[Issue] Second call on the Ip Phone not availible


#1

Morning @everyone,

I have configure my asterisk 13.23.0 and this is what I put in my extension.conf file for my internal call.

[from-internal]
;======================================================================= 
;							Internal Calls

exten => _1XX,1,NoOp(## Internal Call to the ${EXTEN} ##)
 same => n,Verbose(Call start time: ${CDR(start)})
 same => n,Set(CDR(calldate)=${CDR(start)})
 same => n,Set(CDR(useragent)=${CALLERID(name)})
 same => n,Set(NOW=${STRFTIME(${EPOCH},GMT+0,%Y%m%d_%H%M%S)})
 same => n,Set(REC_FILE_NAME=OUT_${NOW}_${EXTEN}_${POSTE}.wav)
 same => n,MixMonitor(${REC_FILE_NAME},b V(1))
 same => n,Dial(PJSIP/${EXTEN},20)
 same => n,VoiceMail(${EXTEN},su)
 same => n,Hangup()

But I can not have any second call on all my Ip Phone. When one of them is online and I call the same ip phone who is online this last one do not make any signal. I’m automaticaly redirect to the answerer.

Could you please tell me what’s happen ?

Best regard,
Lordaker.


#2

Not without both SIP and verbose Asterisk logs.


#3

Hi @david551, alright but I use pjsip protocol and this a part of my pjsip.conf file.

[100]
type=endpoint
context=from-internal
subscribe_context=phones-blf
dtmf_mode=rfc4733
disallow=all
allow=ulaw,alaw,gsm,g722,g729
transport=transport-udp
auth=100
aors=100
direct_media=no
mailboxes=100@default
trust_id_outbound=yes
callerid=Arnold <100>
device_state_busy_at=2

[100]
type=auth
auth_type=userpass
password=******
username=100

[100]
type=aor
max_contacts=1
remove_existing=yes

;======================= EXTENSIONS 115
[115]
type=endpoint
context=from-internal
subscribe_context=phones-blf
dtmf_mode=rfc4733
disallow=all
allow=ulaw,alaw,gsm,g722,g729
transport=transport-udp
auth=115
aors=115
direct_media=no
mailboxes=115@default
trust_id_outbound=yes
callerid=John Doe <115>
device_state_busy_at=2

[115]
type=auth
auth_type=userpass
password=******
username=115

[115]
type=aor
max_contacts=1
remove_existing=yes


;======================= EXTENSIONS 116
[116]
type=endpoint
context=from-internal
subscribe_context=phones-blf
dtmf_mode=rfc4733
disallow=all
allow=ulaw,alaw,gsm,g722,g729
transport=transport-udp
auth=116
aors=116
direct_media=no
mailboxes=116@default
trust_id_outbound=yes
callerid=Jane Doe <116>
device_state_busy_at=2

[116]
type=auth
auth_type=userpass
password=*****
username=116

[116]
type=aor
max_contacts=1
remove_existing=yes

You can see the verbose here:

localhost*CLI>
  == Setting global variable 'SIPDOMAIN' to 'ip address'
    -- Executing [100@from-internal:1] NoOp("PJSIP/115-0000004f", "## Internal Call to the 100 ##") in new stack
    -- Executing [100@from-internal:2] Verbose("PJSIP/115-0000004f", "Call start time: 2018-10-10 10:29:11") in new stack
Call start time: 2018-10-10 10:29:11
    -- Executing [100@from-internal:3] Set("PJSIP/115-0000004f", "CDR(calldate)=2018-10-10 10:29:11") in new stack
    -- Executing [100@from-internal:4] Set("PJSIP/115-0000004f", "CDR(useragent)=John Doe") in new stack
    -- Executing [100@from-internal:5] Set("PJSIP/115-0000004f", "NOW=20181010_082911") in new stack
    -- Executing [100@from-internal:6] Set("PJSIP/115-0000004f", "REC_FILE_NAME=OUT_20181010_082911_100_.wav") in new stack
    -- Executing [100@from-internal:7] MixMonitor("PJSIP/115-0000004f", "OUT_20181010_082911_100_.wav,b V(1)") in new stack
    -- Executing [100@from-internal:8] Dial("PJSIP/115-0000004f", "PJSIP/100,20") in new stack
    -- Called PJSIP/100
  == Begin MixMonitor Recording PJSIP/115-0000004f
    -- PJSIP/100-00000050 is ringing
    -- PJSIP/100-00000050 is ringing
    -- PJSIP/100-00000050 answered PJSIP/115-0000004f
    -- Channel PJSIP/100-00000050 joined 'simple_bridge' basic-bridge <ede5c086-044c-428b-a216-ee2b124be165>
    -- Channel PJSIP/115-0000004f joined 'simple_bridge' basic-bridge <ede5c086-044c-428b-a216-ee2b124be165>
  == Setting global variable 'SIPDOMAIN' to 'ip address'
    -- Executing [100@from-internal:1] NoOp("PJSIP/116-00000051", "## Internal Call to the 100 ##") in new stack
    -- Executing [100@from-internal:2] Verbose("PJSIP/116-00000051", "Call start time: 2018-10-10 10:29:21") in new stack
Call start time: 2018-10-10 10:29:21
    -- Executing [100@from-internal:3] Set("PJSIP/116-00000051", "CDR(calldate)=2018-10-10 10:29:21") in new stack
    -- Executing [100@from-internal:4] Set("PJSIP/116-00000051", "CDR(useragent)=Jane Doe") in new stack
    -- Executing [100@from-internal:5] Set("PJSIP/116-00000051", "NOW=20181010_082921") in new stack
    -- Executing [100@from-internal:6] Set("PJSIP/116-00000051", "REC_FILE_NAME=OUT_20181010_082921_100_.wav") in new stack
    -- Executing [100@from-internal:7] MixMonitor("PJSIP/116-00000051", "OUT_20181010_082921_100_.wav,b V(1)") in new stack
    -- Executing [100@from-internal:8] Dial("PJSIP/116-00000051", "PJSIP/100,20") in new stack
    -- Called PJSIP/100
  == Begin MixMonitor Recording PJSIP/116-00000051
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [100@from-internal:9] VoiceMail("PJSIP/116-00000051", "100,su") in new stack
    -- <PJSIP/116-00000051> Playing 'vm-theperson.ulaw' (language 'fr')
    -- <PJSIP/116-00000051> Playing 'digits/1.ulaw' (language 'fr')
    -- <PJSIP/116-00000051> Playing 'digits/0.ulaw' (language 'fr')
    -- <PJSIP/116-00000051> Playing 'digits/0.ulaw' (language 'fr')
    -- <PJSIP/116-00000051> Playing 'vm-isunavail.ulaw' (language 'fr')
    -- <PJSIP/116-00000051> Playing 'beep.ulaw' (language 'fr')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/100/tmp/Maqenl format: wav49, 0x46a5330
    -- x=1, open writing:  /var/spool/asterisk/voicemail/default/100/tmp/Maqenl format: gsm, 0x46a2d90
    -- x=2, open writing:  /var/spool/asterisk/voicemail/default/100/tmp/Maqenl format: wav, 0x461cf90
    -- User hung up
  == Spawn extension (from-internal, 100, 9) exited non-zero on 'PJSIP/116-00000051'
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording PJSIP/116-00000051
[Oct 10 10:29:26] ERROR[15691]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle.  CDR failed.
[Oct 10 10:29:26] ERROR[15691]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle.  CDR failed.
localhost*CLI> exit
Asterisk cleanly ending (0).
Executing last minute cleanups

#4

Is as like the phone is busy…I remember before this is work well.


#5

No SIP log and insufficient verbosity to see why the device was considered busy.


