<--- SIP read from UDP:192.168.100.15:55030 --->

<------------->
Reliably Transmitting (NAT) to 41.106.54.27:5060:
OPTIONS sip:41.106.54.27 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK08988038;rport
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as3f4637da
To: <sip:41.106.54.27>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: IPBX-2.12.0.0(18.19.0)
Date: Tue, 26 Nov 2024 13:17:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:41.106.54.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.21:5060;received=192.168.100.21;rport=5060;branch=z9hG4bK08988038
To: <sip:41.106.54.27>;tag=h7g4Esbg_gr9s882y9uefpremvihyd5de4p52bj4r
From: "Unknown" <sip:[email protected]:5060>;tag=as3f4637da
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Supported: timer
Supported: 100rel
Supported: path
Supported: replaces
Content-Length: 0
Allow: INVITE
Allow: ACK
Allow: CANCEL
Allow: OPTIONS
Allow: REGISTER
Allow: BYE
Allow: INFO
Allow: REFER
Allow: UPDATE
Allow: SUBSCRIBE
Allow: MESSAGE
Allow: PUBLISH
Allow: NOTIFY
Allow: PRACK
Accept: application/sdp
Accept-Encoding:
Accept-Language: en

<------------->
--- (28 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.100.15:55030 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.15:55030;rport;branch=z9hG4bKPjd1c2f6e3d25640f187b350122015b41a
Max-Forwards: 70
From: "101" <sip:[email protected]>;tag=6a5c12fa64a14f6d8cb3777792afe60d
To: <sip:[email protected]>
Contact: "101" <sip:[email protected]:55030;ob>
Call-ID: 8cfcc0f72f564295946e7dc7cc0908ec
CSeq: 2360 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.21.5
Content-Type: application/sdp
Content-Length: 345

v=0
o=- 3941619437 3941619437 IN IP4 192.168.100.15
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4014 RTP/AVP 8 0 101
c=IN IP4 192.168.100.15
b=TIAS:64000
a=rtcp:4015 IN IP4 192.168.100.15
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:394862871 cname:0ff7656c15a40226
<------------->
--- (15 headers 16 lines) ---
Sending to 192.168.100.15:55030 (NAT)
Sending to 192.168.100.15:55030 (NAT)
Using INVITE request as basis request - 8cfcc0f72f564295946e7dc7cc0908ec
Found peer '101' for '101' from 192.168.100.15:55030

<--- Reliably Transmitting (NAT) to 192.168.100.15:55030 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.15:55030;branch=z9hG4bKPjd1c2f6e3d25640f187b350122015b41a;received=192.168.100.15;rport=55030
From: "101" <sip:[email protected]>;tag=6a5c12fa64a14f6d8cb3777792afe60d
To: <sip:[email protected]>;tag=as1d3c31b1
Call-ID: 8cfcc0f72f564295946e7dc7cc0908ec
CSeq: 2360 INVITE
Server: IPBX-2.12.0.0(18.19.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4441b96e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '8cfcc0f72f564295946e7dc7cc0908ec' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.100.15:55030 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.15:55030;rport;branch=z9hG4bKPjd1c2f6e3d25640f187b350122015b41a
Max-Forwards: 70
From: "101" <sip:[email protected]>;tag=6a5c12fa64a14f6d8cb3777792afe60d
To: <sip:[email protected]>;tag=as1d3c31b1
Call-ID: 8cfcc0f72f564295946e7dc7cc0908ec
CSeq: 2360 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.100.15:55030 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.15:55030;rport;branch=z9hG4bKPj141a1a1103774a8786b6316448378b83
Max-Forwards: 70
From: "101" <sip:[email protected]>;tag=6a5c12fa64a14f6d8cb3777792afe60d
To: <sip:[email protected]>
Contact: "101" <sip:[email protected]:55030;ob>
Call-ID: 8cfcc0f72f564295946e7dc7cc0908ec
CSeq: 2361 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.21.5
Authorization: Digest username="101", realm="asterisk", nonce="4441b96e", uri="sip:[email protected]", response="9aec5091881e07153c602a50d01686b3", algorithm=MD5
Content-Type: application/sdp
Content-Length: 345

v=0
o=- 3941619437 3941619437 IN IP4 192.168.100.15
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4014 RTP/AVP 8 0 101
c=IN IP4 192.168.100.15
b=TIAS:64000
a=rtcp:4015 IN IP4 192.168.100.15
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:394862871 cname:0ff7656c15a40226
<------------->
--- (16 headers 16 lines) ---
Sending to 192.168.100.15:55030 (NAT)
Using INVITE request as basis request - 8cfcc0f72f564295946e7dc7cc0908ec
Found peer '101' for '101' from 192.168.100.15:55030
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Got SDP version 3941619437 and unique parts [- 3941619437 IN IP4 192.168.100.15]
Found RTP audio format 8
Found RTP audio format 0
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 telephone-event for ID 101
Capabilities: us - (ulaw|alaw|silk16|silk12|g723|g729|g722), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.100.15:4014
Looking for +213775224664 in from-internal (domain 192.168.100.21)
sip_route_dump: route/path hop: <sip:[email protected]:55030;ob>

