vitelity.net", nonce="560b2177",
response="0b760f0526b547df29e04690cf3fae78"
Date: Thu, 12 Sep 2024 23:15:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Ionel Chila"
<sip:200@192.168.0.15>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 551255825 551255826 IN IP4 192.168.0.15
s=Asterisk PBX 18.22.0
c=IN IP4 192.168.0.15
t=0 0
m=audio 10466 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:140
a=sendrecv
---
<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.15:5060;received=192.168.0.15;branch=z9hG4bK1ea0a1b9;rport=5060
From: "Ionel Chila" <sip:ione_chila@192.168.0.15:5060>;tag=as3457e8a8
To: <sip:7137767...@outbound.vitelity.net>
Call-ID: 24ff3957072fe9c34620a683334c0799@192.168.0.15:5060
CSeq: 103 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.15:5060;received=192.168.0.15;branch=z9hG4bK1ea0a1b9;rport=5060
From: "Ionel Chila" <sip:ione_chila@192.168.0.15:5060>;tag=as3457e8a8
To: <sip:7137767...@outbound.vitelity.net>;tag=as0544fad3
Call-ID: 24ff3957072fe9c34620a683334c0799@192.168.0.15:5060
CSeq: 103 INVITE
Server: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:7137767000@64.2.142.189:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 249
v=0
o=root 291698510 291698510 IN IP4 64.2.142.189
s=Asterisk PBX 16.8.0
c=IN IP4 64.2.142.189
t=0 0
m=audio 35896 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
a=ptime:20
<------------->
--- (12 headers 12 lines) ---
Got SDP version 291698510 and unique parts [root 291698510 IN IP4 64.2.142.189]
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
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing),
combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
> 0x1494fc0079a0 -- Strict RTP learning after remote address set to:
64.2.142.189:35896
Peer audio RTP is at port 64.2.142.189:35896
sip_route_dump: route/path hop: <sip:7137767000@64.2.142.189:5060;transport=udp>
Transmitting (NAT) to 64.2.142.189:5060:
ACK sip:7137767000@64.2.142.189:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.15:5060;branch=z9hG4bK156664fd;rport
Max-Forwards: 70
From: "Ionel Chila" <sip:ione_chila@192.168.0.15>;tag=as3457e8a8
To: <sip:7137767...@outbound.vitelity.net>;tag=as0544fad3
Contact: <sip:ione_chila@192.168.0.15:5060>
Call-ID: 24ff3957072fe9c34620a683334c0799@192.168.0.15:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 18.22.0
Content-Length: 0
---
-- SIP/vitel-outbound-0000002e answered SIP/200-0000002d
Audio is at 19382
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 192.168.0.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bK0e1c3c1d;received=192.168.0.33
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a312900a5594dae49-e836d22c
To: <sip:7137767000@192.168.0.15>;tag=as18305159
Call-ID: d0574c6a-31290018-c5c71753-afacaace@192.168.0.33
CSeq: 102 INVITE
Server: Asterisk PBX 18.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:7137767000@192.168.0.15:5060>
Content-Type: application/sdp
Content-Length: 264
v=0
o=root 1130024551 1130024551 IN IP4 192.168.0.15
s=Asterisk PBX 18.22.0
c=IN IP4 192.168.0.15
t=0 0
m=audio 19382 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:140
a=sendrecv
<------------>
-- Channel SIP/vitel-outbound-0000002e joined 'simple_bridge' basic-bridge
<34c61e2b-3191-416e-97f5-aac39f648ec5>
-- Channel SIP/200-0000002d joined 'simple_bridge' basic-bridge
<34c61e2b-3191-416e-97f5-aac39f648ec5>
> 0x1494e4042370 -- Strict RTP switching to RTP target address
192.168.0.33:16388 as source
<--- SIP read from UDP:192.168.0.33:51640 --->
ACK sip:7137767000@192.168.0.15:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bK63e0d179
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a312900a5594dae49-e836d22c
To: <sip:7137767000@192.168.0.15>;tag=as18305159
Call-ID: d0574c6a-31290018-c5c71753-afacaace@192.168.0.33
Max-Forwards: 70
Date: Thu, 12 Sep 2024 23:15:35 GMT
CSeq: 102 ACK
User-Agent: Cisco-CP7965G/9.3.1
