Help with .call file, auto dial out, ring just one time

Asterisk Asterisk Support
1

Hi

I have the following issue and I can’t find someone with the same problem here or over internet.
The issue is the following.

I have an asterisk configured with a trunk (Mitel), when I call an external number with the soft phone configured in the asterisk, have a call perfect, and have ring cycle normal, but when I use a .call file, the call out, but the external phone ring just one time.

I’m using the following file test.call and according to wiki.asterisk.org/wiki/display/ … Call+Files the WaitTime option is used to the ring cycle, but this one is not working, just ring one time.

Any help will be appreciated

**** test.call *****

Channel: SIP/7001/external-number
Callerid: test-phone
Context: test-dialplan
WaitTime: 45
MaxRetries: 1
RetryTime: 300
Application: Playback
Data: hello-world

Just FYI

  • If I call to an external number with the soft phone, work fine.
  • If I receive an external call in my extension, work fine.
  • If I call between extensions, work fine.

*************** LOG **************************

-- Attempting call on SIP/7001/9XXXXXXXX for application Playback(hello-world) (Retry 1)

== Using SIP RTP CoS mark 5
Audio is at 13244
Adding codec 100019 (slin) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100013 (siren7) to SDP
Adding codec 100014 (siren14) to SDP
Adding codec 100015 (g719) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100018 (silk8) to SDP
Adding codec 100018 (silk12) to SDP
Adding codec 100018 (silk16) to SDP
Adding codec 100018 (silk24) to SDP
Adding codec 100020 (slin12) to SDP
Adding codec 100021 (slin16) to SDP
Adding codec 100022 (slin24) to SDP
Adding codec 100023 (slin32) to SDP
Adding codec 100024 (slin44) to SDP
Adding codec 100025 (slin48) to SDP
Adding codec 100026 (slin96) to SDP
Adding codec 100027 (slin192) to SDP
Adding codec 100028 (speex32) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to XXX.XXX.XXX.XXX:5060:
INVITE sip:9XXXXXXXX@XXX.XXX.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK093b089d
Max-Forwards: 70
From: “TEST” sip:asterisk@XXX.XXX.XXX.XXX;tag=as1cf96ebb
To: sip:9XXXXXXXX@XXX.XXX.XXX.XXX
Contact: sip:asterisk@XXX.XXX.XXX.XXX:5060
Call-ID: 62eeb2fb3b590f2c4a364e42142592db@XXX.XXX.XXX.XXX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.11.0
Date: Wed, 27 Aug 2014 21:10:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1314

v=0
o=root 1931594334 1931594334 IN IP4 XXX.XXX.XXX.XXX
s=Asterisk PBX 11.11.0
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 13244 RTP/AVP 10 4 3 0 8 112 5 7 18 110 97 111 9 102 115 116 117 96 100 107 108 118 119 101
a=rtpmap:10 L16/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:117 speex/16000
a=rtpmap:96 SILK/8000
a=fmtp:96 maxaveragebitrate=10000
a=fmtp:96 usedtx=0
a=fmtp:96 useinbandfec=1
a=rtpmap:100 SILK/12000
a=fmtp:100 maxaveragebitrate=12000
a=fmtp:100 usedtx=0
a=fmtp:100 useinbandfec=1
a=rtpmap:107 SILK/16000
a=fmtp:107 maxaveragebitrate=20000
a=fmtp:107 usedtx=0
a=fmtp:107 useinbandfec=1
a=rtpmap:108 SILK/24000
a=fmtp:108 maxaveragebitrate=30000
a=fmtp:108 usedtx=0
a=fmtp:108 useinbandfec=1
a=rtpmap:118 L16/16000
a=rtpmap:119 speex/32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:XXX.XXX.XXX.XXX:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK093b089d
From: “TEST” sip:asterisk@XXX.XXX.XXX.XXX;tag=as1cf96ebb
To: sip:9XXXXXXXX@XXX.XXX.XXX.XXX;tag=0_3622416992-331738194
Call-ID: 62eeb2fb3b590f2c4a364e42142592db@XXX.XXX.XXX.XXX:5060
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:XXX.XXX.XXX.XXX:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK093b089d
WWW-Authenticate: Digest realm=“XXX”,nonce=“KO5W(3~AIC’67T|AjH`Bw~14ia>A}?v/Fuj-p]<kV=}*aWe^|.+|7Q}vxN]#Eu#C”,algorithm=MD5,qop="auth"
From: “TEST” sip:asterisk@XXX.XXX.XXX.XXX;tag=as1cf96ebb
To: sip:9XXXXXXXX@XXX.XXX.XXX.XXX;tag=0_3622416992-331738194
Call-ID: 62eeb2fb3b590f2c4a364e42142592db@XXX.XXX.XXX.XXX:5060
CSeq: 102 INVITE
Contact: sip:XXX.XXX.XXX.XXX
Server: Mitel-3300-ICP 12.0.3.15
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Transmitting (no NAT) to XXX.XXX.XXX.XXX:5060:
ACK sip:9XXXXXXXX@XXX.XXX.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK093b089d
Max-Forwards: 70
From: “TEST” sip:asterisk@XXX.XXX.XXX.XXX;tag=as1cf96ebb
To: sip:9XXXXXXXX@XXX.XXX.XXX.XXX;tag=0_3622416992-331738194
Contact: sip:asterisk@XXX.XXX.XXX.XXX:5060
Call-ID: 62eeb2fb3b590f2c4a364e42142592db@XXX.XXX.XXX.XXX:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.11.0
Content-Length: 0

