No se reproduce ningún sonido (hello-world)

[jairgomez03]

Hola a todos, desde ya les comento que estoy comenzando a aprender Asterisk, ando loco con un tema, cuando marco de un softphone a otro no genera ningún sonido, he realizado el típico ejemplo de Playback(hello-world) y VoiceMail y no se escucha nada, he investigado y me dicen que NAT, PRT, Codecs etc pero no encuentro solución.
He instalado un AsteriskNOW sin FreePBX, en sip.conf tengo habilitados los codecs ulaw, alaw, gsm.
Alguien me puede dar una ayuda, desde ya muchas gracias.

[Christian Cabrera]

Activa el comando sip set debug on antes de que hagas tu llamada, y luego péganos el resultado de lo que aparece en tu consola cuando este error se presenta.

[jairgomez03]

Cordial saludo Christian, muchas gracias por tu pronta respuesta, esto es lo que me envía después de activar sip set debug on al realizar la llamada. Muchas gracias por tu ayuda.

<--- SIP read from UDP:190.14.228.30:8606 --->
INVITE sip:102@168.61.37.87 SIP/2.0
Via: SIP/2.0/UDP 10.254.246.30:5060;rport;branch=z9hG4bKPj13e463a7-fc64-42d7-b20e-01817071d1ba
Max-Forwards: 70
From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
To: <sip:102@168.61.37.87>
Contact: <sip:jgomez@190.14.228.30:8606>
Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
CSeq: 21054 INVITE
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
upported: replaces, 100rel
Content-Type: application/sdp
Content-Length: 363

v=0
o=LGX110 3600613138 0 IN IP4 10.254.246.30
s=sflphone
c=IN IP4 10.254.246.30
t=0 0
m=audio 13202 RTP/AVP 3 0 8 9 110 111 112
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:112 speex/32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 16 lines) ---
Sending to 190.14.228.30:8606 (NAT)
Using INVITE request as basis request - f6eb303b-224b-4f7a-8bc9-2d56348d28c6
Found peer 'jgomez' for 'jgomez' from 190.14.228.30:8606

<--- Reliably Transmitting (NAT) to 190.14.228.30:8606 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj13e463a7-fc64-42d7-b20e-01817071d1ba;received=190.14.228.30;rport=8606
From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
To: <sip:102@168.61.37.87>;tag=as3ede4952
Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
CSeq: 21054 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="10315bb8"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'f6eb303b-224b-4f7a-8bc9-2d56348d28c6' in 7680 ms (Method: INVITE)

<--- SIP read from UDP:190.14.228.30:8606 --->
ACK sip:102@168.61.37.87 SIP/2.0
Via: SIP/2.0/UDP 10.254.246.30:5060;rport;branch=z9hG4bKPj13e463a7-fc64-42d7-b20e-01817071d1ba
Max-Forwards: 70
From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
To: <sip:102@168.61.37.87>;tag=as3ede4952
Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
CSeq: 21054 ACK
Content-Length: 0

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

<--- SIP read from UDP:190.14.228.30:8606 --->
INVITE sip:102@168.61.37.87 SIP/2.0
Via: SIP/2.0/UDP 10.254.246.30:5060;rport;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223
Max-Forwards: 70
From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
To: <sip:102@168.61.37.87>
Contact: <sip:jgomez@190.14.228.30:8606>
Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
CSeq: 21055 INVITE
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
upported: replaces, 100rel
Authorization: Digest username="jgomez", realm="asterisk", nonce="10315bb8", uri="sip:102@168.61.37.87", response="6355e94d5ec1eabe381c268ca1afab53", algorithm=MD5
Content-Type: application/sdp
Content-Length: 363

Continua...

[jairgomez03]

v=0
o=LGX110 3600613138 0 IN IP4 10.254.246.30
s=sflphone
c=IN IP4 10.254.246.30
t=0 0
m=audio 13202 RTP/AVP 3 0 8 9 110 111 112
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:112 speex/32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (13 headers 16 lines) ---
Sending to 190.14.228.30:8606 (NAT)
Using INVITE request as basis request - f6eb303b-224b-4f7a-8bc9-2d56348d28c6
Found peer 'jgomez' for 'jgomez' from 190.14.228.30:8606
== Using SIP RTP CoS mark 5
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 112
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format speex for ID 110
Found audio description format speex for ID 111
Found unknown media description format speex for ID 112
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x20000120e (gsm|ulaw|alaw|speex|speex16|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.254.246.30:13202
Looking for 102 in users (domain 168.61.37.87)
list_route: hop: <sip:jgomez@190.14.228.30:8606>

<--- Transmitting (NAT) to 190.14.228.30:8606 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
To: <sip:102@168.61.37.87>
Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
CSeq: 21055 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:102@10.74.234.47:5060>
Content-Length: 0


