[Asterisk-Users] Sipura SPA2000 behind NAT
Guillermo Salas M
gsalas at manta.telconet.net
Sat Jul 2 13:12:58 MST 2005
Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network
adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP:
___________ HOME _______________ ____OFFICE ____
SPA2000 <---> Linux Box <--> Asterisk Box
192.168.0.253 192.168.0.1 eth1 200.93.xxx.a
200.93.xxx.b eth0
My problem is when I try to call to any trunk or extention I can the
audio when the destination is ringing, but I can hear the voice of the
person when it reponds. The person in the other side can hear me, but I
can not hear anything from him. I can not hear the voice prompts for the
voicemail (*98) or the operator voice, but can leave voice messages to
other SIP devices and they can hear my messages.
This is my sip.conf
[105]
username=105
type=friend
secret=105
qualify=no
port=5060
nat=yes
mailbox=105 at default
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="Guilllermo Salas HOME" <105>
My ext on line 1 of the Sipura is 105, and is registred with the * box:
-- Registered SIP '105' at 200.93.220.27 port 5060 expires 3600
asterisk*CLI> sip show peer 105
asterisk*CLI>
* Name : 105
Secret : <Set>
MD5Secret : <Not set>
Context : from-internal
Language : es
FromUser :
FromDomain :
Callgroup : (0)
Pickupgroup : (0)
Mailbox : 105 at default
LastMsgsSent : 2
Dynamic : Yes
Expire : 4
Expiry : 900
Insecure : No
Nat : Always
ACL : No
CanReinvite : No
PromiscRedir : No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 200.93.xxx.xb Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Username : 105
Codecs : 0xc011f (g723|gsm|ulaw|alaw|g726|g729|h261|h263)
Codec Order : (g729|g723|gsm|g726|ulaw|alaw|h261|h263)
Status : UNKNOWN
Useragent :
Full Contact : sip:105 at 192.168.0.253:5060
And this is the output of sip debug peer 105 when I call to *98 (for
voice messages):
asterisk*CLI> sip debug peer 105
SIP Debugging Enabled for IP: 200.93.xxx.xb:5060
Sip read:
NOTIFY sip:sip.mydomain.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-67ea7370
From: Guillermo Salas M <sip:105 at sip.mydomain.net>;tag=4f2df183b116b70c
To: <sip:sip.mydomain.net>
Call-ID: a584ba93-53c0013c at 192.168.0.253
CSeq: 4 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.2
Content-Length: 0
10 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-67ea7370
From: Guillermo Salas M <sip:105 at sip.mydomain.net>;tag=4f2df183b116b70c
To: <sip:sip.mydomain.net>;tag=as038653dd
Call-ID: a584ba93-53c0013c at 192.168.0.253
CSeq: 4 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
to 200.93.xxx.xb:5060
Destroying call 'a584ba93-53c0013c at 192.168.0.253'
asterisk*CLI>
Sip read:
NOTIFY sip:sip.mydomain.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-d386a279
From: Guillermo Salas M <sip:105 at sip.mydomain.net>;tag=4f2df183b116b70c
To: <sip:sip.mydomain.net>
Call-ID: a584ba93-53c0013c at 192.168.0.253
CSeq: 6 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.2
Content-Length: 0
10 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-d386a279
From: Guillermo Salas M <sip:105 at sip.mydomain.net>;tag=4f2df183b116b70c
To: <sip:sip.mydomain.net>;tag=as5099fa8f
Call-ID: a584ba93-53c0013c at 192.168.0.253
CSeq: 6 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
to 200.93.xxx.xb:5060
Destroying call 'a584ba93-53c0013c at 192.168.0.253'
asterisk*CLI>
I dial *98 to get into the voice message system:
asterisk*CLI>
Sip read:
ACK sip:*98 at sip.mydomain.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-600583f3
From: Guillermo Salas M <sip:105 at sip.mydomain.net>;tag=4f2df183b116b70c
To: <sip:*98 at sip.mydomain.net>;tag=as65eec750
Call-ID: 636a9064-eba36dcb at 192.168.0.253
CSeq: 101 ACK
Max-Forwards: 70
Contact: Guillermo Salas M <sip:105 at 192.168.0.253>
User-Agent: Sipura/SPA2000-2.0.2
Content-Length: 0
10 headers, 0 lines
asterisk*CLI>
Sip read:
INVITE sip:*98 at sip.mydomain.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-ec22067b
From: Guillermo Salas M <sip:105 at sip.mydomain.net>;tag=4f2df183b116b70c
To: <sip:*98 at sip.mydomain.net>
Call-ID: 636a9064-eba36dcb at 192.168.0.253
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="105",realm="asterisk",nonce="47a68adb",uri="sip:*98 at sip.mydomain.net",algorithm=MD5,response="8e60f592df094f9b852a59544b9da384"
