Y este es el log Sip debug:
Y respondiendo a tu pregunta actualmente mando los dos audios (dmsg y dmsg1), tal como esta en el ejemplo, de hecho uso los mismos que ya estan actualmente en el repositorio.
-- Attempting call on LOCAL/s@dialer for application AGI(dialer.agi,1,test13 ,35184527,1) (Retry 1)
-- Executing [s@dialer:1] NoOp("Local/s@dialer-00000080;2", ""Iniciando llam ada Dialer01 a 35184527") in new stack
-- Executing [s@dialer:2] Set("Local/s@dialer-00000080;2", "CDR(accountcode) =DIALER") in new stack
-- Executing [s@dialer:3] Set("Local/s@dialer-00000080;2", "CDR(userfield)=35184527") in new stack
-- Executing [s@dialer:4] Dial("Local/s@dialer-00000080;2", "SIP/Movistar/35 184527") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 12432
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 1.1.1.1:5060:
INVITE sip:35184527@1.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK5ccaae8a;rport
Max-Forwards: 70
From: "Unknown" <sip:22222222@1.1.1.1>;tag=as0681c7a9
To: <sip:35184527@1.1.1.1>
Contact: <sip:22222222@2.2.2.2:5060>
Call-ID: 5f42fc173992af9b451317327f3ef5b3@1.1.1.1
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.25.0)
Date: Thu, 28 Sep 2017 08:19:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 266
v=0
o=root 1362864148 1362864148 IN IP4 2.2.2.2
s=Asterisk PBX 11.25.0
c=IN IP4 2.2.2.2
t=0 0
m=audio 12432 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
-- Called SIP/Movistar/35184527
<--- SIP read from UDP:1.1.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK5ccaae8a;rport=5060
Call-ID: 5f42fc173992af9b451317327f3ef5b3@1.1.1.1
From: "Unknown"<sip:22222222@1.1.1.1>;tag=as0681c7a9
To: <sip:35184527@1.1.1.1>
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
<--- SIP read from UDP:1.1.1.1:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK5ccaae8a;rport=5060
Call-ID: 5f42fc173992af9b451317327f3ef5b3@1.1.1.1
From: "Unknown"<sip:22222222@1.1.1.1>;tag=as0681c7a9
To: <sip:35184527@1.1.1.1>;tag=rdtdrwvv-CC-42
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE ,MESSAGE,REFER
Contact: <sip:1.1.1.1:5060;user=phone>
Content-Length: 221
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 266279 266280 IN IP4 1.1.1.1
s=Sip Call
c=IN IP4 1.1.1.1
t=0 0
m=audio 17634 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=fmtp:101 0-15
<------------->
--- (10 headers 10 lines) ---
list_route: hop: <sip:1.1.1.1:5060;user=phone>
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothin g), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon e-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:17634
-- SIP/Movistar-00000096 is ringing
-- SIP/Movistar-00000096 is making progress passing it to Local/s@dialer-000 00080;2
> 0x7f9fec0158b0 -- Probation passed - setting RTP source address to 1.1.1.1:17634
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
<--- SIP read from UDP:1.1.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK5ccaae8a;rport=5060
Call-ID: 5f42fc173992af9b451317327f3ef5b3@1.1.1.1
From: "Unknown"<sip:22222222@1.1.1.1>;tag=as0681c7a9
To: <sip:35184527@1.1.1.1>;tag=rdtdrwvv-CC-42
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE ,MESSAGE,REFER
Require: timer
Session-Expires: 300;refresher=uac
Contact: <sip:1.1.1.1:5060;user=phone>
Content-Length: 221
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 266279 266280 IN IP4 1.1.1.1
s=Sip Call
c=IN IP4 1.1.1.1
t=0 0
m=audio 17634 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=fmtp:101 0-15
<------------->
--- (12 headers 10 lines) ---
list_route: hop: <sip:1.1.1.1:5060;user=phone>
set_destination: Parsing <sip:1.1.1.1:5060;user=phone> for address/port to send to
set_destination: set destination to 1.1.1.1:5060
Transmitting (NAT) to 1.1.1.1:5060:
ACK sip:1.1.1.1:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK5670a4f6;rport
Max-Forwards: 70
From: "Unknown" <sip:22222222@1.1.1.1>;tag=as0681c7a9
To: <sip:35184527@1.1.1.1>;tag=rdtdrwvv-CC-42
Contact: <sip:22222222@2.2.2.2:5060>
Call-ID: 5f42fc173992af9b451317327f3ef5b3@1.1.1.1
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.25.0)
Content-Length: 0
-- SIP/Movistar-00000096 answered Local/s@dialer-00000080;2
> Channel Local/s@dialer-00000080;1 was answered.
> Launching AGI(dialer.agi,1,test13,35184527,1) on Local/s@dialer-0000008 0;1
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialer.agi
== Spawn extension (dialer, s, 4) exited non-zero on 'Local/s@dialer-00000080; 2'
-- Remote UNIX connection
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- <SIP/Movistar-00000096> Playing '/var/lib/asterisk/agi-bin/DialerCamps/te st13/sounds/dmsg.gsm' (language 'en')
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- <SIP/Movistar-00000096> Playing '/var/lib/asterisk/agi-bin/DialerCamps/te st13/sounds/dmsg1.gsm' (escape_digits=) (sample_offset 0) (language 'en')
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- <SIP/Movistar-00000096>AGI Script dialer.agi completed, returning 4
Scheduling destruction of SIP dialog '5f42fc173992af9b451317327f3ef5b3@1.1.1.1' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:1.1.1.1:5060;user=phone> for address/port to send to
set_destination: set destination to 1.1.1.1:5060
Reliably Transmitting (NAT) to 1.1.1.1:5060:
BYE sip:1.1.1.1:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK27f1d074;rport
Max-Forwards: 70
From: "Unknown" <sip:22222222@1.1.1.1>;tag=as0681c7a9
To: <sip:35184527@1.1.1.1>;tag=rdtdrwvv-CC-42
Call-ID: 5f42fc173992af9b451317327f3ef5b3@1.1.1.1
CSeq: 103 BYE
User-Agent: FPBX-2.11.0(11.25.0)
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0
[2017-09-28 04:20:42] NOTICE[7673]: pbx_spool.c:427 attempt_thread: Call complet ed to LOCAL/s@dialer
<--- SIP read from UDP:1.1.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK27f1d074;rport=5060
Call-ID: 5f42fc173992af9b451317327f3ef5b3@1.1.1.1
From: "Unknown"<sip:22222222@1.1.1.1>;tag=as0681c7a9
To: <sip:35184527@1.1.1.1>;tag=rdtdrwvv-CC-42
CSeq: 103 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '5f42fc173992af9b451317327f3ef5b3@1.1.1.1' Method: INVITE
Desde ya agradezco tu apoyo.