#6
<--- Received SIP request (866 bytes) from UDP:192.168.2.48:5060 --->
SUBSCRIBE sip:100@ip_address SIP/2.0
Via: SIP/2.0/UDP 192.168.2.48:5060;branch=z9hG4bK1757754636;rport
From: <sip:100@ip_address>;tag=182656839
To: <sip:100@ip_address>
Call-ID: 692469169-5060-36814@BJC.BGI.C.EI
CSeq: 390211 SUBSCRIBE
Contact: <sip:100@192.168.2.48:5060>
Authorization: Digest username="100", realm="asterisk", nonce="1539161320/f84de006974434aabe3e48b9addfc73c", uri="sip:100@ip_address", response="ab87daacb3758ae0494ce7e1d285893d", algorithm=md5, cnonce="05034649", opaque="7b2b578420d18a55", qop=auth, nc=00000001
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2130 1.0.9.108
Expires: 3600
Supported: replaces, path, timer
Event: message-summary
Accept: application/simple-message-summary
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP response (331 bytes) to UDP:192.168.2.48:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.48:5060;rport=5060;received=192.168.2.48;branch=z9hG4bK1757754636
Call-ID: 692469169-5060-36814@BJC.BGI.C.EI
From: <sip:100@ip_address>;tag=182656839
To: <sip:100@ip_address>;tag=z9hG4bK1757754636
CSeq: 390211 SUBSCRIBE
Server: Asterisk PBX 13.23.0
Content-Length:  0


<--- Received SIP request (601 bytes) from UDP:192.168.2.48:5060 --->
SUBSCRIBE sip:100@ip_address SIP/2.0
Via: SIP/2.0/UDP 192.168.2.48:5060;branch=z9hG4bK196102943;rport
From: <sip:100@ip_address>;tag=1838608699
To: <sip:100@ip_address>
Call-ID: 1203116245-5060-36815@BJC.BGI.C.EI
CSeq: 390220 SUBSCRIBE
Contact: <sip:100@192.168.2.48:5060>
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2130 1.0.9.108
Expires: 3600
Supported: replaces, path, timer
Event: message-summary
Accept: application/simple-message-summary
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP response (481 bytes) to UDP:192.168.2.48:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.48:5060;rport=5060;received=192.168.2.48;branch=z9hG4bK196102943
Call-ID: 1203116245-5060-36815@BJC.BGI.C.EI
From: <sip:100@ip_address>;tag=1838608699
To: <sip:100@ip_address>;tag=z9hG4bK196102943
CSeq: 390220 SUBSCRIBE
WWW-Authenticate: Digest  realm="asterisk",nonce="1539161321/671df6761fb69f61a777f1d60e957ceb",opaque="4f4286eb57fa1d40",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.23.0
Content-Length:  0


<--- Received SIP request (868 bytes) from UDP:192.168.2.48:5060 --->
SUBSCRIBE sip:100@ip_address SIP/2.0
Via: SIP/2.0/UDP 192.168.2.48:5060;branch=z9hG4bK1507502677;rport
From: <sip:100@ip_address>;tag=1838608699
To: <sip:100@ip_address>
Call-ID: 1203116245-5060-36815@BJC.BGI.C.EI
CSeq: 390221 SUBSCRIBE
Contact: <sip:100@192.168.2.48:5060>
Authorization: Digest username="100", realm="asterisk", nonce="1539161321/671df6761fb69f61a777f1d60e957ceb", uri="sip:100@ip_address", response="5a1838524ecbc2a08e0b0b2fcdda80e9", algorithm=md5, cnonce="06479596", opaque="4f4286eb57fa1d40", qop=auth, nc=00000001
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2130 1.0.9.108
Expires: 3600
Supported: replaces, path, timer
Event: message-summary
Accept: application/simple-message-summary
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP response (333 bytes) to UDP:192.168.2.48:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.48:5060;rport=5060;received=192.168.2.48;branch=z9hG4bK1507502677
Call-ID: 1203116245-5060-36815@BJC.BGI.C.EI
From: <sip:100@ip_address>;tag=1838608699
To: <sip:100@ip_address>;tag=z9hG4bK1507502677
CSeq: 390221 SUBSCRIBE
Server: Asterisk PBX 13.23.0
Content-Length:  0


<--- Received SIP response (925 bytes) from UDP:192.168.2.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPjad1a3c59-adfa-41eb-898f-9f6e7bee3f69
From: "John Doe" <sip:115@ip_address>;tag=9844df37-ab90-4380-b9fb-2999aad21a0b
To: <sip:100@192.168.2.48>;tag=423702587
Call-ID: 2ed59503-adf7-494e-afd2-6044ea7bfc3e
CSeq: 570 INVITE
Contact: <sip:100@192.168.2.48:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.9.108
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length:   305

v=0
o=100 8000 8000 IN IP4 192.168.2.48
s=SIP Call
c=IN IP4 192.168.2.48
t=0 0
m=audio 5074 RTP/AVP 0 8 9 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<--- Transmitting SIP request (394 bytes) to UDP:192.168.2.48:5060 --->
ACK sip:100@192.168.2.48:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPj22c39f61-2559-4b79-870c-34a8c2c989e2
From: "John Doe" <sip:115@ip_address>;tag=9844df37-ab90-4380-b9fb-2999aad21a0b
To: <sip:100@192.168.2.48>;tag=423702587
Call-ID: 2ed59503-adf7-494e-afd2-6044ea7bfc3e
CSeq: 570 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Length:  0


-- PJSIP/100-0000008d answered PJSIP/115-0000008c
<--- Transmitting SIP response (895 bytes) to UDP:192.168.40.55:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.40.55:5060;rport=5060;received=192.168.40.55;branch=z9hG4bK1246192483
Call-ID: 419553336-5060-65@BJC.BGI.EA.FF
From: "John Doe" <sip:115@ip_address>;tag=417714761
To: <sip:100@ip_address>;tag=f87153e6-ccb4-4b38-8240-4c6600c9a954
CSeq: 351 INVITE
Server: Asterisk PBX 13.23.0
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: <sip:ip_address:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   320

v=0
o=- 8000 8002 IN IP4 ip_address
s=Asterisk
c=IN IP4 ip_address
t=0 0
m=audio 18922 RTP/AVP 0 8 9 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<?xml version="1.0" encoding="UTF-8"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="21" state="full" entity="sip:100@ip_address:5060">
 <dialog id="100">
  <state>confirmed</state>
 </dialog>
</dialog-info>

    -- Channel PJSIP/100-0000008d joined 'simple_bridge' basic-bridge <f0bdccc8-8a4b-44ad-bd9c-552f8dabe9d3>
    -- Channel PJSIP/115-0000008c joined 'simple_bridge' basic-bridge <f0bdccc8-8a4b-44ad-bd9c-552f8dabe9d3>
<--- Received SIP response (518 bytes) from UDP:192.168.2.41:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPj02114fb1-57c7-4f9b-9b44-62e48fee7380
From: <sip:100@ip_address>;tag=6e3ed535-7043-4715-95e9-f2c260e7ec70
To: <sip:109@ip_address>;tag=1362922442
Call-ID: 1936819372-5060-87@BJC.BGI.C.EB
CSeq: 2349 NOTIFY
Contact: <sip:109@192.168.2.41:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1630 1.0.4.128
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP request (555 bytes) from UDP:192.168.40.55:5060 --->
ACK sip:ip_address:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.40.55:5060;branch=z9hG4bK1072774045;rport
From: "John Doe" <sip:115@ip_address>;tag=417714761
To: <sip:100@ip_address>;tag=f87153e6-ccb4-4b38-8240-4c6600c9a954
Call-ID: 419553336-5060-65@BJC.BGI.EA.FF
CSeq: 351 ACK
Contact: <sip:115@192.168.40.55:5060>
X-Grandstream-PBX: true
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.9.108
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (522 bytes) from UDP:192.168.2.59:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPje4a0468b-21e0-4f6d-9739-061715aa66d9
From: <sip:100@ip_address>;tag=bb67748d-36d4-42d2-af41-7e161609fdcf
To: <sip:108@ip_address>;tag=1721852699
Call-ID: 1926437745-5060-108571@BJC.BGI.C.FJ
CSeq: 8902 NOTIFY
Contact: <sip:108@192.168.2.59:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2135 1.0.9.108
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (522 bytes) from UDP:192.168.2.36:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPjec4ba102-f032-4951-b14d-38277893c6ce
From: <sip:100@ip_address>;tag=fe3b99e1-8588-48df-bac4-6400365bca25
To: <sip:110@ip_address>;tag=420752384
Call-ID: 1796597223-5060-176058@BJC.BGI.C.DG
CSeq: 16633 NOTIFY
Contact: <sip:110@192.168.2.36:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2170 1.0.9.108
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (520 bytes) from UDP:192.168.2.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPj04bb7ebc-48db-4489-9ef3-d528f347b572
From: <sip:100@ip_address>;tag=75931044-680f-4212-8faa-b567b6c5187e
To: <sip:100@ip_address>;tag=1360617744
Call-ID: 843498800-5060-32087@BJC.BGI.C.EI
CSeq: 8782 NOTIFY
Contact: <sip:100@192.168.2.48:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.9.108
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP request (1191 bytes) from UDP:192.168.40.50:5060 --->
INVITE sip:100@ip_address SIP/2.0
Via: SIP/2.0/UDP 192.168.40.50:5060;branch=z9hG4bK898076465;rport
From: "Jane Doe" <sip:116@ip_address>;tag=149243391
To: <sip:100@ip_address>
Call-ID: 1338898576-5060-32@BJC.BGI.EA.FA
CSeq: 310 INVITE
Contact: "Jane Doe" <sip:116@192.168.40.50:5060>
Max-Forwards: 70
User-Agent: Grandstream GXP1625 1.0.4.128
Privacy: none
P-Preferred-Identity: "Jane Doe" <sip:116@ip_address>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=00-1D-AA-A8-19-30
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=00-0B-82-76-3F-97
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   403

