Tengo un servidor Asterisk en Canadá y una extensión remota en México.
Al principio podía hacer llamadas ocacionalmente, pero en los últimos meses, ya no ha funcionado. Se puede marcar de cualquiera de los dos lados y hacer timbrar el teléfono del otro lado, pero no hay audio en ninguno de los teléfonos. Unas dos veces, ha habido audio en la extensión remota cuando llamo del Asterisk, pero no había audio del lado de Asterisk.
Yo sospecho el Cablemas está bloqueando los puertos 10000-20000. Antes tenía un adaptador en México conectado a Voxalot, y lograba hacer llamadas, aunque con muchos problemas, pero dejó de funcionar como a mediados del 2010, así que decidí poner un teléfono conectado directo a mi Asterisk.
Tengo otras extensiones locales y tambien truncales SIP que funcionan bien, por eso creo que le problemas esta en la extension remota.
Aquí les dejo mis datos, para ver si alguien ve algo que yo esté haciendo mal.

CentOS 5.2
Asterisk 1.4 (compilado)
FreePBX 2.5

El Asterisk PBX detrás de un router con firmware Tomato con puertos 5060 UDP y 10000-20000 UDP forwarded al Asterisk. IP 2xx.45.198.xx0
Extension remota teléfno Cisco 7912g SIP detrás de un Ambit U10C022 (adaptador VoIP, modem de cable y router) con puertos 5060 UDP forwarded a 5060 UDP en el teléfono y 10000-20000 UDP forwarded a 16384 UDP en le teléfono. IP 3xx.170.67.xx9
Mi extensión local es 200 (ATA) y la remota (CISCO 7912g) es extensión 400. La IP local de la ext. remota es 192.168.0.4.
He tradado de registrar la extensión remota con midyndns.com (mi dominio) y también con la IP externa del Asterisk, pero con los mismo resultados.

Mi extension tiene:

nat=yes
qualify=yes
host=dynamic (también intenté con la IP estática remota)
disallow=all
allow=ulaw

Mi sip_nat.conf tiene:

nat=yes
externhost=midyndns.com
externrefresh=30
localnet=192.168.1.0/255.255.255.0

Mi rtp.conf tiene:

rtpstart=10000
rtpend=20000

En el teléfono tengo:

UID=400	
PWD=miclave	
Proxy=2xx.45.198.xx0
AltProxyTimeOut=0
UseLoginID=0
LoginID=0
SIPRegInterval=60	
MaxRedirect=5
SIPRegOn=1
NATIP=3xx.170.67.xx9
SIPPort=5060	
MediaPort=16384	
OutBoundProxy=0
MsgRetryLimits=0x00000000
NatServer=0	
NatTimer=0x00000000	
DialPlan=911|1>#t8.r9t2-|0>#t811.rat4-|^1t4>#.-	
IPDialPlan=1

Lo que sigue es un tcpdump de Asterisk, primero haciendo una llamada de ext. 200 a ext. 400 y luego de 400 a 200.
Los paquetes RTP salen de Asterisk cuando hablo de 200 a 400 y tambien cuando recibo la llamada en 200, los paquetes salen de Asterisk,pero no regresan de la extensión remota en ninguno de los casos.

SIP/2.0 200 OK
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK6da4b122;rport
From: "John Doe" <sip:200@2xx.45.198.xx0>;tag=as473f3fd1
To: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>;tag=2784700350
Call-ID: [email]21ee464e40297e59208e576b2ba32a2b@2xx.45.198.xx[/email]0
CSeq: 102 INVITE
Contact: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>
Server: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces
Content-Length: 203
Content-Type: application/sdp

v=0
o=400 35745 35745 IN IP4 192.168.0.4
s=Cisco 7912 SIP Call
c=IN IP4 3xx.170.67.xx9
t=0 0
m=audio 16384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
ACK sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK147fe730;rport
From: "John Doe" <sip:200@2xx.45.198.xx0>;tag=as473f3fd1
To: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>;tag=2784700350
Contact: <sip:200@2xx.45.198.xx0>
Call-ID: [email]21ee464e40297e59208e576b2ba32a2b@2xx.45.198.xx[/email]0
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