<------------>
-- Executing [102@users:1] Answer("SIP/jgomez-00000006", "") in new stack
Audio is at 10908
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 190.14.228.30:8606 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
To: <sip:102@168.61.37.87>;tag=as30d2c9c8
Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
CSeq: 21055 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:102@10.74.234.47:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1533468499 1533468499 IN IP4 10.74.234.47
s=Asterisk PBX 1.8.25.0
c=IN IP4 10.74.234.47
t=0 0
m=audio 10908 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Retransmitting #1 (NAT) to 190.14.228.30:8606:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
To: <sip:102@168.61.37.87>;tag=as30d2c9c8
Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
CSeq: 21055 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:102@10.74.234.47:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1533468499 1533468499 IN IP4 10.74.234.47
s=Asterisk PBX 1.8.25.0
c=IN IP4 10.74.234.47
t=0 0
m=audio 10908 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #2 (NAT) to 190.14.228.30:8606:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
To: <sip:102@168.61.37.87>;tag=as30d2c9c8
Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
CSeq: 21055 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:102@10.74.234.47:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1533468499 1533468499 IN IP4 10.74.234.47
s=Asterisk PBX 1.8.25.0
c=IN IP4 10.74.234.47
t=0 0
m=audio 10908 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[jairgomez03]

---
-- Executing [102@users:2] Playback("SIP/jgomez-00000006", "hello-world") in new stack
-- <SIP/jgomez-00000006> Playing 'hello-world.gsm' (language 'es')
Retransmitting #3 (NAT) to 190.14.228.30:8606:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
To: <sip:102@168.61.37.87>;tag=as30d2c9c8
Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
CSeq: 21055 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:102@10.74.234.47:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1533468499 1533468499 IN IP4 10.74.234.47
s=Asterisk PBX 1.8.25.0
c=IN IP4 10.74.234.47
t=0 0
m=audio 10908 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Executing [102@users:3] Hangup("SIP/jgomez-00000006", "") in new stack
== Spawn extension (users, 102, 3) exited non-zero on 'SIP/jgomez-00000006'
Scheduling destruction of SIP dialog 'f6eb303b-224b-4f7a-8bc9-2d56348d28c6' in 7680 ms (Method: INVITE)
Retransmitting #4 (NAT) to 190.14.228.30:8606:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
To: <sip:102@168.61.37.87>;tag=as30d2c9c8
Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
CSeq: 21055 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:102@10.74.234.47:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1533468499 1533468499 IN IP4 10.74.234.47
s=Asterisk PBX 1.8.25.0
c=IN IP4 10.74.234.47
t=0 0
m=audio 10908 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #5 (NAT) to 190.14.228.30:8606:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
To: <sip:102@168.61.37.87>;tag=as30d2c9c8
Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
CSeq: 21055 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:102@10.74.234.47:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1533468499 1533468499 IN IP4 10.74.234.47
s=Asterisk PBX 1.8.25.0
c=IN IP4 10.74.234.47
t=0 0
m=audio 10908 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Really destroying SIP dialog 'c40fcb53-fabc-4d10-8440-a3ad56f08612' Method: REGISTER
Retransmitting #6 (NAT) to 190.14.228.30:8606:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
To: <sip:102@168.61.37.87>;tag=as30d2c9c8
Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
CSeq: 21055 INVITE
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:102@10.74.234.47:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1533468499 1533468499 IN IP4 10.74.234.47
s=Asterisk PBX 1.8.25.0
c=IN IP4 10.74.234.47
t=0 0
m=audio 10908 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Feb 5 18:17:36] WARNING[32155]: chan_sip.c:3983 retrans_pkt: Retransmission timeout reached on transmission f6eb303b-224b-4f7a-8bc9-2d56348d28c6 for seqno 21055 (Critical Response) -- See https://wiki.asterisk.org/wiki/displ...etransmissions
Packet timed out after 7680ms with no response
Really destroying SIP dialog 'f6eb303b-224b-4f7a-8bc9-2d56348d28c6' Method: INVITE

<--- SIP read from UDP:190.14.228.30:8606 --->
REGISTER sip:168.61.37.87 SIP/2.0
Via: SIP/2.0/UDP 10.254.246.30:5060;rport;branch=z9hG4bKPj26835103-f13d-4c4c-a7b6-927f82b9edc5
Max-Forwards: 70
From: <sip:jgomez@168.61.37.87>;tag=e05afb17-c5c6-4a88-9c30-a16017684432
To: <sip:jgomez@168.61.37.87>
Call-ID: c40fcb53-fabc-4d10-8440-a3ad56f08612
CSeq: 46411 REGISTER
User-Agent: SFLphone
Contact: <sip:jgomez@10.254.246.30:5060>
Expires: 60
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 190.14.228.30:8606 (NAT)