v=0
o=116 8000 8000 IN IP4 192.168.40.50
s=SIP Call
c=IN IP4 192.168.40.50
t=0 0
m=audio 5068 RTP/AVP 0 8 18 4 2 9 97 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<--- Transmitting SIP response (485 bytes) to UDP:192.168.40.50:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.40.50:5060;rport=5060;received=192.168.40.50;branch=z9hG4bK898076465
Call-ID: 1338898576-5060-32@BJC.BGI.EA.FA
From: "Jane Doe" <sip:116@ip_address>;tag=149243391
To: <sip:100@ip_address>;tag=z9hG4bK898076465
CSeq: 310 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1539161326/82b2ca72b7154ae1e2c5e72449055e37",opaque="5c33e6dc1481f226",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.23.0
Content-Length:  0


<--- Received SIP request (284 bytes) from UDP:192.168.40.50:5060 --->
ACK sip:100@ip_address SIP/2.0
Via: SIP/2.0/UDP 192.168.40.50:5060;branch=z9hG4bK898076465;rport
From: "Jane Doe" <sip:116@ip_address>;tag=149243391
To: <sip:100@ip_address>;tag=z9hG4bK898076465
Call-ID: 1338898576-5060-32@BJC.BGI.EA.FA
CSeq: 310 ACK
Content-Length: 0


<--- Received SIP request (1458 bytes) from UDP:192.168.40.50:5060 --->
INVITE sip:100@ip_address SIP/2.0
Via: SIP/2.0/UDP 192.168.40.50:5060;branch=z9hG4bK1281867323;rport
From: "Jane Doe" <sip:116@ip_address>;tag=149243391
To: <sip:100@ip_address>
Call-ID: 1338898576-5060-32@BJC.BGI.EA.FA
CSeq: 311 INVITE
Contact: "Jane Doe" <sip:116@192.168.40.50:5060>
Authorization: Digest username="116", realm="asterisk", nonce="1539161326/82b2ca72b7154ae1e2c5e72449055e37", uri="sip:100@ip_address", response="834c7d87bf8436158d9b0d8731562eb3", algorithm=md5, cnonce="04421707", opaque="5c33e6dc1481f226", qop=auth, nc=00000001
Max-Forwards: 70
User-Agent: Grandstream GXP1625 1.0.4.128
Privacy: none
P-Preferred-Identity: "Jane Doe" <sip:116@ip_address>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=00-1D-AA-A8-19-30
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=00-0B-82-76-3F-97
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   403

v=0
o=116 8000 8000 IN IP4 192.168.40.50
s=SIP Call
c=IN IP4 192.168.40.50
t=0 0
m=audio 5068 RTP/AVP 0 8 18 4 2 9 97 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

  == Setting global variable 'SIPDOMAIN' to 'ip_address'
<--- Transmitting SIP response (312 bytes) to UDP:192.168.40.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.40.50:5060;rport=5060;received=192.168.40.50;branch=z9hG4bK1281867323
Call-ID: 1338898576-5060-32@BJC.BGI.EA.FA
From: "Jane Doe" <sip:116@ip_address>;tag=149243391
To: <sip:100@ip_address>
CSeq: 311 INVITE
Server: Asterisk PBX 13.23.0
Content-Length:  0


    -- Executing [100@from-internal:1] NoOp("PJSIP/116-0000008e", "## Internal Call to the 100 ##") in new stack
    -- Executing [100@from-internal:2] Verbose("PJSIP/116-0000008e", "Call start time: 2018-10-10 10:48:46") in new stack
Call start time: 2018-10-10 10:48:46
    -- Executing [100@from-internal:3] Set("PJSIP/116-0000008e", "CDR(calldate)=2018-10-10 10:48:46") in new stack
    -- Executing [100@from-internal:4] Set("PJSIP/116-0000008e", "CDR(useragent)=Jane Doe") in new stack
    -- Executing [100@from-internal:5] Set("PJSIP/116-0000008e", "NOW=20181010_084846") in new stack
    -- Executing [100@from-internal:6] Set("PJSIP/116-0000008e", "REC_FILE_NAME=OUT_20181010_084846_100_.wav") in new stack
    -- Executing [100@from-internal:7] MixMonitor("PJSIP/116-0000008e", "OUT_20181010_084846_100_.wav,b V(1)") in new stack
    -- Executing [100@from-internal:8] Dial("PJSIP/116-0000008e", "PJSIP/100,20") in new stack
    -- Called PJSIP/100
<--- Transmitting SIP request (1013 bytes) to UDP:192.168.2.48:5060 --->
INVITE sip:100@192.168.2.48:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPjcbcabfca-2f98-47c2-b8b3-6c5d55529079
From: "Jane Doe" <sip:116@ip_address>;tag=b8061174-816c-4e3e-9876-d563c649919b
To: <sip:100@192.168.2.48>
Contact: <sip:asterisk@ip_address:5060>
Call-ID: a8ec40ba-8049-4afb-ab18-26e2ef7dcf20
CSeq: 9181 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Type: application/sdp
Content-Length:   355

v=0
o=- 1032292483 1032292483 IN IP4 ip_address
s=Asterisk
c=IN IP4 ip_address
t=0 0
m=audio 11152 RTP/AVP 0 8 3 9 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

  == Begin MixMonitor Recording PJSIP/116-0000008e
<--- Received SIP response (485 bytes) from UDP:192.168.2.48:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPjcbcabfca-2f98-47c2-b8b3-6c5d55529079
From: "Jane Doe" <sip:116@ip_address>;tag=b8061174-816c-4e3e-9876-d563c649919b
To: <sip:100@192.168.2.48>
Call-ID: a8ec40ba-8049-4afb-ab18-26e2ef7dcf20
CSeq: 9181 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.9.108
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (542 bytes) from UDP:192.168.2.48:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPjcbcabfca-2f98-47c2-b8b3-6c5d55529079
From: "Jane Doe" <sip:116@ip_address>;tag=b8061174-816c-4e3e-9876-d563c649919b
To: <sip:100@192.168.2.48>;tag=950722919
Call-ID: a8ec40ba-8049-4afb-ab18-26e2ef7dcf20
CSeq: 9181 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.9.108
Warning: 399 GS "All lines are in use"
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP request (395 bytes) to UDP:192.168.2.48:5060 --->
ACK sip:100@192.168.2.48:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPjcbcabfca-2f98-47c2-b8b3-6c5d55529079
From: "Jane Doe" <sip:116@ip_address>;tag=b8061174-816c-4e3e-9876-d563c649919b
To: <sip:100@192.168.2.48>;tag=950722919
Call-ID: a8ec40ba-8049-4afb-ab18-26e2ef7dcf20
CSeq: 9181 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Length:  0