Audio is at 13244
Adding codec 100019 (slin) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100013 (siren7) to SDP
Adding codec 100014 (siren14) to SDP
Adding codec 100015 (g719) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100018 (silk8) to SDP
Adding codec 100018 (silk12) to SDP
Adding codec 100018 (silk16) to SDP
Adding codec 100018 (silk24) to SDP
Adding codec 100020 (slin12) to SDP
Adding codec 100021 (slin16) to SDP
Adding codec 100022 (slin24) to SDP
Adding codec 100023 (slin32) to SDP
Adding codec 100024 (slin44) to SDP
Adding codec 100025 (slin48) to SDP
Adding codec 100026 (slin96) to SDP
Adding codec 100027 (slin192) to SDP
Adding codec 100028 (speex32) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to XXX.XXX.XXX.XXX:5060:
INVITE sip:9XXXXXXXX@XXX.XXX.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4ada7750
Max-Forwards: 70
From: “TEST” sip:asterisk@XXX.XXX.XXX.XXX;tag=as1cf96ebb
To: sip:9XXXXXXXX@XXX.XXX.XXX.XXX
Contact: sip:asterisk@XXX.XXX.XXX.XXX:5060
Call-ID: 62eeb2fb3b590f2c4a364e42142592db@XXX.XXX.XXX.XXX:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.11.0
Authorization: Digest username=“7001”, realm=“XXX”, algorithm=MD5, uri="sip:9XXXXXXXX@XXX.XXX.XXX.XXX", nonce=“KO5W(3~AIC’67T|AjH`Bw~14ia>A}?v/Fuj-p]<kV=}*aWe^|.+|7Q}vxN]#Eu#C”, response=“dee137196b0dc81f86162b1cf94562f8”, qop=auth, cnonce=“1b4ac81d”, nc=00000001
Date: Wed, 27 Aug 2014 21:10:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1314

v=0
o=root 1931594334 1931594335 IN IP4 XXX.XXX.XXX.XXX
s=Asterisk PBX 11.11.0
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 13244 RTP/AVP 10 4 3 0 8 112 5 7 18 110 97 111 9 102 115 116 117 96 100 107 108 118 119 101
a=rtpmap:10 L16/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:117 speex/16000
a=rtpmap:96 SILK/8000
a=fmtp:96 maxaveragebitrate=10000
a=fmtp:96 usedtx=0
a=fmtp:96 useinbandfec=1
a=rtpmap:100 SILK/12000
a=fmtp:100 maxaveragebitrate=12000
a=fmtp:100 usedtx=0
a=fmtp:100 useinbandfec=1
a=rtpmap:107 SILK/16000
a=fmtp:107 maxaveragebitrate=20000
a=fmtp:107 usedtx=0
a=fmtp:107 useinbandfec=1
a=rtpmap:108 SILK/24000
a=fmtp:108 maxaveragebitrate=30000
a=fmtp:108 usedtx=0
a=fmtp:108 useinbandfec=1
a=rtpmap:118 L16/16000
a=rtpmap:119 speex/32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:XXX.XXX.XXX.XXX:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4ada7750
From: “TEST” sip:asterisk@XXX.XXX.XXX.XXX;tag=as1cf96ebb
To: sip:9XXXXXXXX@XXX.XXX.XXX.XXX;tag=0_3622436992-331738195
Call-ID: 62eeb2fb3b590f2c4a364e42142592db@XXX.XXX.XXX.XXX:5060
CSeq: 103 INVITE
Content-Length: 0