Authorization: Digest
username="200",realm="asterisk",uri="sip:7137767000@192.168.0.15;user=phone",response="f169192c259509d8a04fff6d44e1550d",nonce="7c348cd1",algorithm=MD5
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
> 0x1494fc0079a0 -- Strict RTP switching to RTP target address
64.2.142.189:35896 as source
Really destroying SIP dialog
'a5566676-f2ca-481f-9f74-0c9c7d2d7161_2be73c3a02c23b17072ed101488be6a8@192.168.0.15'
Method: OPTIONS
> 0x1494e4042370 -- Strict RTP learning complete - Locking on source
address 192.168.0.33:16388
<--- SIP read from UDP:192.168.0.33:51640 --->
BYE sip:7137767000@192.168.0.15:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bKd7068255
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a312900a5594dae49-e836d22c
To: <sip:7137767000@192.168.0.15>;tag=as18305159
Call-ID: d0574c6a-31290018-c5c71753-afacaace@192.168.0.33
Max-Forwards: 70
Date: Thu, 12 Sep 2024 23:15:39 GMT
CSeq: 103 BYE
User-Agent: Cisco-CP7965G/9.3.1
Content-Length: 0
Authorization: Digest
username="200",realm="asterisk",uri="sip:7137767000@192.168.0.15:5060",response="8d59a11d6a24d798470c807805bd96e9",nonce="7c348cd1",algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.0.33:5060 (no NAT)
Scheduling destruction of SIP dialog
'd0574c6a-31290018-c5c71753-afacaace@192.168.0.33' in 6400 ms (Method: BYE)
<--- Transmitting (no NAT) to 192.168.0.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bKd7068255;received=192.168.0.33
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a312900a5594dae49-e836d22c
To: <sip:7137767000@192.168.0.15>;tag=as18305159
Call-ID: d0574c6a-31290018-c5c71753-afacaace@192.168.0.33
CSeq: 103 BYE
Server: Asterisk PBX 18.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
-- Channel SIP/200-0000002d left 'simple_bridge' basic-bridge
<34c61e2b-3191-416e-97f5-aac39f648ec5>
-- Channel SIP/vitel-outbound-0000002e left 'simple_bridge' basic-bridge
<34c61e2b-3191-416e-97f5-aac39f648ec5>
== Spawn extension (voip, 7137767000, 2) exited non-zero on 'SIP/200-0000002d'
Scheduling destruction of SIP dialog
'24ff3957072fe9c34620a683334c0799@192.168.0.15:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 64.2.142.189:5060:
BYE sip:7137767000@64.2.142.189:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.15:5060;branch=z9hG4bK1765256d;rport
Max-Forwards: 70
From: "Ionel Chila" <sip:ione_chila@192.168.0.15>;tag=as3457e8a8
To: <sip:7137767...@outbound.vitelity.net>;tag=as0544fad3
Call-ID: 24ff3957072fe9c34620a683334c0799@192.168.0.15:5060
CSeq: 104 BYE
User-Agent: Asterisk PBX 18.22.0
Authorization: Digest username="ione_chila", realm="asterisk", algorithm=MD5,
uri="sip:7137767000@64.2.142.189:5060", nonce="560b2177",
response="32965d36068db1bd76f8d9d09e7a94a6"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/200-0000002d
<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.15:5060;received=192.168.0.15;branch=z9hG4bK1765256d;rport=5060
From: "Ionel Chila" <sip:ione_chila@192.168.0.15:5060>;tag=as3457e8a8
To: <sip:7137767...@outbound.vitelity.net>;tag=as0544fad3
Call-ID: 24ff3957072fe9c34620a683334c0799@192.168.0.15:5060
CSeq: 104 BYE
Server: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog
'24ff3957072fe9c34620a683334c0799@192.168.0.15:5060' Method: INVITE
VM-HOME-PBX*CLI> sip set debug off
SIP Debugging Disabled
VM-HOME-PBX*CLI>
VM-HOME-PBX*CLI> sip set debug on
SIP Debugging enabled
VM-HOME-PBX*CLI>
VM-HOME-PBX*CLI>
VM-HOME-PBX*CLI>
VM-HOME-PBX*CLI>
<--- SIP read from UDP:192.168.0.33:51640 --->
INVITE sip:8322068840@192.168.0.15;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bK62d8abc0
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a3129009db5065bf2-e94af2bd
To: <sip:8322068840@192.168.0.15>
Call-ID: d0574c6a-31290016-8fd35654-ba96cb67@192.168.0.33
Max-Forwards: 70
Date: Thu, 12 Sep 2024 23:11:37 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7965G/9.3.1