REGISTER sip:2xx.45.198.xx0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK567d8b8e9299827
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=1530313901
To: <sip:400@2xx.45.198.xx0;user=phone>
Call-ID: [email]1213881949@3xx.170.67.xx[/email]9
CSeq: 15 REGISTER
Max-Forwards: 70
Contact: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>;expires=60;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0012001441E0>"
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Authorization: Digest username="400",realm="asterisk",nonce="4c5325e1",uri="sip:2xx.45.198.xx0",response="0252dfb6d50378a0ae4f2a52802db870"
Content-Length: 0

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK567d8b8e9299827;received=3xx.170.67.xx9
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=1530313901
To: <sip:400@2xx.45.198.xx0;user=phone>
Call-ID: [email]1213881949@3xx.170.67.xx[/email]9
CSeq: 15 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK567d8b8e9299827;received=3xx.170.67.xx9
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=1530313901
To: <sip:400@2xx.45.198.xx0;user=phone>;tag=as0b1f27b4
Call-ID: [email]1213881949@3xx.170.67.xx[/email]9
CSeq: 15 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40ff7952"
Content-Length: 0

REGISTER sip:2xx.45.198.xx0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK6ed2d5fac1b82c57
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=1530313901
To: <sip:400@2xx.45.198.xx0;user=phone>
Call-ID: [email]1213881949@3xx.170.67.xx[/email]9
CSeq: 16 REGISTER
Max-Forwards: 70
Contact: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>;expires=60;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0012001441E0>"
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Authorization: Digest username="400",realm="asterisk",nonce="40ff7952",uri="sip:2xx.45.198.xx0",response="0d8a01b0c144b40bc7cfe0aa019c4319"
Content-Length: 0

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK6ed2d5fac1b82c57;received=3xx.170.67.xx9
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=1530313901
To: <sip:400@2xx.45.198.xx0;user=phone>
Call-ID: [email]1213881949@3xx.170.67.xx[/email]9
CSeq: 16 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

OPTIONS sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK6fb5e96d;rport
From: "Unknown" <sip:Unknown@2xx.45.198.xx0>;tag=as037107dc
To: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>
Contact: <sip:Unknown@2xx.45.198.xx0>
Call-ID: [email]706d52c93ce8a22a5f79477f2afa4ce4@2xx.45.198.xx[/email]0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 16 Mar 2011 18:32:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK6ed2d5fac1b82c57;received=3xx.170.67.xx9
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=1530313901
To: <sip:400@2xx.45.198.xx0;user=phone>;tag=as0b1f27b4
Call-ID: [email]1213881949@3xx.170.67.xx[/email]9
CSeq: 16 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Expires: 60
Contact: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>;expires=60
Date: Wed, 16 Mar 2011 18:32:47 GMT
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK6fb5e96d;rport
From: "Unknown" <sip:Unknown@2xx.45.198.xx0>;tag=as037107dc
To: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>;tag=2784700350
Call-ID: [email]706d52c93ce8a22a5f79477f2afa4ce4@2xx.45.198.xx[/email]0
CSeq: 102 OPTIONS
Server: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces
Content-Length: 278
Content-Type: application/sdp

v=0
o=400 36716 36716 IN IP4 192.168.0.4
s=Cisco 7912 SIP Call
c=IN IP4 3xx.170.67.xx9
t=0 0
m=audio 16384 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
BYE sip:200@2xx.45.198.xx0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK547cb3462f02355b
From: <sip:400@3xx.170.67.xx9;user=phone;transport=udp>;tag=2784700350
To: "John Doe" <sip:200@2xx.45.198.xx0>;tag=as473f3fd1
Call-ID: [email]21ee464e40297e59208e576b2ba32a2b@2xx.45.198.xx[/email]0
CSeq: 1 BYE
Max-Forwards: 70
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK547cb3462f02355b;received=3xx.170.67.xx9
From: <sip:400@3xx.170.67.xx9;user=phone;transport=udp>;tag=2784700350
To: "John Doe" <sip:200@2xx.45.198.xx0>;tag=as473f3fd1
Call-ID: [email]21ee464e40297e59208e576b2ba32a2b@2xx.45.198.xx[/email]0
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