Contact: Guillermo Salas M <sip:105 at 192.168.0.253>
Expires: 240
User-Agent: Sipura/SPA2000-2.0.2
Content-Length: 422
Content-Type: application/sdp
v=0
o=- 12384 12384 IN IP4 192.168.0.253
s=-
c=IN IP4 192.168.0.253
t=0 0
m=audio 16468 RTP/AVP 4 0 2 8 18 96 97 98 100 101
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
13 headers, 19 lines
Using latest request as basis request
Sending to 192.168.0.253 : 5060 (NAT)
Found user '105'
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.253:16468
Found description format G723
Found description format PCMU
Found description format G726-32
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0xc011f (g723|gsm|ulaw|alaw|g726|g729|h261|h263),
peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing),
combined - 0x11d (g723|ulaw|alaw|g726|g729)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for *98 in from-internal
list_route: hop: <sip:105 at 192.168.0.253>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.253;branch=z9hG4bK-ec22067b;received=200.93.xxx.xb;rport=5060
From: Guillermo Salas M <sip:105 at sip.mydomain.net>;tag=4f2df183b116b70c
To: <sip:*98 at sip.mydomain.net>;tag=as58095e00
Call-ID: 636a9064-eba36dcb at 192.168.0.253
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*98 at 200.93.xxx.xa>
Content-Length: 0
to 200.93.xxx.xb:5060
-- Executing Answer("SIP/105-6408", "") in new stack
We're at 200.93.xxx.xa port 12436
Video is at 200.93.xxx.xa port 16274
Answering with preferred capability 0x100 (g729)
Answering with preferred capability 0x1 (g723)
Answering with preferred capability 0x2 (gsm)
Answering with preferred capability 0x10 (g726)
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x40000 (h261)
Answering with preferred capability 0x80000 (h263)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.253;branch=z9hG4bK-ec22067b;received=200.93.xxx.xb;rport=5060
From: Guillermo Salas M <sip:105 at sip.mydomain.net>;tag=4f2df183b116b70c
To: <sip:*98 at sip.mydomain.net>;tag=as58095e00
Call-ID: 636a9064-eba36dcb at 192.168.0.253
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*98 at 200.93.xxx.xa>
Content-Type: application/sdp
Content-Length: 340
v=0
=root 7393 7393 IN IP4 200.93.xxx.xa
s=session
c=IN IP4 200.93.xxx.xa
t=0 0
m=audio 12436 RTP/AVP 18 4 3 2 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 200.93.xxx.xb:5060
-- Executing Wait("SIP/105-6408", "1") in new stack
asterisk*CLI>
Sip read:
ACK sip:*98 at 200.93.xxx.xa SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-ec22067b
From: Guillermo Salas M <sip:105 at sip.mydomain.net>;tag=4f2df183b116b70c
To: <sip:*98 at sip.mydomain.net>;tag=as58095e00
Call-ID: 636a9064-eba36dcb at 192.168.0.253
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest
username="105",realm="asterisk",nonce="47a68adb",uri="sip:*98 at sip.mydomain.net",algorithm=MD5,response="74dd50faa2bb97fdb1a0fe6ce93489de"
Contact: Guillermo Salas M <sip:105 at 192.168.0.253>
User-Agent: Sipura/SPA2000-2.0.2
Content-Length: 0
11 headers, 0 lines
-- Executing VoiceMailMain("SIP/105-6408", "default") in new stack
-- Playing 'vm-login' (language 'es')
asterisk*CLI>
Sip read:
NOTIFY sip:sip.mydomain.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-8ecd1b3e
From: Guillermo Salas M <sip:105 at sip.mydomain.net>;tag=4f2df183b116b70c
To: <sip:sip.mydomain.net>
Call-ID: a584ba93-53c0013c at 192.168.0.253
CSeq: 9 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.2
Content-Length: 0
10 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-8ecd1b3e
From: Guillermo Salas M <sip:105 at sip.mydomain.net>;tag=4f2df183b116b70c
To: <sip:sip.mydomain.net>;tag=as45caf3ff
Call-ID: a584ba93-53c0013c at 192.168.0.253
CSeq: 9 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
to 200.93.xxx.xb:5060
Destroying call 'a584ba93-53c0013c at 192.168.0.253'
-- No username but # key pressed. Using CID '105'
-- Playing 'vm-password' (language 'es')
-- Incorrect password '' for user '105' (context = <any>)
-- Playing 'vm-incorrect-mailbox' (language 'es')
asterisk*CLI>
Any hint will be very appreciated,
Regards,
Guill3rm0
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