Problems with asterisk with a call

Asterisk Asterisk Support
1

Someone can help me, I have a switchboard and when I call a particular company, this gives me some options in the IVR and when I type to communicate with a consultant; When entering the queue instead of sounding the music waiting for them or the timbre. I sound the waiting audio of the company I work for.

If someone has had a failure of these to illustrate that I am not seeing in the trace or configuration.

Attached capture trace taken.

[Jul 13 09:17:59]
<— SIP read from UDP:172.16.83.47:65106 —>
INVITE sip:94856666@192.168.100.38 SIP/2.0
Via: SIP/2.0/UDP 172.16.83.47:65106;branch=z9hG4bK-524287-1—ce3b776d6adb8427;rport
Max-Forwards: 70
Contact: sip:9998@172.16.83.47:65106;rinstance=b8d93def1e14e409
To: sip:94856666@192.168.100.38
From: "9998"sip:9998@192.168.100.38;tag=63827f42
Call-ID: 83108YmNhYTY0NDU1NGQ2NjNjYjM2ODVmNjAyMTM4ZmY1YjU
CSeq: 1 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.9.7 stamp 83108
Content-Length: 211

v=0
o=- 13144429098949180 1 IN IP4 172.16.83.47
s=X-Lite release 4.9.7 stamp 83108
c=IN IP4 172.16.83.47
t=0 0
m=audio 51614 RTP/AVP 8 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
[Jul 13 09:17:59] — (13 headers 9 lines) —
[Jul 13 09:17:59] Sending to 172.16.83.47:65106 (no NAT)
[Jul 13 09:17:59] Using INVITE request as basis request - 83108YmNhYTY0NDU1NGQ2NjNjYjM2ODVmNjAyMTM4ZmY1YjU
[Jul 13 09:17:59] Found peer ‘9998’ for ‘9998’ from 172.16.83.47:65106
[Jul 13 09:17:59]
<— Reliably Transmitting (NAT) to 172.16.83.47:65106 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.83.47:65106;branch=z9hG4bK-524287-1—ce3b776d6adb8427;received=172.16.83.47;rport=65106
From: "9998"sip:9998@192.168.100.38;tag=63827f42
To: sip:94856666@192.168.100.38;tag=as582a8d6a
Call-ID: 83108YmNhYTY0NDU1NGQ2NjNjYjM2ODVmNjAyMTM4ZmY1YjU
CSeq: 1 INVITE
Server: Dialvox
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="18aba751"
Content-Length: 0

<------------>
[Jul 13 09:17:59] Scheduling destruction of SIP dialog ‘83108YmNhYTY0NDU1NGQ2NjNjYjM2ODVmNjAyMTM4ZmY1YjU’ in 6400 ms (Method: INVITE)
[Jul 13 09:17:59]
<— SIP read from UDP:172.16.83.47:65106 —>
ACK sip:94856666@192.168.100.38 SIP/2.0
Via: SIP/2.0/UDP 172.16.83.47:65106;branch=z9hG4bK-524287-1—ce3b776d6adb8427;rport
Max-Forwards: 70
To: sip:94856666@192.168.100.38;tag=as582a8d6a
From: "9998"sip:9998@192.168.100.38;tag=63827f42
Call-ID: 83108YmNhYTY0NDU1NGQ2NjNjYjM2ODVmNjAyMTM4ZmY1YjU
CSeq: 1 ACK
Content-Length: 0

<------------->
[Jul 13 09:17:59] — (8 headers 0 lines) —
[Jul 13 09:17:59]
<— SIP read from UDP:172.16.83.47:65106 —>
INVITE sip:94856666@192.168.100.38 SIP/2.0
Via: SIP/2.0/UDP 172.16.83.47:65106;branch=z9hG4bK-524287-1—a336aa54b8e4a908;rport
Max-Forwards: 70
Contact: sip:9998@172.16.83.47:65106;rinstance=b8d93def1e14e409
To: sip:94856666@192.168.100.38
From: “9998"sip:9998@192.168.100.38;tag=63827f42
Call-ID: 83108YmNhYTY0NDU1NGQ2NjNjYjM2ODVmNjAyMTM4ZmY1YjU
CSeq: 2 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.9.7 stamp 83108
Authorization: Digest username=“9998”,realm=“asterisk”,nonce=“18aba751”,uri="sip:94856666@192.168.100.38”,response=“bccd12fe91fca8fc1ff58337eb063c85”,algorithm=MD5
Content-Length: 211