NOTIFY sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK4536d65f;rport
From: "Unknown" <sip:Unknown@2xx.45.198.xx0>;tag=as52a7c7cf
To: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>
Contact: <sip:Unknown@2xx.45.198.xx0>
Call-ID: [email]17b617ed4359421953b05b0c712a0434@2xx.45.198.xx[/email]0
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 89

Messages-Waiting: no
Message-Account: sip:*97@2xx.45.198.xx0
Voice-Message: 0/0 (0/0)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK4536d65f;rport
From: "Unknown" <sip:Unknown@2xx.45.198.xx0>;tag=as52a7c7cf
To: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>;tag=2784700350
Call-ID: [email]17b617ed4359421953b05b0c712a0434@2xx.45.198.xx[/email]0
CSeq: 102 NOTIFY
Server: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces
Content-Length: 0

INVITE sip:200@2xx.45.198.xx0;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK1000cbe7bbc24227
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=2784700350
To: <sip:200@2xx.45.198.xx0;user=phone>
Call-ID: [email]322380413@3xx.170.67.xx[/email]9
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Expires: 300
Content-Length: 278
Content-Type: application/sdp

v=0
o=400 38794 38794 IN IP4 192.168.0.4
s=Cisco 7912 SIP Call
c=IN IP4 3xx.170.67.xx9
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK1000cbe7bbc24227;received=3xx.170.67.xx9
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=2784700350
To: <sip:200@2xx.45.198.xx0;user=phone>;tag=as4117842f
Call-ID: [email]322380413@3xx.170.67.xx[/email]9
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5d905ae6"
Content-Length: 0

ACK sip:200@2xx.45.198.xx0;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK1000cbe7bbc24227
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=2784700350
To: <sip:200@2xx.45.198.xx0;user=phone>;tag=as4117842f
Call-ID: [email]322380413@3xx.170.67.xx[/email]9
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: Cisco-CP7912/8.0.1-060412A
Content-Length: 0

INVITE sip:200@2xx.45.198.xx0;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK3282ce62635f24
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=2784700350
To: <sip:200@2xx.45.198.xx0;user=phone>
Call-ID: [email]322380413@3xx.170.67.xx[/email]9
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Proxy-Authorization: Digest username="400",realm="asterisk",nonce="5d905ae6",uri="sip:200@2xx.45.198.xx0",response="d0ed51a539f45abd67f2942966180389"
Expires: 300
Content-Length: 278
Content-Type: application/sdp

v=0
o=400 38809 38809 IN IP4 192.168.0.4
s=Cisco 7912 SIP Call
c=IN IP4 3xx.170.67.xx9
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK3282ce62635f24;received=3xx.170.67.xx9
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=2784700350
To: <sip:200@2xx.45.198.xx0;user=phone>
Call-ID: [email]322380413@3xx.170.67.xx[/email]9
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:200@2xx.45.198.xx0>
Content-Length: 0

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK3282ce62635f24;received=3xx.170.67.xx9
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=2784700350
To: <sip:200@2xx.45.198.xx0;user=phone>;tag=as2f169cf8
Call-ID: [email]322380413@3xx.170.67.xx[/email]9
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:200@2xx.45.198.xx0>
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK3282ce62635f24;received=3xx.170.67.xx9
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=2784700350
To: <sip:200@2xx.45.198.xx0;user=phone>;tag=as2f169cf8
Call-ID: [email]322380413@3xx.170.67.xx[/email]9
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:200@2xx.45.198.xx0>
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 8551 8551 IN IP4 2xx.45.198.xx0
s=session
c=IN IP4 2xx.45.198.xx0
t=0 0
m=audio 10684 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
ACK sip:200@2xx.45.198.xx0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK611bd62e7cc429ef
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=2784700350
To: <sip:200@2xx.45.198.xx0;user=phone>;tag=as2f169cf8
Call-ID: [email]322380413@3xx.170.67.xx[/email]9
CSeq: 2 ACK
Max-Forwards: 70
User-Agent: Cisco-CP7912/8.0.1-060412A
Proxy-Authorization: Digest username="400",realm="asterisk",nonce="5d905ae6",uri="sip:200@2xx.45.198.xx0",response="d0ed51a539f45abd67f2942966180389"
Content-Length: 0