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

<— SIP read from UDP:XXX.XXX.XXX.XXX:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4ada7750
From: “TEST” sip:asterisk@XXX.XXX.XXX.XXX;tag=as1cf96ebb
To: sip:9XXXXXXXX@XXX.XXX.XXX.XXX;tag=0_3622436992-331738195
Call-ID: 62eeb2fb3b590f2c4a364e42142592db@XXX.XXX.XXX.XXX:5060
CSeq: 103 INVITE
Contact: sip:9XXXXXXXX@XXX.XXX.XXX.XXX:5060;transport=udp
Server: Mitel-3300-ICP 12.0.3.15
Content-Type: application/sdp
Content-Length: 156

v=0
o=- 3734 3734 IN IP4 XXX.XXX.XXX.XXX
s=-
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 50114 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (10 headers 8 lines) —
list_route: hop: sip:9XXXXXXXX@XXX.XXX.XXX.XXX:5060;transport=udp
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|silk8|silk12|silk16|silk24), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port XXX.XXX.XXX.XXX:50114

<— SIP read from UDP:XXX.XXX.XXX.XXX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4ada7750
From: “TEST” sip:asterisk@XXX.XXX.XXX.XXX;tag=as1cf96ebb
To: sip:9XXXXXXXX@XXX.XXX.XXX.XXX;tag=0_3622436992-331738195
Call-ID: 62eeb2fb3b590f2c4a364e42142592db@XXX.XXX.XXX.XXX:5060
CSeq: 103 INVITE
Contact: sip:9XXXXXXXX@XXX.XXX.XXX.XXX:5060;transport=udp
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
Content-Type: application/sdp
Server: Mitel-3300-ICP 12.0.3.15
Supported: replaces
Content-Length: 156

v=0
o=- 3734 3734 IN IP4 XXX.XXX.XXX.XXX
s=-
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 50114 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (12 headers 8 lines) —
list_route: hop: sip:9XXXXXXXX@XXX.XXX.XXX.XXX:5060;transport=udp
set_destination: Parsing sip:9XXXXXXXX@XXX.XXX.XXX.XXX:5060;transport=udp for address/port to send to
set_destination: set destination to XXX.XXX.XXX.XXX:5060
Transmitting (no NAT) to XXX.XXX.XXX.XXX:5060:
ACK sip:9XXXXXXXX@XXX.XXX.XXX.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK6d78d13c
Max-Forwards: 70
From: “TEST” sip:asterisk@XXX.XXX.XXX.XXX;tag=as1cf96ebb
To: sip:9XXXXXXXX@XXX.XXX.XXX.XXX;tag=0_3622436992-331738195
Contact: sip:asterisk@XXX.XXX.XXX.XXX:5060
Call-ID: 62eeb2fb3b590f2c4a364e42142592db@XXX.XXX.XXX.XXX:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.11.0
Content-Length: 0

-- <SIP/7001-00000034> Playing 'hello-world.alaw' (language 'en')

Scheduling destruction of SIP dialog '62eeb2fb3b590f2c4a364e42142592db@XXX.XXX.XXX.XXX:5060’ in 32000 ms (Method: INVITE)
set_destination: Parsing sip:9XXXXXXXX@XXX.XXX.XXX.XXX:5060;transport=udp for address/port to send to
set_destination: set destination to XXX.XXX.XXX.XXX:5060
Reliably Transmitting (no NAT) to XXX.XXX.XXX.XXX:5060:
BYE sip:9XXXXXXXX@XXX.XXX.XXX.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4d684417
Max-Forwards: 70
From: “TEST” sip:asterisk@XXX.XXX.XXX.XXX;tag=as1cf96ebb
To: sip:9XXXXXXXX@XXX.XXX.XXX.XXX;tag=0_3622436992-331738195
Call-ID: 62eeb2fb3b590f2c4a364e42142592db@XXX.XXX.XXX.XXX:5060
CSeq: 104 BYE
User-Agent: Asterisk PBX 11.11.0
Authorization: Digest username=“7001”, realm=“XXX”, algorithm=MD5, uri=“sip:9XXXXXXXX@XXX.XXX.XXX.XXX:5060”, nonce=“KO5W(3~AIC’67T|AjH`Bw~14ia>A}?v/Fuj-p]<kV=}*aWe^|.+|7Q}vxN]#Eu#C”, response=“462b0c4d399ab829f0a73b394f3f1403”, qop=auth, cnonce=“586694b7”, nc=00000002
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