<--- Transmitting (NAT) to 190.14.228.30:8606 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj26835103-f13d-4c4c-a7b6-927f82b9edc5;received=190.14.228.30;rport=8606
From: <sip:jgomez@168.61.37.87>;tag=e05afb17-c5c6-4a88-9c30-a16017684432
To: <sip:jgomez@168.61.37.87>;tag=as7daa5eed
Call-ID: c40fcb53-fabc-4d10-8440-a3ad56f08612
CSeq: 46411 REGISTER
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f6c34dd"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'c40fcb53-fabc-4d10-8440-a3ad56f08612' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:190.14.228.30:8606 --->
REGISTER sip:168.61.37.87 SIP/2.0
Via: SIP/2.0/UDP 10.254.246.30:5060;rport;branch=z9hG4bKPja4fe68e8-e6cb-4b0a-ac17-7dffbc01a983
Max-Forwards: 70
From: <sip:jgomez@168.61.37.87>;tag=e05afb17-c5c6-4a88-9c30-a16017684432
To: <sip:jgomez@168.61.37.87>
Call-ID: c40fcb53-fabc-4d10-8440-a3ad56f08612
CSeq: 46412 REGISTER
User-Agent: SFLphone
Contact: <sip:jgomez@10.254.246.30:5060>
Expires: 60
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
Authorization: Digest username="jgomez", realm="asterisk", nonce="1f6c34dd", uri="sip:168.61.37.87", response="e95cac695e3f42c1c76ab840a9645b07", algorithm=MD5
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 190.14.228.30:8606 (NAT)
Reliably Transmitting (NAT) to 190.14.228.30:8606:
OPTIONS sip:jgomez@10.254.246.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.74.234.47:5060;branch=z9hG4bK367461a9;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.74.234.47>;tag=as3d6b3703
To: <sip:jgomez@10.254.246.30:5060>
Contact: <sip:asterisk@10.74.234.47:5060>
Call-ID: 39daa4b027d89787562b1ec138c49d02@10.74.234.47:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.25.0
Date: Wed, 05 Feb 2014 18:17:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 190.14.228.30:8606 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPja4fe68e8-e6cb-4b0a-ac17-7dffbc01a983;received=190.14.228.30;rport=8606
From: <sip:jgomez@168.61.37.87>;tag=e05afb17-c5c6-4a88-9c30-a16017684432
To: <sip:jgomez@168.61.37.87>;tag=as7daa5eed
Call-ID: c40fcb53-fabc-4d10-8440-a3ad56f08612
CSeq: 46412 REGISTER
Server: Asterisk PBX 1.8.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:jgomez@10.254.246.30:5060>;expires=60
Date: Wed, 05 Feb 2014 18:17:58 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '020f770241f1c1803db96d9a4fe44b1d@10.74.234.47:506 0' in 7680 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 190.14.228.30:8606:
NOTIFY sip:jgomez@10.254.246.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.74.234.47:5060;branch=z9hG4bK36a4396d;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.74.234.47>;tag=as704a21cf
To: <sip:jgomez@10.254.246.30:5060>
Contact: <sip:asterisk@10.74.234.47:5060>
Call-ID: 020f770241f1c1803db96d9a4fe44b1d@10.74.234.47:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.8.25.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 92

Messages-Waiting: no
Message-Account: sip:asterisk@10.74.234.47
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of SIP dialog 'c40fcb53-fabc-4d10-8440-a3ad56f08612' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:190.14.228.30:8606 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.74.234.47:5060;rport=5060;received=168.61.37.87 ;branch=z9hG4bK367461a9
Call-ID: 39daa4b027d89787562b1ec138c49d02@10.74.234.47:5060
From: "asterisk" <sip:asterisk@10.74.234.47>;tag=as3d6b3703
To: <sip:jgomez@10.254.246.30:5060>;tag=z9hG4bK367461a 9
CSeq: 102 OPTIONS
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
Accept: message/sipfrag;version=2.0, application/sdp, text/plain, application/sdp
Supported: replaces, 100rel
Allow-Events: refer
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '39daa4b027d89787562b1ec138c49d02@10.74.234.47:506 0' Method: OPTIONS

<--- SIP read from UDP:190.14.228.30:8606 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.74.234.47:5060;rport=5060;received=168.61.37.87 ;branch=z9hG4bK36a4396d
Call-ID: 020f770241f1c1803db96d9a4fe44b1d@10.74.234.47:5060
From: "asterisk" <sip:asterisk@10.74.234.47>;tag=as704a21cf
To: <sip:jgomez@10.254.246.30:5060>;tag=z9hG4bK36a4396 d
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '020f770241f1c1803db96d9a4fe44b1d@10.74.234.47:506 0' Method: NOTIFY

[Christian Cabrera]

La linea

Peer audio RTP is at port 10.254.246.30:13202

te dice que es un problema de RTP que se está enviando a la dirección/puerto incorrecto, ya que tu usuario está en una IP pública diferente y el RTP está tratando de enviarse a una local.

¿Cuál es la red local donde Asterisk está?
¿Hacia que IP estás registrando el teléfono del usuario (normalmente es lo que pones en el campo de domain del softphone)?