Contact: <sip:200@192.168.0.33:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported:
replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 352
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 13156 0 IN IP4 192.168.0.33
s=SIP Call
t=0 0
m=audio 16384 RTP/AVP 0 8 18 102 116 101
c=IN IP4 192.168.0.33
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (18 headers 16 lines) ---
Sending to 192.168.0.33:5060 (no NAT)
Sending to 192.168.0.33:5060 (no NAT)
Using INVITE request as basis request -
d0574c6a-31290016-8fd35654-ba96cb67@192.168.0.33
Found peer '200' for '200' from 192.168.0.33:51640
<--- Reliably Transmitting (no NAT) to 192.168.0.33:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bK62d8abc0;received=192.168.0.33
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a3129009db5065bf2-e94af2bd
To: <sip:8322068840@192.168.0.15>;tag=as52e811a0
Call-ID: d0574c6a-31290016-8fd35654-ba96cb67@192.168.0.33
CSeq: 101 INVITE
Server: Asterisk PBX 18.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="03dc90f3"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'd0574c6a-31290016-8fd35654-ba96cb67@192.168.0.33' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:192.168.0.33:51920 --->
ACK sip:8322068840@192.168.0.15;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bK62d8abc0
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a3129009db5065bf2-e94af2bd
To: <sip:8322068840@192.168.0.15>;tag=as52e811a0
Call-ID: d0574c6a-31290016-8fd35654-ba96cb67@192.168.0.33
Max-Forwards: 70
Date: Thu, 12 Sep 2024 23:11:37 GMT
CSeq: 101 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:192.168.0.33:51640 --->
INVITE sip:8322068840@192.168.0.15;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bKe21eabb9
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a3129009db5065bf2-e94af2bd
To: <sip:8322068840@192.168.0.15>
Call-ID: d0574c6a-31290016-8fd35654-ba96cb67@192.168.0.33
Max-Forwards: 70
Date: Thu, 12 Sep 2024 23:11:37 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7965G/9.3.1
Contact: <sip:200@192.168.0.33:5060;transport=udp>
Authorization: Digest
username="200",realm="asterisk",uri="sip:8322068840@192.168.0.15;user=phone",response="dd2d216342cf5bbd9b7ce6dffa078192",nonce="03dc90f3",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported:
replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 352
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 13156 0 IN IP4 192.168.0.33
s=SIP Call
t=0 0
m=audio 16384 RTP/AVP 0 8 18 102 116 101
c=IN IP4 192.168.0.33
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (19 headers 16 lines) ---
Sending to 192.168.0.33:5060 (no NAT)
Using INVITE request as basis request -
d0574c6a-31290016-8fd35654-ba96cb67@192.168.0.33
Found peer '200' for '200' from 192.168.0.33:51640
[Sep 12 18:11:39] ERROR[1119][C-00000014]: netsock2.c:303 ast_sockaddr_resolve:
getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname
[Sep 12 18:11:39] WARNING[1119][C-00000014]: acl.c:890 resolve_first: Unable to
lookup 'VM-HOME-PBX'
== Using SIP RTP CoS mark 5
Got SDP version 0 and unique parts [Cisco-SIPUA 13156 IN IP4 192.168.0.33]
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 102
Found RTP audio format 116
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format L16 for ID 102
Found audio description format iLBC for ID 116
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer -
audio=(ulaw|alaw|g729|slin16|ilbc)/video=(nothing)/text=(nothing), combined -
(ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
> 0x1494e4042370 -- Strict RTP learning after remote address set to:
192.168.0.33:16384
Peer audio RTP is at port 192.168.0.33:16384
Looking for 8322068840 in voip (domain 192.168.0.15)
sip_route_dump: route/path hop: <sip:200@192.168.0.33:5060;transport=udp>
<--- Transmitting (no NAT) to 192.168.0.33:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bKe21eabb9;received=192.168.0.33