v=0
o=- 13144429098949180 1 IN IP4 172.16.83.47
s=X-Lite release 4.9.7 stamp 83108
c=IN IP4 172.16.83.47
t=0 0
m=audio 51614 RTP/AVP 8 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
[Jul 13 09:17:59] — (14 headers 9 lines) —
[Jul 13 09:17:59] Sending to 172.16.83.47:65106 (NAT)
[Jul 13 09:17:59] Using INVITE request as basis request - 83108YmNhYTY0NDU1NGQ2NjNjYjM2ODVmNjAyMTM4ZmY1YjU
[Jul 13 09:17:59] Found peer ‘9998’ for ‘9998’ from 172.16.83.47:65106
[Jul 13 09:17:59] Found RTP audio format 8
[Jul 13 09:17:59] Found RTP audio format 0
[Jul 13 09:17:59] Found RTP audio format 101
[Jul 13 09:17:59] Found audio description format telephone-event for ID 101
[Jul 13 09:17:59] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Jul 13 09:17:59] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jul 13 09:17:59] Peer audio RTP is at port 172.16.83.47:51614
[Jul 13 09:17:59] Looking for 94856666 in admin (domain 192.168.100.38)
[Jul 13 09:17:59] list_route: hop: sip:9998@172.16.83.47:65106;rinstance=b8d93def1e14e409
[Jul 13 09:17:59]
<— Transmitting (NAT) to 172.16.83.47:65106 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.83.47:65106;branch=z9hG4bK-524287-1—a336aa54b8e4a908;received=172.16.83.47;rport=65106
From: "9998"sip:9998@192.168.100.38;tag=63827f42
To: sip:94856666@192.168.100.38
Call-ID: 83108YmNhYTY0NDU1NGQ2NjNjYjM2ODVmNjAyMTM4ZmY1YjU
CSeq: 2 INVITE
Server: Dialvox
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:94856666@192.168.100.38:5060
Content-Length: 0

<------------>
[Jul 13 09:17:59]
<— Transmitting (NAT) to 172.16.83.47:65106 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.83.47:65106;branch=z9hG4bK-524287-1—a336aa54b8e4a908;received=172.16.83.47;rport=65106
From: "9998"sip:9998@192.168.100.38;tag=63827f42
To: sip:94856666@192.168.100.38;tag=as7a22be34
Call-ID: 83108YmNhYTY0NDU1NGQ2NjNjYjM2ODVmNjAyMTM4ZmY1YjU
CSeq: 2 INVITE
Server: Dialvox
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:94856666@192.168.100.38:5060
Content-Length: 0

<------------>
[Jul 13 09:18:00] Audio is at 17406
[Jul 13 09:18:00] Adding codec 0x8 (alaw) to SDP
[Jul 13 09:18:00] Adding codec 0x4 (ulaw) to SDP
[Jul 13 09:18:00] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 13 09:18:00]
<— Reliably Transmitting (NAT) to 172.16.83.47:65106 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.83.47:65106;branch=z9hG4bK-524287-1—a336aa54b8e4a908;received=172.16.83.47;rport=65106
From: "9998"sip:9998@192.168.100.38;tag=63827f42
To: sip:94856666@192.168.100.38;tag=as7a22be34
Call-ID: 83108YmNhYTY0NDU1NGQ2NjNjYjM2ODVmNjAyMTM4ZmY1YjU
CSeq: 2 INVITE
Server: Dialvox
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:94856666@192.168.100.38:5060
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 864960487 864960487 IN IP4 192.168.100.38
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.100.38
t=0 0
m=audio 17406 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Jul 13 09:18:00]
<— SIP read from UDP:172.16.83.47:65106 —>
ACK sip:94856666@192.168.100.38:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.83.47:65106;branch=z9hG4bK-524287-1—2371625bbf87dc37;rport
Max-Forwards: 70
Contact: sip:9998@172.16.83.47:65106;rinstance=b8d93def1e14e409
To: sip:94856666@192.168.100.38;tag=as7a22be34
From: "9998"sip:9998@192.168.100.38;tag=63827f42
Call-ID: 83108YmNhYTY0NDU1NGQ2NjNjYjM2ODVmNjAyMTM4ZmY1YjU
CSeq: 2 ACK
User-Agent: X-Lite release 4.9.7 stamp 83108
Content-Length: 0