  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [100@from-internal:9] VoiceMail("PJSIP/116-0000008e", "100,su") in new stack
<--- Transmitting SIP response (896 bytes) to UDP:192.168.40.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.40.50:5060;rport=5060;received=192.168.40.50;branch=z9hG4bK1281867323
Call-ID: 1338898576-5060-32@BJC.BGI.EA.FA
From: "Jane Doe" <sip:116@ip_address>;tag=149243391
To: <sip:100@ip_address>;tag=d4118de0-f1a1-46cf-a450-919360630e76
CSeq: 311 INVITE
Server: Asterisk PBX 13.23.0
Contact: <sip:ip_address:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   320

v=0
o=- 8000 8002 IN IP4 ip_address
s=Asterisk
c=IN IP4 ip_address
t=0 0
m=audio 13834 RTP/AVP 0 8 9 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (530 bytes) from UDP:192.168.40.50:5060 --->
ACK sip:ip_address:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.40.50:5060;branch=z9hG4bK154345992;rport
From: "Jane Doe" <sip:116@ip_address>;tag=149243391
To: <sip:100@ip_address>;tag=d4118de0-f1a1-46cf-a450-919360630e76
Call-ID: 1338898576-5060-32@BJC.BGI.EA.FA
CSeq: 311 ACK
Contact: <sip:116@192.168.40.50:5060>
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.4.128
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


    -- <PJSIP/116-0000008e> Playing 'vm-theperson.ulaw' (language 'fr')
    -- <PJSIP/116-0000008e> Playing 'digits/1.ulaw' (language 'fr')
    -- <PJSIP/116-0000008e> Playing 'digits/0.ulaw' (language 'fr')
    -- <PJSIP/116-0000008e> Playing 'digits/0.ulaw' (language 'fr')
    -- <PJSIP/116-0000008e> Playing 'vm-isunavail.ulaw' (language 'fr')
    -- <PJSIP/116-0000008e> Playing 'beep.ulaw' (language 'fr')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/100/tmp/kQCONV format: wav49, 0x7f5204013760
    -- x=1, open writing:  /var/spool/asterisk/voicemail/default/100/tmp/kQCONV format: gsm, 0x7f520400ee00
    -- x=2, open writing:  /var/spool/asterisk/voicemail/default/100/tmp/kQCONV format: wav, 0x7f520400bca0

<--- Received SIP request (531 bytes) from UDP:192.168.40.50:5060 --->
BYE sip:ip_address:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.40.50:5060;branch=z9hG4bK2098725826;rport
From: "Jane Doe" <sip:116@ip_address>;tag=149243391
To: <sip:100@ip_address>;tag=d4118de0-f1a1-46cf-a450-919360630e76
Call-ID: 1338898576-5060-32@BJC.BGI.EA.FA
CSeq: 312 BYE
Contact: <sip:116@192.168.40.50:5060>
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.4.128
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP response (346 bytes) to UDP:192.168.40.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.40.50:5060;rport=5060;received=192.168.40.50;branch=z9hG4bK2098725826
Call-ID: 1338898576-5060-32@BJC.BGI.EA.FA
From: "Jane Doe" <sip:116@ip_address>;tag=149243391
To: <sip:100@ip_address>;tag=d4118de0-f1a1-46cf-a450-919360630e76
CSeq: 312 BYE
Server: Asterisk PBX 13.23.0
Content-Length:  0


    -- User hung up
  == Spawn extension (from-internal, 100, 9) exited non-zero on 'PJSIP/116-0000008e'
  == MixMonitor close filestream (mixed)
<--- Transmitting SIP request (628 bytes) to UDP:192.168.2.48:5060 --->
NOTIFY sip:100@192.168.2.48:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPj175cc5f5-e91b-4d5f-88d4-99ec85e1643c
From: <sip:100@ip_address>;tag=6f53d6bb-94c5-4639-8683-df5b07365043
To: <sip:100@192.168.2.48>
Contact: <sip:100@ip_address:5060>
Call-ID: 38416384-a12a-493a-8db3-8579ef4457dc
CSeq: 3284 NOTIFY
Subscription-State: terminated
Event: message-summary
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Type: application/simple-message-summary
Content-Length:    49

Messages-Waiting: yes
Voice-Message: 3/4 (0/0)

  == End MixMonitor Recording PJSIP/116-0000008e
[Oct 10 10:48:56] ERROR[15691]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle.  CDR failed.
[Oct 10 10:48:56] ERROR[15691]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle.  CDR failed.
<--- Received SIP response (485 bytes) from UDP:192.168.2.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPj175cc5f5-e91b-4d5f-88d4-99ec85e1643c
From: <sip:100@ip_address>;tag=6f53d6bb-94c5-4639-8683-df5b07365043
To: <sip:100@192.168.2.48>;tag=1564492303
Call-ID: 38416384-a12a-493a-8db3-8579ef4457dc
CSeq: 3284 NOTIFY
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.9.108
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP request (565 bytes) from UDP:192.168.2.48:5060 --->
BYE sip:asterisk@ip_address:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.48:5060;branch=z9hG4bK45034168;rport
From: <sip:100@192.168.2.48>;tag=423702587
To: "John Doe" <sip:115@ip_address>;tag=9844df37-ab90-4380-b9fb-2999aad21a0b
Call-ID: 2ed59503-adf7-494e-afd2-6044ea7bfc3e
CSeq: 571 BYE
Contact: <sip:100@192.168.2.48:5060>
X-Grandstream-PBX: true
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.9.108
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP response (346 bytes) to UDP:192.168.2.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.48:5060;rport=5060;received=192.168.2.48;branch=z9hG4bK45034168
Call-ID: 2ed59503-adf7-494e-afd2-6044ea7bfc3e
From: <sip:100@192.168.2.48>;tag=423702587
To: "John Doe" <sip:115@ip_address>;tag=9844df37-ab90-4380-b9fb-2999aad21a0b
CSeq: 571 BYE
Server: Asterisk PBX 13.23.0
Content-Length:  0


    -- Channel PJSIP/100-0000008d left 'simple_bridge' basic-bridge <f0bdccc8-8a4b-44ad-bd9c-552f8dabe9d3>
<--- Transmitting SIP request (808 bytes) to UDP:192.168.2.41:5060 --->
NOTIFY sip:109@192.168.2.41:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPjd5f07c81-b8f5-4f09-99b6-1a55b3e3ec1e
From: <sip:100@ip_address>;tag=6e3ed535-7043-4715-95e9-f2c260e7ec70
To: <sip:109@ip_address>;tag=1362922442
Contact: <sip:ip_address:5060>
Call-ID: 1936819372-5060-87@BJC.BGI.C.EB
CSeq: 2350 NOTIFY
Event: dialog
Subscription-State: active;expires=2530
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Type: application/dialog-info+xml
Content-Length:   230

<?xml version="1.0" encoding="UTF-8"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="22" state="full" entity="sip:100@ip_address:5060">
 <dialog id="100">
  <state>terminated</state>
 </dialog>
</dialog-info>

<--- Transmitting SIP request (810 bytes) to UDP:192.168.2.48:5060 --->
NOTIFY sip:100@192.168.2.48:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPj153dff40-d78e-47be-bb75-db182d6b2646
From: <sip:100@ip_address>;tag=75931044-680f-4212-8faa-b567b6c5187e
To: <sip:100@ip_address>;tag=1360617744
Contact: <sip:ip_address:5060>
Call-ID: 843498800-5060-32087@BJC.BGI.C.EI
CSeq: 8783 NOTIFY
Event: dialog
Subscription-State: active;expires=2577
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Type: application/dialog-info+xml
Content-Length:   230