<--- Transmitting (NAT) to 192.168.100.15:55030 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.15:55030;branch=z9hG4bKPj141a1a1103774a8786b6316448378b83;received=192.168.100.15;rport=55030
From: "101" <sip:[email protected]>;tag=6a5c12fa64a14f6d8cb3777792afe60d
To: <sip:[email protected]>
Call-ID: 8cfcc0f72f564295946e7dc7cc0908ec
CSeq: 2361 INVITE
Server: IPBX-2.12.0.0(18.19.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
    -- Executing [+213775224664@from-internal:1] Macro("SIP/101-00000006", "user-callerid,LIMIT,EXTERNAL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/101-00000006", "TOUCH_MONITOR=1732627037.79") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/101-00000006", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/101-00000006", "0?trq") in new stack
    -- Executing [s@macro-user-callerid:4] Goto("SIP/101-00000006", "resume,1") in new stack
    -- Goto (macro-user-callerid,resume,1)
    -- Executing [+213775224664@from-internal:2] Set("SIP/101-00000006", "MOHCLASS=default") in new stack
    -- Executing [+213775224664@from-internal:3] Set("SIP/101-00000006", "_NODEST=") in new stack
    -- Executing [+213775224664@from-internal:4] Gosub("SIP/101-00000006", "sub-record-check,s,1(out,+213775224664,)") in new stack
    -- Executing [s@sub-record-check:1] Set("SIP/101-00000006", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:2] GotoIf("SIP/101-00000006", "1?check") in new stack
    -- Goto (sub-record-check,s,6)
    -- Executing [s@sub-record-check:6] Set("SIP/101-00000006", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:7] GotoIf("SIP/101-00000006", "1?next") in new stack
    -- Goto (sub-record-check,s,10)
    -- Executing [s@sub-record-check:10] ExecIf("SIP/101-00000006", "0?Return()") in new stack
    -- Executing [s@sub-record-check:11] ExecIf("SIP/101-00000006", "0?Set(__REC_POLICY_MODE=)") in new stack
    -- Executing [s@sub-record-check:12] GotoIf("SIP/101-00000006", "0?out,1") in new stack
    -- Executing [s@sub-record-check:13] Set("SIP/101-00000006", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:14] Set("SIP/101-00000006", "NOW=1732627037") in new stack
    -- Executing [s@sub-record-check:15] Set("SIP/101-00000006", "__DAY=26") in new stack
    -- Executing [s@sub-record-check:16] Set("SIP/101-00000006", "__MONTH=11") in new stack
    -- Executing [s@sub-record-check:17] Set("SIP/101-00000006", "__YEAR=2024") in new stack
    -- Executing [s@sub-record-check:18] Set("SIP/101-00000006", "__TIMESTR=20241126-141717") in new stack
    -- Executing [s@sub-record-check:19] Set("SIP/101-00000006", "__FROMEXTEN=101") in new stack
    -- Executing [s@sub-record-check:20] Set("SIP/101-00000006", "__CALLFILENAME=out-+213775224664-101-20241126-141717-1732627037.79") in new stack
    -- Executing [s@sub-record-check:21] Goto("SIP/101-00000006", "out,1") in new stack
    -- Goto (sub-record-check,out,1)
    -- Executing [out@sub-record-check:1] ExecIf("SIP/101-00000006", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
    -- Executing [out@sub-record-check:2] GosubIf("SIP/101-00000006", "0?record,1(exten,+213775224664,101)") in new stack
    -- Executing [out@sub-record-check:3] Return("SIP/101-00000006", "") in new stack
    -- Executing [+213775224664@from-internal:5] Macro("SIP/101-00000006", "dialout-trunk,1,+213775224664,,off") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/101-00000006", "DIAL_TRUNK=1") in new stack
    -- Executing [s@macro-dialout-trunk:2] AGI("SIP/101-00000006", "trunkbalance.php,1,+213775224664") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/trunkbalance.php
    -- trunkbalance.php,1,+213775224664: Dialed digits: +213775224664
 trunkbalance.php,1,+213775224664: SEVERE: Unknown database type: mysqli
 trunkbalance.php,1,+213775224664: SEVERE: Unable to connect to database.
    -- trunkbalance.php,1,+213775224664: No balancing rules are defined for this trunk
    -- <SIP/101-00000006>AGI Script trunkbalance.php completed, returning 0
    -- Executing [s@macro-dialout-trunk:3] ExecIf("SIP/101-00000006", "0?Set(DIAL_OPTIONS=tr)") in new stack
    -- Executing [s@macro-dialout-trunk:4] GosubIf("SIP/101-00000006", "0?sub-pincheck,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:5] GotoIf("SIP/101-00000006", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/101-00000006", "DIAL_NUMBER=+213775224664") in new stack
    -- Executing [s@macro-dialout-trunk:7] Set("SIP/101-00000006", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:8] Set("SIP/101-00000006", "OUTBOUND_GROUP=OUT_1") in new stack
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/101-00000006", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,11)
    -- Executing [s@macro-dialout-trunk:11] GotoIf("SIP/101-00000006", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:12] Set("SIP/101-00000006", "DIAL_TRUNK_OPTIONS=T") in new stack