<------------->
[Jul 13 09:18:00] — (10 headers 0 lines) —
[Jul 13 09:18:01] NOTICE[12441]: pbx.c:4299 pbx_extension_helper: No such label ‘nonlp’ in extension ‘s’ in context ‘macro-notifica-perdida’
[Jul 13 09:18:02] Reliably Transmitting (NAT) to 172.16.83.47:65106:
OPTIONS sip:9998@172.16.83.47:65106;rinstance=b8d93def1e14e409 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.38:5060;branch=z9hG4bK1b454e74;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.100.38;tag=as28772d56
To: sip:9998@172.16.83.47:65106;rinstance=b8d93def1e14e409
Contact: sip:asterisk@192.168.100.38:5060
Call-ID: 7c890ea3772db52559a2540516326059@192.168.100.38:5060
CSeq: 102 OPTIONS
User-Agent: Dialvox
Date: Thu, 13 Jul 2017 14:18:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

[Jul 13 09:18:02]
<— SIP read from UDP:172.16.83.47:65106 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.38:5060;branch=z9hG4bK1b454e74;rport=5060
Contact: sip:172.16.83.47:65106
To: sip:9998@172.16.83.47:65106;rinstance=b8d93def1e14e409;tag=c101c052
From: “asterisk” sip:asterisk@192.168.100.38;tag=as28772d56
Call-ID: 7c890ea3772db52559a2540516326059@192.168.100.38:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Supported: replaces
User-Agent: X-Lite release 4.9.7 stamp 83108
Allow-Events: talk, hold
Content-Length: 0

<------------->
[Jul 13 09:18:02] — (14 headers 0 lines) —
[Jul 13 09:18:02] Really destroying SIP dialog ‘7c890ea3772db52559a2540516326059@192.168.100.38:5060’ Method: OPTIONS
[Jul 13 09:18:02] NOTICE[13070]: pbx.c:4299 pbx_extension_helper: No such label ‘nonlp’ in extension ‘s’ in context ‘macro-notifica-perdida’
[Jul 13 09:18:04] NOTICE[12541]: pbx.c:4299 pbx_extension_helper: No such label ‘nonlp’ in extension ‘s’ in context ‘macro-notifica-perdida’
[Jul 13 09:18:07] NOTICE[12560]: pbx.c:4299 pbx_extension_helper: No such label ‘nonlp’ in extension ‘s’ in context ‘macro-notifica-perdida’
[Jul 13 09:18:08] DTMF[13041]: channel.c:4119 __ast_read: DTMF begin ‘1’ received on SIP/9998-000907ae
[Jul 13 09:18:08] DTMF[13041]: channel.c:4129 __ast_read: DTMF begin passthrough ‘1’ on SIP/9998-000907ae
[Jul 13 09:18:09] DTMF[13041]: channel.c:4034 __ast_read: DTMF end ‘1’ received on SIP/9998-000907ae, duration 160 ms
[Jul 13 09:18:09] DTMF[13041]: channel.c:4074 __ast_read: DTMF end accepted with begin ‘1’ on SIP/9998-000907ae
[Jul 13 09:18:09] DTMF[13041]: channel.c:4103 __ast_read: DTMF end passthrough ‘1’ on SIP/9998-000907ae

[Jul 13 09:18:16] DTMF[13041]: channel.c:4119 __ast_read: DTMF begin ‘1’ received on SIP/9998-000907ae
[Jul 13 09:18:16] DTMF[13041]: channel.c:4129 __ast_read: DTMF begin passthrough ‘1’ on SIP/9998-000907ae
[Jul 13 09:18:16] DTMF[13041]: channel.c:4034 __ast_read: DTMF end ‘1’ received on SIP/9998-000907ae, duration 160 ms
[Jul 13 09:18:16] DTMF[13041]: channel.c:4074 __ast_read: DTMF end accepted with begin ‘1’ on SIP/9998-000907ae
[Jul 13 09:18:16] DTMF[13041]: channel.c:4103 __ast_read: DTMF end passthrough ‘1’ on SIP/9998-000907ae
[Jul 13 09:18:20] DTMF[13041]: channel.c:4119 __ast_read: DTMF begin ‘6’ received on SIP/9998-000907ae
[Jul 13 09:18:20] DTMF[13041]: channel.c:4129 __ast_read: DTMF begin passthrough ‘6’ on SIP/9998-000907ae
[Jul 13 09:18:20] DTMF[13041]: channel.c:4034 __ast_read: DTMF end ‘6’ received on SIP/9998-000907ae, duration 160 ms
[Jul 13 09:18:20] DTMF[13041]: channel.c:4074 __ast_read: DTMF end accepted with begin ‘6’ on SIP/9998-000907ae
[Jul 13 09:18:20] DTMF[13041]: channel.c:4103 __ast_read: DTMF end passthrough ‘6’ on SIP/9998-000907ae