<?xml version="1.0" encoding="UTF-8"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="22" state="full" entity="sip:100@ip_address:5060">
 <dialog id="100">
  <state>terminated</state>
 </dialog>
</dialog-info>


<?xml version="1.0" encoding="UTF-8"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="22" state="full" entity="sip:100@ip_address:5060">
 <dialog id="100">
  <state>terminated</state>
 </dialog>
</dialog-info>

    -- Channel PJSIP/115-0000008c left 'simple_bridge' basic-bridge <f0bdccc8-8a4b-44ad-bd9c-552f8dabe9d3>
  == Spawn extension (from-internal, 100, 8) exited non-zero on 'PJSIP/115-0000008c'
  == MixMonitor close filestream (mixed)
<--- Transmitting SIP request (416 bytes) to UDP:192.168.40.55:5060 --->
BYE sip:115@192.168.40.55:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPj6f7a1f8e-d213-467e-9888-e33f58ff3bae
From: <sip:100@ip_address>;tag=f87153e6-ccb4-4b38-8240-4c6600c9a954
To: "John Doe" <sip:115@ip_address>;tag=417714761
Call-ID: 419553336-5060-65@BJC.BGI.EA.FF
CSeq: 29442 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Length:  0


  == End MixMonitor Recording PJSIP/115-0000008c
[Oct 10 10:49:01] ERROR[15691]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle.  CDR failed.

<--- Received SIP response (527 bytes) from UDP:192.168.40.55:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPj6f7a1f8e-d213-467e-9888-e33f58ff3bae
From: <sip:100@ip_address>;tag=f87153e6-ccb4-4b38-8240-4c6600c9a954
To: "John Doe" <sip:115@ip_address>;tag=417714761
Call-ID: 419553336-5060-65@BJC.BGI.EA.FF
CSeq: 29442 BYE
Contact: <sip:115@192.168.40.55:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.9.108
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<--- Received SIP response (520 bytes) from UDP:192.168.2.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPj153dff40-d78e-47be-bb75-db182d6b2646
From: <sip:100@ip_address>;tag=75931044-680f-4212-8faa-b567b6c5187e
To: <sip:100@ip_address>;tag=1360617744
Call-ID: 843498800-5060-32087@BJC.BGI.C.EI
CSeq: 8783 NOTIFY
Contact: <sip:100@192.168.2.48:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.9.108
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


#7

<--- Received SIP response (542 bytes) from UDP:192.168.2.48:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPjcbcabfca-2f98-47c2-b8b3-6c5d55529079
From: "Jane Doe" <sip:116@ip_address>;tag=b8061174-816c-4e3e-9876-d563c649919b
To: <sip:100@192.168.2.48>;tag=950722919
Call-ID: a8ec40ba-8049-4afb-ab18-26e2ef7dcf20
CSeq: 9181 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.9.108
Warning: 399 GS "All lines are in use"
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

The Grandstream thinks itself busy. This is not related to the Asterisk configuration.


#8

So this is coming from my Grandstream.


#9

Ok, I found the way to have my second call on my Grandstream new version. I must to edit the Multi-Purpose Keys and add a second account like a “default” or “line” because I have only a one account.

Like this one bellow:


#10

But I do not have a second call on the incoming calls, when one of ip phone is online with another caller. The call go directly on the answerer.


#11

Here you can see my cli :

<--- Received SIP request (1204 bytes) from UDP:192.168.40.50:5060 --->
INVITE sip:028992018@ip_address SIP/2.0
Via: SIP/2.0/UDP 192.168.40.50:5060;branch=z9hG4bK616006517;rport
From: "Jane Doe" <sip:116@ip_address>;tag=1043865744
To: <sip:028992018@ip_address>
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
CSeq: 350 INVITE
Contact: "Jane Doe" <sip:116@192.168.40.50:5060>
Max-Forwards: 70
User-Agent: Grandstream GXP1625 1.0.4.128
Privacy: none
P-Preferred-Identity: "Jane Doe" <sip:116@ip_address>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=00-1D-AA-A8-19-30
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=00-0B-82-76-3F-97
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   403

v=0
o=116 8000 8000 IN IP4 192.168.40.50
s=SIP Call
c=IN IP4 192.168.40.50
t=0 0
m=audio 5076 RTP/AVP 0 8 18 4 2 9 97 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<--- Transmitting SIP response (492 bytes) to UDP:192.168.40.50:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.40.50:5060;rport=5060;received=192.168.40.50;branch=z9hG4bK616006517
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
From: "Jane Doe" <sip:116@ip_address>;tag=1043865744
To: <sip:028992018@ip_address>;tag=z9hG4bK616006517
CSeq: 350 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1539180496/b30db263bc407c62f5fdbb74e284d316",opaque="777b70bd484d1d02",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.23.0
Content-Length:  0


<--- Received SIP request (297 bytes) from UDP:192.168.40.50:5060 --->
ACK sip:028992018@ip_address SIP/2.0
Via: SIP/2.0/UDP 192.168.40.50:5060;branch=z9hG4bK616006517;rport
From: "Jane Doe" <sip:116@ip_address>;tag=1043865744
To: <sip:028992018@ip_address>;tag=z9hG4bK616006517
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
CSeq: 350 ACK
Content-Length: 0


<--- Received SIP request (1477 bytes) from UDP:192.168.40.50:5060 --->
INVITE sip:028992018@ip_address SIP/2.0
Via: SIP/2.0/UDP 192.168.40.50:5060;branch=z9hG4bK2143641284;rport
From: "Jane Doe" <sip:116@ip_address>;tag=1043865744
To: <sip:028992018@ip_address>
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
CSeq: 351 INVITE
Contact: "Jane Doe" <sip:116@192.168.40.50:5060>
Authorization: Digest username="116", realm="asterisk", nonce="1539180496/b30db263bc407c62f5fdbb74e284d316", uri="sip:028992018@ip_address", response="f666b74aaf71ca5a273b7c306bf5c85b", algorithm=md5, cnonce="02528634", opaque="777b70bd484d1d02", qop=auth, nc=00000001
Max-Forwards: 70
User-Agent: Grandstream GXP1625 1.0.4.128
Privacy: none
P-Preferred-Identity: "Jane Doe" <sip:116@ip_address>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=00-1D-AA-A8-19-30
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=00-0B-82-76-3F-97
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   403

v=0
o=116 8000 8000 IN IP4 192.168.40.50
s=SIP Call
c=IN IP4 192.168.40.50
t=0 0
m=audio 5076 RTP/AVP 0 8 18 4 2 9 97 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

  == Setting global variable 'SIPDOMAIN' to 'ip_address'
<--- Transmitting SIP response (319 bytes) to UDP:192.168.40.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.40.50:5060;rport=5060;received=192.168.40.50;branch=z9hG4bK2143641284
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
From: "Jane Doe" <sip:116@ip_address>;tag=1043865744
To: <sip:028992018@ip_address>
CSeq: 351 INVITE
Server: Asterisk PBX 13.23.0
Content-Length:  0