    -- Executing [s@macro-dialout-trunk:13] Macro("SIP/101-00000006", "outbound-callerid,1") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/101-00000006", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/101-00000006", "1?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/101-00000006", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/101-00000006", "USEROUTCID="213982320375" <+213982320375>") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/101-00000006", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/101-00000006", "TRUNKOUTCID=+213982320375") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/101-00000006", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,14)
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/101-00000006", "1?Set(CALLERID(all)=+213982320375)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/101-00000006", "1?Set(CALLERID(all)="213982320375" <+213982320375>)") in new stack
    -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/101-00000006", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/101-00000006", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:18] Set("SIP/101-00000006", "CDR(outbound_cnum)=+213982320375") in new stack
    -- Executing [s@macro-outbound-callerid:19] Set("SIP/101-00000006", "CDR(outbound_cnam)=213982320375") in new stack
    -- Executing [s@macro-dialout-trunk:14] GosubIf("SIP/101-00000006", "0?sub-flp-1,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:15] Set("SIP/101-00000006", "OUTNUM=+213775224664") in new stack
    -- Executing [s@macro-dialout-trunk:16] Set("SIP/101-00000006", "custom=SIP/213982320375") in new stack
    -- Executing [s@macro-dialout-trunk:17] ExecIf("SIP/101-00000006", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
    -- Executing [s@macro-dialout-trunk:18] ExecIf("SIP/101-00000006", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:19] Macro("SIP/101-00000006", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/101-00000006", "") in new stack
    -- Executing [s@macro-dialout-trunk:20] GotoIf("SIP/101-00000006", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:21] ExecIf("SIP/101-00000006", "1?Set(CONNECTEDLINE(num,i)=+213775224664)") in new stack
    -- Executing [s@macro-dialout-trunk:22] ExecIf("SIP/101-00000006", "1?Set(CONNECTEDLINE(name,i)=CID:+213982320375)") in new stack
    -- Executing [s@macro-dialout-trunk:23] GotoIf("SIP/101-00000006", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:24] ExecIf("SIP/101-00000006", "0?Set(DIAL_TRUNK_OPTIONS=)") in new stack
    -- Executing [s@macro-dialout-trunk:25] Set("SIP/101-00000006", "DIALSTR=SIP/213982320375/+213775224664") in new stack
    -- Executing [s@macro-dialout-trunk:26] GosubIf("SIP/101-00000006", "0?pjsipdial,1()") in new stack
    -- Executing [s@macro-dialout-trunk:27] Dial("SIP/101-00000006", "SIP/213982320375/+213775224664,300,T") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 14788
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 41.106.54.27:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK6d659bf4;rport
Max-Forwards: 70
From: "213982320375" <sip:[email protected]>;tag=as29db2f77
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: IPBX-2.12.0.0(18.19.0)
Date: Tue, 26 Nov 2024 13:17:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 1575777243 1575777243 IN IP4 192.168.100.21
s=Asterisk PBX 18.19.0
c=IN IP4 192.168.100.21
t=0 0
m=audio 14788 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=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/213982320375/+213775224664
Retransmitting #1 (NAT) to 41.106.54.27:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK6d659bf4;rport
Max-Forwards: 70
From: "213982320375" <sip:[email protected]>;tag=as29db2f77
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: IPBX-2.12.0.0(18.19.0)
Date: Tue, 26 Nov 2024 13:17:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 1575777243 1575777243 IN IP4 192.168.100.21
s=Asterisk PBX 18.19.0
c=IN IP4 192.168.100.21
t=0 0
m=audio 14788 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=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:41.106.54.27:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.21:5060;received=192.168.100.21;rport=5060;branch=z9hG4bK6d659bf4
To: <sip:[email protected]>
From: "213982320375" <sip:[email protected]>;tag=as29db2f77
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Reliably Transmitting (NAT) to 192.168.100.15:55030:
OPTIONS sip:[email protected]:55030;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK5b0c4d12;rport
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as5c63d1cb
To: <sip:[email protected]:55030;ob>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: IPBX-2.12.0.0(18.19.0)
Date: Tue, 26 Nov 2024 13:17:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.15:55030 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.21:5060;rport=5060;received=192.168.100.21;branch=z9hG4bK5b0c4d12
Call-ID: [email protected]:5060
From: "Unknown" <sip:[email protected]>;tag=as5c63d1cb
To: <sip:[email protected];ob>;tag=z9hG4bK5b0c4d12
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub, trickle-ice
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.21.5
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.100.15:55030 --->