[Aug 27 15:10:06] NOTICE[12473]: pbx_spool.c:402 attempt_thread: Call completed to SIP/7001/9XXXXXXXX

<— SIP read from UDP:XXX.XXX.XXX.XXX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4d684417
From: “TEST” sip:asterisk@XXX.XXX.XXX.XXX;tag=as1cf96ebb
To: sip:9XXXXXXXX@XXX.XXX.XXX.XXX;tag=0_3622436992-331738195
Call-ID: 62eeb2fb3b590f2c4a364e42142592db@XXX.XXX.XXX.XXX:5060
CSeq: 104 BYE
Contact: sip:9XXXXXXXX@XXX.XXX.XXX.XXX:5060;transport=udp
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
Server: Mitel-3300-ICP 12.0.3.15
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog '62eeb2fb3b590f2c4a364e42142592db@XXX.XXX.XXX.XXX:5060’ Method: INVITE

2

The call was answered.

If that isn’t really true, take it up with your ITSP.

3

david55,
Apparently, but I never take the phone or the call, the phone ring just one time and in the log that says that I answered the call.

But if I dial the external number from the soft phone, the ring cycle is normal on the phone.

So I think that this could be a option or parameter that I miss

4

My guess is that your ITSP is answering the call prematurely, so the parameter to change is the name of your ITSP.

Originate proceeds on the actual answer supervision signal, but the human waits for the ringback to stop.

5

Hi David55

Thanks for your reply, I’m new on this environment and I’m trying to learn this because I need this solution on my work and Asterisk is so powerful on this solution…

But now I’m so lost, where I change the ITSP name?

That term is new to me. Will be very appreciated for all your help

This is my sip.conf

;
; SIP Configuration example for Asterisk
;

[general]
context=inbound ; Default context for incoming calls

udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)

srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host

;localnet=192.168.96.0/255.255.255.0 ; RFC 1918 addresses
;externaddr = 201.195.112.68 ; use this address.
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
register => 7001:70012014@172.20.100.133/7001
disallow = all
allow = ulaw
allow = alaw

[7000]
type=friend
secret=12345
username=7000
context=prueba-dialplan
nat=no
canreinvite=no
host=dynamic
callerid=“Roberto MAC Softphoe” 7000
mailbox=602

[7001]
type=friend
secret=70012014
username=7001
context=prueba-dialplan
nat=no
canreinvite=no
host=dynamic
callerid=“Mitel Phone” 7001
mailbox=7001

This is my extension.conf

; extensions.conf - the Asterisk dial plan
[general]
static=yes
writeprotect=no

; If autofallthrough is set, then if an extension runs out of
clearglobalvars=no

[globals]

CONSOLE=Console/dsp ; Console interface for demo
; Configuracion
;

[prueba-dialplan]
exten => 7000,1,Dial(SIP/7000)
exten => 7001,1,Dial(SIP/7001)
exten => _99ZXXXXXXX,1,Dial(SIP/7001/${EXTEN:1})
exten => _7XXXX,1,Dial(SIP/7001/${EXTEN:1})