    -- Executing [028992018@from-internal:1] NoOp("PJSIP/116-0000042b", "Outgoing Call from "Jane Doe" <116> to 028992018") in new stack
    -- Executing [028992018@from-internal:2] Verbose("PJSIP/116-0000042b", "Call start time: 2018-10-10 16:08:16") in new stack
Call start time: 2018-10-10 16:08:16
    -- Executing [028992018@from-internal:3] Set("PJSIP/116-0000042b", "CDR(calldate)=2018-10-10 16:08:16") in new stack
    -- Executing [028992018@from-internal:4] Set("PJSIP/116-0000042b", "CDR(useragent)=Jane Doe") in new stack
    -- Executing [028992018@from-internal:5] Set("PJSIP/116-0000042b", "POSTE=116") in new stack
    -- Executing [028992018@from-internal:6] NoOp("PJSIP/116-0000042b", "SendedCID = 116") in new stack
    -- Executing [028992018@from-internal:7] Set("PJSIP/116-0000042b", "CALLERID(num)=042770677") in new stack
    -- Executing [028992018@from-internal:8] NoOp("PJSIP/116-0000042b", "SendedCID = 042770677") in new stack
    -- Executing [028992018@from-internal:9] Set("PJSIP/116-0000042b", "NOW=2018_10_10_16_08_16") in new stack
    -- Executing [028992018@from-internal:10] System("PJSIP/116-0000042b", "echo "--appel_sortant --- callerid : 042770677 ---- 2018_10_10_16_08_16 ----" >> /var/spool/asterisk/log/debug.txt") in new stack
    -- Executing [028992018@from-internal:11] Set("PJSIP/116-0000042b", "REC_FILE_NAME=OUT_2018_10_10_16_08_16_028992018_116.wav") in new stack
    -- Executing [028992018@from-internal:12] Answer("PJSIP/116-0000042b", "") in new stack
<--- Transmitting SIP response (903 bytes) to UDP:192.168.40.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.40.50:5060;rport=5060;received=192.168.40.50;branch=z9hG4bK2143641284
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
From: "Jane Doe" <sip:116@ip_address>;tag=1043865744
To: <sip:028992018@ip_address>;tag=dba3540b-532b-4289-ab92-fafb39d568bd
CSeq: 351 INVITE
Server: Asterisk PBX 13.23.0
Contact: <sip:ip_address:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   320

v=0
o=- 8000 8002 IN IP4 ip_address
s=Asterisk
c=IN IP4 ip_address
t=0 0
m=audio 14994 RTP/AVP 0 8 9 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (537 bytes) from UDP:192.168.40.50:5060 --->
ACK sip:ip_address:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.40.50:5060;branch=z9hG4bK152621513;rport
From: "Jane Doe" <sip:116@ip_address>;tag=1043865744
To: <sip:028992018@ip_address>;tag=dba3540b-532b-4289-ab92-fafb39d568bd
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
CSeq: 351 ACK
Contact: <sip:116@192.168.40.50:5060>
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.4.128
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


    -- Executing [028992018@from-internal:13] NoOp("PJSIP/116-0000042b", "n° Poste = 116") in new stack
    -- Executing [028992018@from-internal:14] MixMonitor("PJSIP/116-0000042b", "OUT_2018_10_10_16_08_16_028992018_116.wav,b V(1)") in new stack
    -- Executing [028992018@from-internal:15] Goto("PJSIP/116-0000042b", "SetProv") in new stack
    -- Goto (from-internal,028992018,17)
    -- Executing [028992018@from-internal:17] Set("PJSIP/116-0000042b", "PROV2USE=BelgiumVoIP") in new stack
    -- Executing [028992018@from-internal:18] NoOp("PJSIP/116-0000042b", "Provider to use : BelgiumVoIP") in new stack
    -- Executing [028992018@from-internal:19] GotoIf("PJSIP/116-0000042b", "0?WideVoIP") in new stack
    -- Executing [028992018@from-internal:20] GotoIf("PJSIP/116-0000042b", "0?Selfone") in new stack
    -- Executing [028992018@from-internal:21] GotoIf("PJSIP/116-0000042b", "1?BelgiumVoIP") in new stack
    -- Goto (from-internal,028992018,23)
    -- Executing [028992018@from-internal:23] Set("PJSIP/116-0000042b", "NUM2DIAL=028992018") in new stack
    -- Executing [028992018@from-internal:24] System("PJSIP/116-0000042b", "echo "--BelgiumVoIP  --- callerid : 042770677 ---- 2018_10_10_16_08_17 ----" >> /var/spool/asterisk/log/debug.txt") in new stack
  == Begin MixMonitor Recording PJSIP/116-0000042b
    -- Executing [028992018@from-internal:25] NoOp("PJSIP/116-0000042b", "CD BelgiumVoIP") in new stack
    -- Executing [028992018@from-internal:26] Dial("PJSIP/116-0000042b", "PJSIP/028992018@belgium-voip,60") in new stack
    -- Called PJSIP/028992018@belgium-voip
<--- Transmitting SIP request (1057 bytes) to UDP:188.66.8.19:5060 --->
INVITE sip:028992018@voip.belgium-voip.com:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPjf84a1734-56b3-4493-8d06-f8c04d473d6e
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>
Contact: <sip:asterisk@ip_address:5060>
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22269 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Type: application/sdp
Content-Length:   353

v=0
o=- 351740866 351740866 IN IP4 ip_address
s=Asterisk
c=IN IP4 ip_address
t=0 0
m=audio 11458 RTP/AVP 0 8 3 9 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (563 bytes) from UDP:188.66.8.19:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP ip_address:5060;rport=39748;branch=z9hG4bKPjf84a1734-56b3-4493-8d06-f8c04d473d6e;received=ip_address
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>;tag=6a4cc70e519e85e8bc5c654eeaf70770.2786
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22269 INVITE
Proxy-Authenticate: Digest realm="voip.belgium-voip.com", nonce="W74I/Vu+B9GTBdq3AS2Zd7ivAKcgC/4w"
Server: Enswitch SIP proxy
Content-Length: 0


<--- Transmitting SIP request (469 bytes) to UDP:188.66.8.19:5060 --->
ACK sip:028992018@voip.belgium-voip.com:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPjf84a1734-56b3-4493-8d06-f8c04d473d6e
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>;tag=6a4cc70e519e85e8bc5c654eeaf70770.2786
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22269 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Length:  0


<--- Transmitting SIP request (1274 bytes) to UDP:188.66.8.19:5060 --->
INVITE sip:028992018@voip.belgium-voip.com:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPja839b4d7-d17a-4c3f-adf7-f2603a641d6d
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>
Contact: <sip:asterisk@ip_address:5060>
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22270 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Proxy-Authorization: Digest username="0427714121", realm="voip.belgium-voip.com", nonce="W74I/Vu+B9GTBdq3AS2Zd7ivAKcgC/4w", uri="sip:028992018@voip.belgium-voip.com:5060", response="f030afed2b84de6cfd9720541e4409be"
Content-Type: application/sdp
Content-Length:   353

v=0
o=- 351740866 351740866 IN IP4 ip_address
s=Asterisk
c=IN IP4 ip_address
t=0 0
m=audio 11458 RTP/AVP 0 8 3 9 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (430 bytes) from UDP:188.66.8.19:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ip_address:5060;rport=39748;branch=z9hG4bKPja839b4d7-d17a-4c3f-adf7-f2603a641d6d;received=ip_address
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22270 INVITE
Server: Enswitch SIP proxy
Content-Length: 0


<--- Received SIP response (655 bytes) from UDP:188.66.8.19:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ip_address:5060;received=ip_address;rport=39748;branch=z9hG4bKPja839b4d7-d17a-4c3f-adf7-f2603a641d6d
Record-Route: <sip:188.66.8.19;lr=on>
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>;tag=as1d37edb5
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22270 INVITE
Server: 3StarsNet VoipSwitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:028992018@188.66.8.52:5060>
Content-Length: 0


    -- PJSIP/belgium-voip-0000042c is ringing
    -- PJSIP/belgium-voip-0000042c is ringing
<--- Received SIP request (1302 bytes) from UDP:188.66.8.19:5060 --->
INVITE sip:028992018@ip_address:39748 SIP/2.0
Record-Route: <sip:188.66.8.19;lr=on>
Via: SIP/2.0/UDP 188.66.8.19;branch=z9hG4bK6092.1c17d19bd75143ad831581f6251e186f.0
Via: SIP/2.0/UDP 188.66.8.52:5060;received=188.66.8.52;branch=z9hG4bK208f6491;rport=5060
Max-Forwards: 69
From: "Jane Doe" <sip:042770677@188.66.8.52>;tag=as53b1f9bd
To: <sip:028992018@ip_address:39748>
Contact: <sip:042770677@188.66.8.52:5060>
Call-ID: 327ced0e1f5b5bf9347129773ecf5b4c@188.66.8.52:5060
CSeq: 102 INVITE
User-Agent: 3StarsNet VoipSwitch
Date: Wed, 10 Oct 2018 14:08:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Enswitch-Uniqueid: 1539180497.140627
Diversion: <sip:028992018@ast29>
Content-Type: application/sdp
Content-Length: 370
X-Enswitch-RURI: sip:028992018@ip_address:39748
X-Enswitch-Source: 188.66.8.52:5060
X-Enswitch-External: yes

v=0
o=root 1420374773 1420374773 IN IP4 188.66.8.27
s=3StarsNet VoipSwitch
c=IN IP4 188.66.8.27
t=0 0
m=audio 14040 RTP/AVP 8 0 18 3 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=sdpmangled:yes