<— SIP read from UDP:172.16.83.47:65106 —>

<------------->
[Jul 13 09:18:24] DTMF[13041]: channel.c:4119 __ast_read: DTMF begin ‘1’ received on SIP/9998-000907ae
[Jul 13 09:18:24] DTMF[13041]: channel.c:4129 __ast_read: DTMF begin passthrough ‘1’ on SIP/9998-000907ae
[Jul 13 09:18:24] DTMF[13041]: channel.c:4034 __ast_read: DTMF end ‘1’ received on SIP/9998-000907ae, duration 160 ms
[Jul 13 09:18:24] DTMF[13041]: channel.c:4074 __ast_read: DTMF end accepted with begin ‘1’ on SIP/9998-000907ae
[Jul 13 09:18:24] DTMF[13041]: channel.c:4103 __ast_read: DTMF end passthrough ‘1’ on SIP/9998-000907ae
<— SIP read from UDP:172.16.83.47:65106 —>

[Jul 13 09:19:02] Reliably Transmitting (NAT) to 172.16.83.47:65106:
OPTIONS sip:9998@172.16.83.47:65106;rinstance=b8d93def1e14e409 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.38:5060;branch=z9hG4bK2940d3b9;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.100.38;tag=as35cf17ce
To: sip:9998@172.16.83.47:65106;rinstance=b8d93def1e14e409
Contact: sip:asterisk@192.168.100.38:5060
Call-ID: 68e7814c12984fc61fc8d9f3692afff4@192.168.100.38:5060
CSeq: 102 OPTIONS
User-Agent: Dialvox
Date: Thu, 13 Jul 2017 14:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:172.16.83.47:65106 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.38:5060;branch=z9hG4bK2940d3b9;rport=5060
Contact: sip:172.16.83.47:65106
To: sip:9998@172.16.83.47:65106;rinstance=b8d93def1e14e409;tag=48f5051b
From: “asterisk” sip:asterisk@192.168.100.38;tag=as35cf17ce
Call-ID: 68e7814c12984fc61fc8d9f3692afff4@192.168.100.38:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Supported: replaces
User-Agent: X-Lite release 4.9.7 stamp 83108
Allow-Events: talk, hold
Content-Length: 0

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

[Jul 13 09:19:20]
<— SIP read from UDP:172.16.83.47:65106 —>
BYE sip:94856666@192.168.100.38:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.83.47:65106;branch=z9hG4bK-524287-1—8c38ec2d48824b54;rport
Max-Forwards: 70
Contact: sip:9998@172.16.83.47:65106;rinstance=b8d93def1e14e409
To: sip:94856666@192.168.100.38;tag=as7a22be34
From: "9998"sip:9998@192.168.100.38;tag=63827f42
Call-ID: 83108YmNhYTY0NDU1NGQ2NjNjYjM2ODVmNjAyMTM4ZmY1YjU
CSeq: 3 BYE
User-Agent: X-Lite release 4.9.7 stamp 83108
Authorization: Digest username=“9998”,realm=“asterisk”,nonce=“18aba751”,uri=“sip:94856666@192.168.100.38:5060”,response=“0b789d02fa78daa1e019c30592aef5a8”,algorithm=MD5
Content-Length: 0

<------------->
[Jul 13 09:19:20] — (11 headers 0 lines) —
[Jul 13 09:19:20] Sending to 172.16.83.47:65106 (NAT)
[Jul 13 09:19:20] Scheduling destruction of SIP dialog ‘83108YmNhYTY0NDU1NGQ2NjNjYjM2ODVmNjAyMTM4ZmY1YjU’ in 6400 ms (Method: BYE)
[Jul 13 09:19:20]
<— Transmitting (NAT) to 172.16.83.47:65106 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.83.47:65106;branch=z9hG4bK-524287-1—8c38ec2d48824b54;received=172.16.83.47;rport=65106
From: "9998"sip:9998@192.168.100.38;tag=63827f42
To: sip:94856666@192.168.100.38;tag=as7a22be34
Call-ID: 83108YmNhYTY0NDU1NGQ2NjNjYjM2ODVmNjAyMTM4ZmY1YjU
CSeq: 3 BYE
Server: Dialvox
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

svrasterisk*CLI>

2

Your log doesn’t show the B side of the call.

You have not use </> so your log is garbled.

If the B side used the SIP procedure to put you on hold, what you describe would be the expected behaviour.

3

In several revisions that I made detect that I enter the music on hold and it stops, this is when I make the call and the rare thing is that the music in waiting that sounds is mine but not the music of the company that I call.