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a3129009db5065bf2-e94af2bd
To: <sip:8322068840@192.168.0.15>
Call-ID: d0574c6a-31290016-8fd35654-ba96cb67@192.168.0.33
CSeq: 102 INVITE
Server: Asterisk PBX 18.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:8322068840@192.168.0.15:5060>
Content-Length: 0
<------------>
-- Executing [8322068840@voip:1] MixMonitor("SIP/200-0000002b", "200-Ionel
Chila-20240912-181139-1726182699.97.wav") in new stack
== Begin MixMonitor Recording SIP/200-0000002b
-- Executing [8322068840@voip:2] Dial("SIP/200-0000002b",
"SIP/vitel-outbound/8322068840,90,tr") in new stack
[Sep 12 18:11:39] ERROR[1894][C-00000014]: netsock2.c:303 ast_sockaddr_resolve:
getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname
[Sep 12 18:11:39] WARNING[1894][C-00000014]: acl.c:890 resolve_first: Unable to
lookup 'VM-HOME-PBX'
== Using SIP RTP CoS mark 5
Audio is at 11262
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 64.2.142.189:5060:
INVITE sip:8322068...@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.15:5060;branch=z9hG4bK76a113f0;rport
Max-Forwards: 70
From: "Ionel Chila" <sip:ione_chila@192.168.0.15>;tag=as2901f58f
To: <sip:8322068...@outbound.vitelity.net>
Contact: <sip:ione_chila@192.168.0.15:5060>
Call-ID: 65f5b9b06347fe632fc516443135b8d9@192.168.0.15:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.22.0
Date: Thu, 12 Sep 2024 23:11:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Ionel Chila"
<sip:200@192.168.0.15>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 1300024001 1300024001 IN IP4 192.168.0.15
s=Asterisk PBX 18.22.0
c=IN IP4 192.168.0.15
t=0 0
m=audio 11262 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:140
a=sendrecv
---
-- Called SIP/vitel-outbound/8322068840
<--- Transmitting (no NAT) to 192.168.0.33:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bKe21eabb9;received=192.168.0.33
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a3129009db5065bf2-e94af2bd
To: <sip:8322068840@192.168.0.15>;tag=as7f165881
Call-ID: d0574c6a-31290016-8fd35654-ba96cb67@192.168.0.33
CSeq: 102 INVITE
Server: Asterisk PBX 18.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:8322068840@192.168.0.15:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.15:5060;received=192.168.0.15;branch=z9hG4bK76a113f0;rport=5060
From: "Ionel Chila" <sip:ione_chila@192.168.0.15:5060>;tag=as2901f58f
To: <sip:8322068...@outbound.vitelity.net>
Call-ID: 65f5b9b06347fe632fc516443135b8d9@192.168.0.15:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.0.15:5060;received=192.168.0.15;branch=z9hG4bK76a113f0;rport=5060
From: "Ionel Chila" <sip:ione_chila@192.168.0.15:5060>;tag=as2901f58f
To: <sip:8322068...@outbound.vitelity.net>;tag=as13bd1d91
Call-ID: 65f5b9b06347fe632fc516443135b8d9@192.168.0.15:5060
CSeq: 102 INVITE
Server: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4bc34554"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 64.2.142.189:5060:
ACK sip:8322068...@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.15:5060;branch=z9hG4bK76a113f0;rport
Max-Forwards: 70
From: "Ionel Chila" <sip:ione_chila@192.168.0.15>;tag=as2901f58f
To: <sip:8322068...@outbound.vitelity.net>;tag=as13bd1d91
Contact: <sip:ione_chila@192.168.0.15:5060>
Call-ID: 65f5b9b06347fe632fc516443135b8d9@192.168.0.15:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 18.22.0
Content-Length: 0
---
Audio is at 11262
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 64.2.142.189:5060:
INVITE sip:8322068...@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.15:5060;branch=z9hG4bK5da73f52;rport
Max-Forwards: 70
From: "Ionel Chila" <sip:ione_chila@192.168.0.15>;tag=as2901f58f
To: <sip:8322068...@outbound.vitelity.net>
Contact: <sip:ione_chila@192.168.0.15:5060>
Call-ID: 65f5b9b06347fe632fc516443135b8d9@192.168.0.15:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 18.22.0