  == Setting global variable 'SIPDOMAIN' to 'ip_address'
<--- Transmitting SIP response (486 bytes) to UDP:188.66.8.19:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 188.66.8.19;rport=5060;received=188.66.8.19;branch=z9hG4bK6092.1c17d19bd75143ad831581f6251e186f.0
Via: SIP/2.0/UDP 188.66.8.52:5060;rport=5060;received=188.66.8.52;branch=z9hG4bK208f6491
Record-Route: <sip:188.66.8.19;lr>
Call-ID: 327ced0e1f5b5bf9347129773ecf5b4c@188.66.8.52:5060
From: "Jane Doe" <sip:042770677@188.66.8.52>;tag=as53b1f9bd
To: <sip:028992018@ip_address>
CSeq: 102 INVITE
Server: Asterisk PBX 13.23.0
Content-Length:  0


    -- Executing [028992018@from-external:1] NoOp("PJSIP/belgium-voip-0000042d", "## Incoming Call from "Jane Doe" <042770677> ##") in new stack
    -- Executing [028992018@from-external:2] Verbose("PJSIP/belgium-voip-0000042d", "Call start time: 2018-10-10 16:08:17") in new stack
Call start time: 2018-10-10 16:08:17
    -- Executing [028992018@from-external:3] Set("PJSIP/belgium-voip-0000042d", "CDR(calldate)=2018-10-10 16:08:17") in new stack
    -- Executing [028992018@from-external:4] Set("PJSIP/belgium-voip-0000042d", "CDR(useragent)=Jane Doe") in new stack
    -- Executing [028992018@from-external:5] Set("PJSIP/belgium-voip-0000042d", "POSTE_EXT=042770677") in new stack
    -- Executing [028992018@from-external:6] Ringing("PJSIP/belgium-voip-0000042d", "") in new stack
    -- Executing [028992018@from-external:7] System("PJSIP/belgium-voip-0000042d", "echo "--appel_sortant --- callerid : 042770677 ---- 2018/10/10 16:08:17 ----" >> /var/spool/asterisk/log/debug.txt") in new stack
<--- Transmitting SIP response (673 bytes) to UDP:188.66.8.19:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 188.66.8.19;rport=5060;received=188.66.8.19;branch=z9hG4bK6092.1c17d19bd75143ad831581f6251e186f.0
Via: SIP/2.0/UDP 188.66.8.52:5060;rport=5060;received=188.66.8.52;branch=z9hG4bK208f6491
Record-Route: <sip:188.66.8.19;lr>
Call-ID: 327ced0e1f5b5bf9347129773ecf5b4c@188.66.8.52:5060
From: "Jane Doe" <sip:042770677@188.66.8.52>;tag=as53b1f9bd
To: <sip:028992018@ip_address>;tag=faf14a39-eea3-4020-89db-ed76ae7324f0
CSeq: 102 INVITE
Server: Asterisk PBX 13.23.0
Contact: <sip:ip_address:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Content-Length:  0


<--- Received SIP response (655 bytes) from UDP:188.66.8.19:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ip_address:5060;received=ip_address;rport=39748;branch=z9hG4bKPja839b4d7-d17a-4c3f-adf7-f2603a641d6d
Record-Route: <sip:188.66.8.19;lr=on>
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>;tag=as1d37edb5
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22270 INVITE
Server: 3StarsNet VoipSwitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:028992018@188.66.8.52:5060>
Content-Length: 0


    -- PJSIP/belgium-voip-0000042c is ringing
    -- PJSIP/belgium-voip-0000042c is ringing
    -- Executing [028992018@from-external:8] Set("PJSIP/belgium-voip-0000042d", "REC_FILE_NAME=IN__028992018_042770677.wav") in new stack
    -- Executing [028992018@from-external:9] MixMonitor("PJSIP/belgium-voip-0000042d", "IN__028992018_042770677.wav,b V(1)") in new stack
    -- Executing [028992018@from-external:10] ChanIsAvail("PJSIP/belgium-voip-0000042d", "PJSIP/100,sa") in new stack
    -- Executing [028992018@from-external:11] Set("PJSIP/belgium-voip-0000042d", "PHONESTATUS=2") in new stack
    -- Executing [028992018@from-external:12] Set("PJSIP/belgium-voip-0000042d", "PHONEAVAIL=") in new stack
    -- Executing [028992018@from-external:13] NoOp("PJSIP/belgium-voip-0000042d", "## Status of device is 2 ##") in new stack
    -- Executing [028992018@from-external:14] GotoIf("PJSIP/belgium-voip-0000042d", "0?busy:call") in new stack
    -- Goto (from-external,028992018,18)
    -- Executing [028992018@from-external:18] Dial("PJSIP/belgium-voip-0000042d", ",20") in new stack
[Oct 10 16:08:17] WARNING[20678][C-00000255]: app_dial.c:2281 dial_exec_full: Dial requires an argument (technology/resource)
  == Spawn extension (from-external, 028992018, 18) exited non-zero on 'PJSIP/belgium-voip-0000042d'
<--- Transmitting SIP response (662 bytes) to UDP:188.66.8.19:5060 --->
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 188.66.8.19;rport=5060;received=188.66.8.19;branch=z9hG4bK6092.1c17d19bd75143ad831581f6251e186f.0
Via: SIP/2.0/UDP 188.66.8.52:5060;rport=5060;received=188.66.8.52;branch=z9hG4bK208f6491
Record-Route: <sip:188.66.8.19;lr>
Call-ID: 327ced0e1f5b5bf9347129773ecf5b4c@188.66.8.52:5060
From: "Jane Doe" <sip:042770677@188.66.8.52>;tag=as53b1f9bd
To: <sip:028992018@ip_address>;tag=faf14a39-eea3-4020-89db-ed76ae7324f0
CSeq: 102 INVITE
Server: Asterisk PBX 13.23.0
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Reason: Q.850;cause=0
Content-Length:  0


  == Begin MixMonitor Recording PJSIP/belgium-voip-0000042d
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording PJSIP/belgium-voip-0000042d
<--- Received SIP request (378 bytes) from UDP:188.66.8.19:5060 --->
ACK sip:028992018@ip_address:39748 SIP/2.0
Via: SIP/2.0/UDP 188.66.8.19;branch=z9hG4bK6092.1c17d19bd75143ad831581f6251e186f.0
Max-Forwards: 69
From: "Jane Doe" <sip:042770677@188.66.8.52>;tag=as53b1f9bd
To: <sip:028992018@ip_address>;tag=faf14a39-eea3-4020-89db-ed76ae7324f0
Call-ID: 327ced0e1f5b5bf9347129773ecf5b4c@188.66.8.52:5060
CSeq: 102 ACK
Content-Length: 0


<--- Received SIP response (646 bytes) from UDP:188.66.8.19:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP ip_address:5060;received=ip_address;rport=39748;branch=z9hG4bKPja839b4d7-d17a-4c3f-adf7-f2603a641d6d
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>;tag=as1d37edb5
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22270 INVITE
Server: 3StarsNet VoipSwitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


<--- Transmitting SIP request (442 bytes) to UDP:188.66.8.19:5060 --->
ACK sip:028992018@voip.belgium-voip.com:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPja839b4d7-d17a-4c3f-adf7-f2603a641d6d
From: "Jane Doe" <sip:042770677@voip.belgium-voip.com>;tag=4521b7c5-cbe0-4285-ac33-2e713bec8dc2
To: <sip:028992018@voip.belgium-voip.com>;tag=as1d37edb5
Call-ID: 94db5827-5e60-40ce-b3ba-2f10b822d0ce
CSeq: 22270 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Length:  0