<------------->
[2024-11-26 14:17:31] NOTICE[11556]: chan_sip.c:15889 sip_reregister:    -- Re-registration for  [email protected]
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 41.106.54.27:5060:
REGISTER sip:voip.ims.algerietelecom.dz SIP/2.0
Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK757bfd72;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as2cf459e9
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 110 REGISTER
Supported: replaces, timer
User-Agent: IPBX-2.12.0.0(18.19.0)
Authorization: Digest username="213982320375", realm="ims.mnc009.mcc603.3gppnetwork.org", algorithm=MD5, uri="sip:voip.ims.algerietelecom.dz", nonce="7323997597C74567000000002CBC040C", response="b2d85c4e7e35d92cb35e0e2823821c0a", qop=auth, cnonce="012d4bec", nc=00000008
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0


---

<--- SIP read from UDP:41.106.54.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.21:5060;received=192.168.100.21;rport=5060;branch=z9hG4bK757bfd72
To: <sip:[email protected]>;tag=vs4itl1ufdr2nni2oeswhcfw9rftdxse
From: <sip:[email protected]>;tag=as2cf459e9
Call-ID: [email protected]
CSeq: 110 REGISTER
Contact: <sip:[email protected]:5060>;expires=120
P-Associated-Uri: <sip:[email protected]>
P-Associated-Uri: <sip:[email protected]>
P-Associated-Uri: <tel:+213982320375>
P-Associated-Uri: <sip:[email protected]>
Supported: replaces
Supported: timer
User-Agent: IPBX-2.12.0.0(18.19.0)
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
[2024-11-26 14:17:31] NOTICE[11556]: chan_sip.c:24961 handle_response_register: Outbound Registration: Expiry for 41.106.54.27 is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog '[email protected]' Method: REGISTER

<--- SIP read from UDP:192.168.100.15:55030 --->

<------------->

<--- SIP read from UDP:41.106.54.27:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.21:5060;received=192.168.100.21;rport=5060;branch=z9hG4bK6d659bf4
To: <sip:[email protected]>;tag=h7g4Esbg_p65544t1732627038m313433c919611108s1_2903422615-1
From: "213982320375" <sip:[email protected]>;tag=as29db2f77
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:41.106.54.27;transport=udp;lr>
P-Early-Media: sendonly
Server: Ericsson MTAS - CXP9020729/8 R19H02
Supported: timer
Content-Type: application/sdp
Content-Length: 297

v=0
o=- 2142690152 2903422477 IN IP4 41.106.54.7
s=Asterisk PBX 18.19.0
c=IN IP4 41.106.54.7
b=AS:80
t=0 0
a=sendrecv
m=audio 3794 RTP/AVP 0 101
c=IN IP4 41.106.54.7
b=AS:80
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:40
<------------->
--- (13 headers 16 lines) ---
sip_route_dump: route/path hop: <sip:41.106.54.27;transport=udp;lr>
Got SDP version 2903422477 and unique parts [- 2142690152 IN IP4 41.106.54.7]
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|alaw|g729), 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|)
Peer audio RTP is at port 41.106.54.7:3794
    -- SIP/213982320375-00000007 is making progress passing it to SIP/101-00000006
Audio is at 14256
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 192.168.100.15:55030 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.15:55030;branch=z9hG4bKPj141a1a1103774a8786b6316448378b83;received=192.168.100.15;rport=55030
From: "101" <sip:[email protected]>;tag=6a5c12fa64a14f6d8cb3777792afe60d
To: <sip:[email protected]>;tag=as3996fafd
Call-ID: 8cfcc0f72f564295946e7dc7cc0908ec
CSeq: 2361 INVITE
Server: IPBX-2.12.0.0(18.19.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 278