Authorization: Digest username="ione_chila", realm="asterisk", algorithm=MD5,
uri="sip:8322068...@outbound.vitelity.net", nonce="4bc34554",
response="995b4d67afa22e6a811ba6b573708154"
Date: Thu, 12 Sep 2024 23:11:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Ionel Chila"
<sip:200@192.168.0.15>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 1300024001 1300024002 IN IP4 192.168.0.15
s=Asterisk PBX 18.22.0
c=IN IP4 192.168.0.15
t=0 0
m=audio 11262 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:140
a=sendrecv
---
<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.15:5060;received=192.168.0.15;branch=z9hG4bK5da73f52;rport=5060
From: "Ionel Chila" <sip:ione_chila@192.168.0.15:5060>;tag=as2901f58f
To: <sip:8322068...@outbound.vitelity.net>
Call-ID: 65f5b9b06347fe632fc516443135b8d9@192.168.0.15:5060
CSeq: 103 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP
192.168.0.15:5060;received=192.168.0.15;branch=z9hG4bK5da73f52;rport=5060
From: "Ionel Chila" <sip:ione_chila@192.168.0.15:5060>;tag=as2901f58f
To: <sip:8322068...@outbound.vitelity.net>;tag=as0d039007
Call-ID: 65f5b9b06347fe632fc516443135b8d9@192.168.0.15:5060
CSeq: 103 INVITE
Server: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 64.2.142.189:5060:
ACK sip:8322068...@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.15:5060;branch=z9hG4bK5da73f52;rport
Max-Forwards: 70
From: "Ionel Chila" <sip:ione_chila@192.168.0.15>;tag=as2901f58f
To: <sip:8322068...@outbound.vitelity.net>;tag=as0d039007
Contact: <sip:ione_chila@192.168.0.15:5060>
Call-ID: 65f5b9b06347fe632fc516443135b8d9@192.168.0.15:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 18.22.0
Content-Length: 0
---
-- SIP/vitel-outbound-0000002c redirecting info has changed, passing it to
SIP/200-0000002b
<--- Transmitting (no NAT) to 192.168.0.33:5060 --->
SIP/2.0 181 Call is being forwarded
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bKe21eabb9;received=192.168.0.33
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a3129009db5065bf2-e94af2bd
To: <sip:8322068840@192.168.0.15>;tag=as7f165881
Call-ID: d0574c6a-31290016-8fd35654-ba96cb67@192.168.0.33
CSeq: 102 INVITE
Server: Asterisk PBX 18.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:8322068840@192.168.0.15:5060>
Content-Length: 0
<------------>
-- SIP/vitel-outbound-0000002c is busy
Scheduling destruction of SIP dialog
'65f5b9b06347fe632fc516443135b8d9@192.168.0.15:5060' in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [8322068840@voip:3] Congestion("SIP/200-0000002b", "") in new
stack
<--- Reliably Transmitting (no NAT) to 192.168.0.33:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bKe21eabb9;received=192.168.0.33
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a3129009db5065bf2-e94af2bd
To: <sip:8322068840@192.168.0.15>;tag=as7f165881
Call-ID: d0574c6a-31290016-8fd35654-ba96cb67@192.168.0.33
CSeq: 102 INVITE
Server: Asterisk PBX 18.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0
<------------>
== Spawn extension (voip, 8322068840, 3) exited non-zero on 'SIP/200-0000002b'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/200-0000002b
<--- SIP read from UDP:192.168.0.33:52258 --->
ACK sip:8322068840@192.168.0.15;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bKe21eabb9
From: "Ionel Chila" <sip:200@192.168.0.15>;tag=d0574c6a3129009db5065bf2-e94af2bd
To: <sip:8322068840@192.168.0.15>;tag=as7f165881
Call-ID: d0574c6a-31290016-8fd35654-ba96cb67@192.168.0.33
Max-Forwards: 70
Date: Thu, 12 Sep 2024 23:11:40 GMT
CSeq: 102 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'd0574c6a-31290016-8fd35654-ba96cb67@192.168.0.33'
Method: ACK
VM-HOME-PBX*CLI> sip set debug off
SIP Debugging Disabled
VM-HOME-PBX*CLI>
_______________________________________________ Astlinux-users mailing list Astlinux-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org.