BYE sip:200@2xx.45.198.xx0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK8521a031f7750eee
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=2784700350
To: <sip:200@2xx.45.198.xx0;user=phone>;tag=as2f169cf8
Call-ID: [email]322380413@3xx.170.67.xx[/email]9
CSeq: 3 BYE
Max-Forwards: 70
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Proxy-Authorization: Digest username="400",realm="asterisk",nonce="5d905ae6",uri="sip:200@2xx.45.198.xx0",response="6071ac25b458f4a8e532322a237b9312"
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK8521a031f7750eee;received=3xx.170.67.xx9
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=2784700350
To: <sip:200@2xx.45.198.xx0;user=phone>;tag=as2f169cf8
Call-ID: [email]322380413@3xx.170.67.xx[/email]9
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

REGISTER sip:2xx.45.198.xx0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK5d1a60f76c2b6f97
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=1530313901
To: <sip:400@2xx.45.198.xx0;user=phone>
Call-ID: [email]1213881949@3xx.170.67.xx[/email]9
CSeq: 17 REGISTER
Max-Forwards: 70
Contact: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>;expires=60;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0012001441E0>"
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Authorization: Digest username="400",realm="asterisk",nonce="40ff7952",uri="sip:2xx.45.198.xx0",response="0d8a01b0c144b40bc7cfe0aa019c4319"
Content-Length: 0

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK5d1a60f76c2b6f97;received=3xx.170.67.xx9
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=1530313901
To: <sip:400@2xx.45.198.xx0;user=phone>
Call-ID: [email]1213881949@3xx.170.67.xx[/email]9
CSeq: 17 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK5d1a60f76c2b6f97;received=3xx.170.67.xx9
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=1530313901
To: <sip:400@2xx.45.198.xx0;user=phone>;tag=as1b6e1592
Call-ID: [email]1213881949@3xx.170.67.xx[/email]9
CSeq: 17 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="062b4dc0"
Content-Length: 0

REGISTER sip:2xx.45.198.xx0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK99ada71dbc721779
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=1530313901
To: <sip:400@2xx.45.198.xx0;user=phone>
Call-ID: [email]1213881949@3xx.170.67.xx[/email]9
CSeq: 18 REGISTER
Max-Forwards: 70
Contact: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>;expires=60;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0012001441E0>"
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Authorization: Digest username="400",realm="asterisk",nonce="062b4dc0",uri="sip:2xx.45.198.xx0",response="709a7eb6fdff875d34f9b407240f1afd"
Content-Length: 0

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK99ada71dbc721779;received=3xx.170.67.xx9
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=1530313901
To: <sip:400@2xx.45.198.xx0;user=phone>
Call-ID: [email]1213881949@3xx.170.67.xx[/email]9
CSeq: 18 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

OPTIONS sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK10e603e9;rport
From: "Unknown" <sip:Unknown@2xx.45.198.xx0>;tag=as666bb023
To: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>
Contact: <sip:Unknown@2xx.45.198.xx0>
Call-ID: [email]53579afc01673dc1369ba8fe2a7ed4e0@2xx.45.198.xx[/email]0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 16 Mar 2011 18:33:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK99ada71dbc721779;received=3xx.170.67.xx9
From: <sip:400@2xx.45.198.xx0;user=phone>;tag=1530313901
To: <sip:400@2xx.45.198.xx0;user=phone>;tag=as1b6e1592
Call-ID: [email]1213881949@3xx.170.67.xx[/email]9
CSeq: 18 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Expires: 60
Contact: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>;expires=60
Date: Wed, 16 Mar 2011 18:33:39 GMT
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 2xx.45.198.xx0:5060;branch=z9hG4bK10e603e9;rport
From: "Unknown" <sip:Unknown@2xx.45.198.xx0>;tag=as666bb023
To: <sip:400@3xx.170.67.xx9:5060;user=phone;transport=udp>;tag=2784700350
Call-ID: [email]53579afc01673dc1369ba8fe2a7ed4e0@2xx.45.198.xx[/email]0
CSeq: 102 OPTIONS
Server: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces
Content-Length: 278
Content-Type: application/sdp

v=0
o=400 41949 41949 IN IP4 192.168.0.4
s=Cisco 7912 SIP Call
c=IN IP4 3xx.170.67.xx9
t=0 0
m=audio 16384 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

¿Ven algo que estoy haciendo mal?