  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [028992018@from-internal:27] Playback("PJSIP/116-0000042b", "cannot-complete-temp-error") in new stack
    -- <PJSIP/116-0000042b> Playing 'cannot-complete-temp-error.ulaw' (language 'fr')
    -- Executing [028992018@from-internal:28] Goto("PJSIP/116-0000042b", "end") in new stack
    -- Goto (from-internal,028992018,29)
    -- Executing [028992018@from-internal:29] Hangup("PJSIP/116-0000042b", "") in new stack
  == Spawn extension (from-internal, 028992018, 29) exited non-zero on 'PJSIP/116-0000042b'
  == MixMonitor close filestream (mixed)
<--- Transmitting SIP request (423 bytes) to UDP:192.168.40.50:5060 --->
BYE sip:116@192.168.40.50:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPj8b81c975-72bc-42e1-b08d-94d40ca7661c
From: <sip:028992018@ip_address>;tag=dba3540b-532b-4289-ab92-fafb39d568bd
To: "Jane Doe" <sip:116@ip_address>;tag=1043865744
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
CSeq: 3805 BYE
Reason: Q.850;cause=17
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Length:  0


  == End MixMonitor Recording PJSIP/116-0000042b
[Oct 10 16:08:21] ERROR[15691]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle.  CDR failed.
[Oct 10 16:08:21] ERROR[15691]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle.  CDR failed.
<--- Received SIP response (534 bytes) from UDP:192.168.40.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip_address:5060;rport=5060;branch=z9hG4bKPj8b81c975-72bc-42e1-b08d-94d40ca7661c
From: <sip:028992018@ip_address>;tag=dba3540b-532b-4289-ab92-fafb39d568bd
To: "Jane Doe" <sip:116@ip_address>;tag=1043865744
Call-ID: 1942712888-5060-36@BJC.BGI.EA.FA
CSeq: 3805 BYE
Contact: <sip:116@192.168.40.50:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.4.128
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP request (575 bytes) from UDP:192.168.2.48:5060 --->
BYE sip:asterisk@ip_address:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.48:5060;branch=z9hG4bK953260347;rport
From: <sip:100@192.168.2.48>;tag=1726513566
To: "John Doe" <sip:042770677@ip_address>;tag=3d9031f7-765c-4152-9625-d9a09dd0a5fc
Call-ID: ce716248-370e-49cb-806f-5f4516f67d9b
CSeq: 28875 BYE
Contact: <sip:100@192.168.2.48:5060>
X-Grandstream-PBX: true
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.9.108
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP response (356 bytes) to UDP:192.168.2.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.48:5060;rport=5060;received=192.168.2.48;branch=z9hG4bK953260347
Call-ID: ce716248-370e-49cb-806f-5f4516f67d9b
From: <sip:100@192.168.2.48>;tag=1726513566
To: "John Doe" <sip:042770677@ip_address>;tag=3d9031f7-765c-4152-9625-d9a09dd0a5fc
CSeq: 28875 BYE
Server: Asterisk PBX 13.23.0
Content-Length:  0


    -- Channel PJSIP/100-0000042a left 'simple_bridge' basic-bridge <c109bca5-3c7d-4af7-9eea-e69f12027367>
    -- Channel PJSIP/belgium-voip-00000428 left 'simple_bridge' basic-bridge <c109bca5-3c7d-4af7-9eea-e69f12027367>
  == Spawn extension (from-external, 028992018, 18) exited non-zero on 'PJSIP/belgium-voip-00000428'
  == MixMonitor close filestream (mixed)
<--- Transmitting SIP request (478 bytes) to UDP:188.66.8.19:5060 --->
BYE sip:042770677@188.66.8.52:5060 SIP/2.0
Via: SIP/2.0/UDP ip_address:5060;rport;branch=z9hG4bKPj642abaa4-4aaf-432a-a686-6adb4787c1d6
From: <sip:028992018@ip_address>;tag=34f021ef-f396-4a35-a69d-18d4e610f64b
To: "John Doe" <sip:042770677@188.66.8.52>;tag=as08e14896
Call-ID: 0a3369672c687c3725cd9e16769f3618@188.66.8.52:5060
CSeq: 29894 BYE
Route: <sip:188.66.8.19;lr>
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 13.23.0
Content-Length:  0

#12

Puedes revisar el archivo de tu sip.conf, debes validar que el transporte sea igual que el de conexión. y que maneje directmedia. que version de asterisk es?


#13

Your dialplan is invalid.


#14

This is happen when I try call the same CID who is connected to the extension 100. And also when this last one is only with another caller.

    -- PJSIP/belgium-voip-0000042c is ringing
    -- PJSIP/belgium-voip-0000042c is ringing
    -- Executing [028992018@from-external:8] Set("PJSIP/belgium-voip-0000042d", "REC_FILE_NAME=IN__028992018_042770677.wav") in new stack
    -- Executing [028992018@from-external:9] MixMonitor("PJSIP/belgium-voip-0000042d", "IN__028992018_042770677.wav,b V(1)") in new stack
    -- Executing [028992018@from-external:10] ChanIsAvail("PJSIP/belgium-voip-0000042d", "PJSIP/100,sa") in new stack
    -- Executing [028992018@from-external:11] Set("PJSIP/belgium-voip-0000042d", "PHONESTATUS=2") in new stack
    -- Executing [028992018@from-external:12] Set("PJSIP/belgium-voip-0000042d", "PHONEAVAIL=") in new stack
    -- Executing [028992018@from-external:13] NoOp("PJSIP/belgium-voip-0000042d", "## Status of device is 2 ##") in new stack
    -- Executing [028992018@from-external:14] GotoIf("PJSIP/belgium-voip-0000042d", "0?busy:call") in new stack
    -- Goto (from-external,028992018,18)
    -- Executing [028992018@from-external:18] Dial("PJSIP/belgium-voip-0000042d", ",20") in new stack
[Oct 10 16:08:17] WARNING[20678][C-00000255]: app_dial.c:2281 dial_exec_full: Dial requires an argument (technology/resource)

I do not know why I have nothing in the variable PHONEAVAIL, this is my syntax for the incoming call on the extension 100:

exten => ${WEBMASTER_IN},1,NoOp(## Incoming Call from ${CALLERID(all)} ##)
 same => n,Verbose(Call start time: ${CDR(start)})
 same => n,Set(CDR(calldate)=${CDR(start)})
 same => n,Set(CDR(useragent)=${CALLERID(name)})
 same => n,Set(POSTE_EXT=${CALLERID(num)})
 same => n,Ringing()
 same => n,System(echo "--appel_sortant --- callerid : ${CALLERID(num)} ---- ${STRFTIME(${EPOCH},,%Y/%m/%d %H:%M:%S)} ----" >> /var/spool/asterisk/log/debug.txt)
 same => n,Set(REC_FILE_NAME=IN_${NOW}_${EXTEN}_${POSTE_EXT}.wav)
 same => n,MixMonitor(${REC_FILE_NAME},b V(1))
 same => n,ChanIsAvail(PJSIP/100,sa)
 same => n,Set(PHONESTATUS=${AVAILSTATUS})
 same => n,Set(PHONEAVAIL=${AVAILORIGCHAN})
 same => n,NoOp(## Status of device is ${PHONESTATUS} ##)
 same => n,GotoIf($["${PHONESTATUS}"="3"]?busy:call)
 same => n(busy),Playback(ivr/REPONDEUR_2)
 same => n,VoiceMail(100@default,s)
 same => n,Goto(end)
 same => n(call),Dial(${PHONEAVAIL},30)
 same => n,VoiceMail(100@default,su)
 same => n,Goto(end)
 same => n(end),Hangup()

But on a direct incoming call this is work well. It’s when a second caller calls the CID of this extension at the moment this one is online that the ${PHONEAVAIL} variable is empty.