[outboundmsg1]
exten => s,1,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,2,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
exten => s,3,Answer
exten => s,4,Wait(1)
exten => s,5,Playback(hello-world)
;exten => s,5,Background(outboundmsgs/msg1) ; “play outbound msg”
;exten => s,6,Background(outboundmsgs/how_to_ack) ; "Press 1 to replay or 2 to acknowledge receiving this message"
exten => 1,1,Goto(s,5) ; replay message
;exten => 2,1,Goto(msgack,s,1) ; acknowledge message
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup

user.conf

;
; User configuration
;
; Creating entries in users.conf is a “shorthand” for creating individual
; entries in each configuration file. Using users.conf is not intended to
; provide you with as much flexibility as using the separate configuration
; files (e.g. sip.conf, iax.conf, etc) but is intended to accelerate the
; simple task of adding users. Note that creating individual items (e.g.
; custom SIP peers, IAX friends, etc.) will allow you to override specific
; parameters within this file. Parameter names here are the same as they
; appear in the other configuration files. There is no way to change the
; value of a parameter here for just one subsystem.
;

[general]
;
; Full name of a user
;
fullname = New User
;
; Starting point of allocation of extensions
;
userbase = 6000
;
; Create voicemail mailbox and use use macro-stdexten
;
hasvoicemail = yes
;
; Set voicemail mailbox 6000 password to 1234
;
vmsecret = 1234
;
; Create SIP Peer
;
hassip = yes
;
; Create IAX friend
;
hasiax = yes
;
; Create H.323 friend
;
;hash323 = yes
;
; Create manager entry
;
hasmanager = no
;
; Set permissions for manager entry (see manager.conf.sample for documentation)
; (defaults to all permissions)
;managerread = system,call,log,verbose,command,agent,user,config
;managerwrite = system,call,log,verbose,command,agent,user,config
;
;
; MAC Address for res_phoneprov
;
;macaddress = 112233445566
;
; Auto provision the phone with res_phoneprov
;
;autoprov = yes
;
; Line Keys for hardphone
;
;LINEKEYS = 1
;
; Line number for hardphone
;
;linenumber = 1
;
; Local Caller ID number used with res_phoneprov and Asterisk GUI
;
;cid_number = 6000
;
; Remaining options are not specific to users.conf entries but are general.
;
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
;nat = no

;[6000]
;fullname = Joe User
;description = Courtesy Phone In Lobby ; Used to provide a description of the
; peer in console output
;email = joe@foo.bar
;secret = 1234
;dahdichan = 1
;hasvoicemail = yes
;vmsecret = 1234
;hassip = yes
;hasiax = no
;hash323 = no
;hasmanager = no
;callwaiting = no
;context = international
;
; Some administrators choose alphanumeric extensions, but still want their
; users to be reachable by traditional numeric extensions, specified by the
; alternateexts entry.
;
;alternateexts = 7057,3249
;macaddress = 112233445566
;autoprov = yes
;LINEKEYS = 1
;linenumber = 1
;cid_number = 6000

[7000]
callwaiting = no
context = prueba-dialplan
email = xxxxxxx
fullname = Roberto MAC Softphone
hasagent = no
hasdirectory = no
hasiax = no
hasmanager = no
hassip = yes
hasvoicemail = yes
deletevoicemail = no
host = dynamic
mailbox = 7000
threewaycalling = no
vmsecret = 8125
registeriax = no
registersip = yes
label = 7000
canreinvite = no
nat = no
dtmfmode = rfc2833
disallow = all
allow = ulaw
allow = alaw
signalling = fxo_ks
secret=12345

[7001]
callwaiting = no
context = prueba-dialplan
email = xxxxxxxxxx
fullname = Mitel Phone
hasagent = no
hasdirectory = no
hasiax = no
hasmanager = no
hassip = yes
hasvoicemail = yes
deletevoicemail = no
host = dynamic
mailbox = 7001
threewaycalling = no
vmsecret = 8125
registeriax = no
registersip = yes
label = 7001
canreinvite = no
nat = no
dtmfmode = rfc2833
disallow = all
allow = ulaw
allow = alaw
signalling = fxo_ks
secret=12345

Thanks for you help again

6

ITSP = Internet Telephony Service Provider

The name is implied by the host=line.

Basically if your ITSP has broken answer supervision handling, the solution is to find one that doesn’t.

Also, please edit your posts to put code blocks around the logs and configuration.

7

I hope this(70012014) is not your actual secret in registration string.
register => 7001:70012014@172.20.100.133/7001

If yes then you are at risk and should change it asap as you have disclosed it on Internet.

Regarding your problem, you should talk to your ITSP on not to answer a call before it actually connects.

–Satish Barot