[Jul 13 11:21:25] – Executing [s@macro-dial:40] Dial(“SIP/9998-00091341”, “SIP/Claro/4856666,tTrkKjL(14400000:60000:20000)”) in new stack
[Jul 13 11:21:25] – SIP/Claro-00091342 is making progress passing it to SIP/9998-00091341
[Jul 13 11:21:25] – SIP/Claro-00091342 answered SIP/9998-00091341
[Jul 13 11:21:25] DEBUG[2551]: channel.c:6339 ast_set_owners_and_peers: setting peeraccount to M for SIP/Claro-00091342 from data on channel SIP/9998-00091341
[Jul 13 11:21:25] DEBUG[2551]: channel.c:6344 ast_set_owners_and_peers: setting peeraccount to M for SIP/9998-00091341 from data on channel SIP/Claro-00091342
[Jul 13 11:21:46] DTMF[2551]: channel.c:4119 __ast_read: DTMF begin ‘1’ received on SIP/9998-00091341
[Jul 13 11:21:46] DTMF[2551]: channel.c:4129 __ast_read: DTMF begin passthrough ‘1’ on SIP/9998-00091341
[Jul 13 11:21:46] DTMF[2551]: channel.c:4034 __ast_read: DTMF end ‘1’ received on SIP/9998-00091341, duration 160 ms
[Jul 13 11:21:46] DTMF[2551]: channel.c:4074 __ast_read: DTMF end accepted with begin ‘1’ on SIP/9998-00091341
[Jul 13 11:21:46] DTMF[2551]: channel.c:4103 __ast_read: DTMF end passthrough ‘1’ on SIP/9998-00091341
[Jul 13 11:21:50] DTMF[2551]: channel.c:4119 __ast_read: DTMF begin ‘6’ received on SIP/9998-00091341
[Jul 13 11:21:50] DTMF[2551]: channel.c:4129 __ast_read: DTMF begin passthrough ‘6’ on SIP/9998-00091341
[Jul 13 11:21:51] DTMF[2551]: channel.c:4034 __ast_read: DTMF end ‘6’ received on SIP/9998-00091341, duration 160 ms
[Jul 13 11:21:51] DTMF[2551]: channel.c:4074 __ast_read: DTMF end accepted with begin ‘6’ on SIP/9998-00091341
[Jul 13 11:21:51] DTMF[2551]: channel.c:4103 __ast_read: DTMF end passthrough ‘6’ on SIP/9998-00091341
[Jul 13 11:21:55] DTMF[2551]: channel.c:4119 __ast_read: DTMF begin ‘1’ received on SIP/9998-00091341
[Jul 13 11:21:55] DTMF[2551]: channel.c:4129 __ast_read: DTMF begin passthrough ‘1’ on SIP/9998-00091341
[Jul 13 11:21:55] DTMF[2551]: channel.c:4034 __ast_read: DTMF end ‘1’ received on SIP/9998-00091341, duration 160 ms
[Jul 13 11:21:55] DTMF[2551]: channel.c:4074 __ast_read: DTMF end accepted with begin ‘1’ on SIP/9998-00091341
[Jul 13 11:21:55] DTMF[2551]: channel.c:4103 __ast_read: DTMF end passthrough ‘1’ on SIP/9998-00091341
[Jul 13 11:22:08] – Started music on hold, class ‘default’, on SIP/9998-00091341
[Jul 13 11:22:08] – Stopped music on hold on SIP/9998-00091341
[Jul 13 11:22:20] – Executing [h@local_salidadialvox:1] NoOp(“SIP/9998-00091341”, "ANSWER : ") in new stack
[Jul 13 11:22:20] – Executing [h@local_salidadialvox:2] ExecIf(“SIP/9998-00091341”, “1?Set(CDR(status)=CONTESTADA)”) in new stack
[Jul 13 11:22:20] – Executing [h@local_salidadialvox:3] GotoIf(“SIP/9998-00091341”, “1?hangup”) in new stack
[Jul 13 11:22:20] – Executing [h@local_salidadialvox:12] Hangup(“SIP/9998-00091341”, “”) in new stack
[Jul 13 11:22:20] == Spawn extension (local_salidadialvox, h, 12) exited non-zero on ‘SIP/9998-00091341’
[Jul 13 11:22:20] == Spawn extension (macro-dial, s, 40) exited non-zero on ‘SIP/9998-00091341’ in macro ‘dial’
[Jul 13 11:22:20] == Spawn extension (local_salidadialvox, 94856666, 59) exited non-zero on ‘SIP/9998-00091341’
[Jul 13 11:22:20] == End MixMonitor Recording SIP/9998-00091341

This is the log that comes out in my asterisk.

Any help is welcome thanks.