v=0
o=root 293989085 293989085 IN IP4 192.168.100.21
s=Asterisk PBX 18.19.0
c=IN IP4 192.168.100.21
t=0 0
m=audio 14256 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=ptime:20
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:192.168.100.15:55030 --->
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.15:55030;rport;branch=z9hG4bKPj141a1a1103774a8786b6316448378b83
Max-Forwards: 70
From: "101" <sip:[email protected]>;tag=6a5c12fa64a14f6d8cb3777792afe60d
To: <sip:[email protected]>
Call-ID: 8cfcc0f72f564295946e7dc7cc0908ec
CSeq: 2361 CANCEL
User-Agent: MicroSIP/3.21.5
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.100.15:55030 (NAT)

<--- Reliably Transmitting (NAT) to 192.168.100.15:55030 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.100.15:55030;branch=z9hG4bKPj141a1a1103774a8786b6316448378b83;received=192.168.100.15;rport=55030
From: "101" <sip:[email protected]>;tag=6a5c12fa64a14f6d8cb3777792afe60d
To: <sip:[email protected]>;tag=as3996fafd
Call-ID: 8cfcc0f72f564295946e7dc7cc0908ec
CSeq: 2361 INVITE
Server: IPBX-2.12.0.0(18.19.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 192.168.100.15:55030 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.15:55030;branch=z9hG4bKPj141a1a1103774a8786b6316448378b83;received=192.168.100.15;rport=55030
From: "101" <sip:[email protected]>;tag=6a5c12fa64a14f6d8cb3777792afe60d
To: <sip:[email protected]>;tag=as3996fafd
Call-ID: 8cfcc0f72f564295946e7dc7cc0908ec
CSeq: 2361 CANCEL
Server: IPBX-2.12.0.0(18.19.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 41.106.54.27:5060:
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK6d659bf4;rport
Max-Forwards: 70
From: "213982320375" <sip:[email protected]>;tag=as29db2f77
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 CANCEL
User-Agent: IPBX-2.12.0.0(18.19.0)
Content-Length: 0


---
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
  == Spawn extension (macro-dialout-trunk, s, 27) exited non-zero on 'SIP/101-00000006' in macro 'dialout-trunk'
  == Spawn extension (from-internal, +213775224664, 5) exited non-zero on 'SIP/101-00000006'
    -- Executing [h@from-internal:1] Hangup("SIP/101-00000006", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000006'

<--- SIP read from UDP:192.168.100.15:55030 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.15:55030;rport;branch=z9hG4bKPj141a1a1103774a8786b6316448378b83
Max-Forwards: 70
From: "101" <sip:[email protected]>;tag=6a5c12fa64a14f6d8cb3777792afe60d
To: <sip:[email protected]>;tag=as3996fafd
Call-ID: 8cfcc0f72f564295946e7dc7cc0908ec
CSeq: 2361 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '8cfcc0f72f564295946e7dc7cc0908ec' Method: ACK

<--- SIP read from UDP:41.106.54.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.21:5060;received=192.168.100.21;rport=5060;branch=z9hG4bK6d659bf4
To: <sip:[email protected]>;tag=h7g4Esbg_p65544t1732627038m313433c919611108s1_2903422615-1
From: "213982320375" <sip:[email protected]>;tag=as29db2f77
Call-ID: [email protected]
CSeq: 102 CANCEL
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:41.106.54.27:5060 --->
SIP/2.0 408 Request Timeout 040351913
Via: SIP/2.0/UDP 192.168.100.21:5060;received=192.168.100.21;rport=5060;branch=z9hG4bK6d659bf4
To: <sip:[email protected]>;tag=h7g4Esbg_p65544t1732627038m313433c919611108s1_2871369884-1731184383
From: "213982320375" <sip:[email protected]>;tag=as29db2f77
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected];transport=udp>
Server: Ericsson MTAS - CXP9020729/8 R19H02
Supported: timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 41.106.54.27:5060:
ACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK6d659bf4;rport
Max-Forwards: 70
From: "213982320375" <sip:[email protected]>;tag=as29db2f77
To: <sip:[email protected]>;tag=h7g4Esbg_p65544t1732627038m313433c919611108s1_2871369884-1731184383
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: IPBX-2.12.0.0(18.19.0)
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.15:55030 --->