[Asterisk-Users] More beginner questions

Sat Dec 20 11:00:17 MST 2003

Using DIAX softphone which seems to be working OK can get to VM/echotest etc
in the demo context

Am trying to setup FWD but get the following problems

Can hear it ringing when dialing FWD no 612 for time.  Connects but no sound
from remote end.

Does anyone have any suggestions.

Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to
the internet port 5060 being forwarded to the asterisk box.

This seems to be quite useful software but it's frustratingly difficult to
get running.

Jon

SIP debug shows following

mrpenguin*CLI>
    -- Registered '2203' (AUTHENTICATED) at 192.168.0.2:5036
    -- Accepting AUTHENTICATED call from 192.168.0.2, requested format = 2,
actu
al format = 2
    -- Executing SetCallerID("IAX[2203 at 2203]/9", "91184") in new stack
    -- Executing SetCIDName("IAX[2203 at 2203]/9", "calisto") in new stack
    -- Executing Dial("IAX[2203 at 2203]/9", "SIP/612 at fwd.pulver.com") in new
stack
We're at 82.38.193.149 port 16612
Answering with preferred capability 4
Answering with preferred capability 2
Answering with non-codec capability 1
11 headers, 10 lines
Reliably Transmitting:
INVITE sip:612 at fwd.pulver.com SIP/2.0
Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372
From: "calisto" <sip:91184 at 82.38.193.149>;tag=as1f0e4544
To: <sip:612 at fwd.pulver.com>
Contact: <sip:91184 at 82.38.193.149>
Call-ID: 4336592d02a1a1ba7173c842778e8b8e at 82.38.193.149
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 15141 15141 IN IP4 82.38.193.149
s=session
c=IN IP4 82.38.193.149
t=0 0
m=audio 16612 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
 (NAT) to 192.246.69.223:5060
    -- Called 612 at fwd.pulver.com
Sip read: CLI>
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372
From: "calisto" <sip:91184 at 82.38.193.149>;tag=as1f0e4544
To: <sip:612 at fwd.pulver.com>
Call-ID: 4336592d02a1a1ba7173c842778e8b8e at 82.38.193.149
CSeq: 102 INVITE
Server: Free World Dialup (0.8.11rc3 (i386/linux))
Content-Length: 0

8 headers, 0 lines
Sip read: CLI>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372
From: "calisto" <sip:91184 at 82.38.193.149>;tag=as1f0e4544
To: <sip:612 at fwd.pulver.com>;tag=as63b4567c
Call-ID: 4336592d02a1a1ba7173c842778e8b8e at 82.38.193.149
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Contact: <sip:612 at 65.39.205.112:5028>
Content-Length: 0

9 headers, 0 lines
    -- SIP/fwd.pulver.com-43fd is ringing
Sip read: CLI>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372
From: "calisto" <sip:91184 at 82.38.193.149>;tag=as1f0e4544
To: <sip:612 at fwd.pulver.com>;tag=as2046b5cb
Call-ID: 4336592d02a1a1ba7173c842778e8b8e at 82.38.193.149
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:30342 at 65.121.72.14>
Content-Length: 0

10 headers, 0 lines
    -- SIP/fwd.pulver.com-43fd is ringing
Sip read: CLI>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372
Record-Route: <sip:612 at 192.246.69.223;ftag=as1f0e4544;lr=on>
From: "calisto" <sip:91184 at 82.38.193.149>;tag=as1f0e4544
To: <sip:612 at fwd.pulver.com>;tag=as2046b5cb
Call-ID: 4336592d02a1a1ba7173c842778e8b8e at 82.38.193.149
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:30342 at 65.121.72.14>
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 11472 11472 IN IP4 65.121.72.14
s=session
c=IN IP4 65.121.72.14
t=0 0
m=audio 12268 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

12 headers, 10 lines
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format GSM
Found description format PCMU
Found description format telephone-event
Capabilities: us - 6, them - 6/0, combined - 6
Non-codec capabilities: us - 1, them - 1, combined - 1
list_route: hop: <sip:612 at 192.246.69.223;ftag=as1f0e4544;lr=on>
list_route: hop: <sip:30342 at 65.121.72.14>
set_destination: Parsing <sip:612 at 192.246.69.223;ftag=as1f0e4544;lr=on> for
addr
ess/port to send to
set_destination: set destination to 192.246.69.223, port 5060
Transmitting:
ACK sip:30342 at 65.121.72.14 SIP/2.0
Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372
Route: <sip:30342 at 65.121.72.14>
From: "calisto" <sip:91184 at 82.38.193.149>;tag=as1f0e4544
To: <sip:612 at fwd.pulver.com>;tag=as2046b5cb
Contact: <sip:91184 at 82.38.193.149>
Call-ID: 4336592d02a1a1ba7173c842778e8b8e at 82.38.193.149
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 192.246.69.223:5060
    -- SIP/fwd.pulver.com-43fd answered IAX[2203 at 2203]/9
set_destination: Parsing <sip:612 at 192.246.69.223;ftag=as1f0e4544;lr=on> for
addr
ess/port to send to
set_destination: set destination to 192.246.69.223, port 5060
Reliably Transmitting:
BYE sip:30342 at 65.121.72.14 SIP/2.0
Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372
Route: <sip:30342 at 65.121.72.14>
From: "calisto" <sip:91184 at 82.38.193.149>;tag=as1f0e4544
To: <sip:612 at fwd.pulver.com>;tag=as2046b5cb
Contact: <sip:91184 at 82.38.193.149>
Call-ID: 4336592d02a1a1ba7173c842778e8b8e at 82.38.193.149
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 192.246.69.223:5060
  == Spawn extension (default, 7612, 3) exited non-zero on
'IAX[2203 at 2203]/9'
    -- Hungup 'IAX[2203 at 2203]/9'
Sip read: CLI>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372
Record-Route: <sip:30342 at 192.246.69.223;ftag=as1f0e4544;lr=on>
From: "calisto" <sip:91184 at 82.38.193.149>;tag=as1f0e4544
To: <sip:612 at fwd.pulver.com>;tag=as2046b5cb
Call-ID: 4336592d02a1a1ba7173c842778e8b8e at 82.38.193.149
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:30342 at 65.121.72.14>
Content-Length: 0

11 headers, 0 lines
mrpenguin*CLI>

SIP.CONF

mrpenguin:/etc/asterisk# more sip.conf
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
externip = [REMOVED]        ; Address that we're going to put in SIP message
s if we're behind a NAT
context = default               ; Default for incoming calls
disallow=all                    ; Disallow all codecs
allow=ulaw
;allow=alaw
allow=gsm

register => 91184:[REMOVED]@fwd.pulver.com/2203

;[2203]
;type=friend
;username=2203
;host=dynamic
;defaultip=192.168.0.2
;dtmfmode=inband
;canreinvite=no

[fwd.pulver.com]
type=friend
secret=[REMOVED]
username=91184
host=fwd.pulver.com
nat=yes
canreinvite=no
reinvite=no

IAX.conf (for the DIAX softphone)
[general]
port=5036
bandwidth=low
allow=ulaw
allow=gsm
jitterbuffer=no
tos=lowdelay
[guest]
type=user
context=default
callerid="Guest IAX User"

[iaxtel]
type=user
context=default
auth=rsa
inkeys=iaxtel

[iaxtel2]
type=user
context=default
deny=0.0.0.0/0.0.0.0
permit=216.207.245.47/255.255.255.255

[demo]
type=peer
username=asterisk
secret=supersecret
host=216.207.245.47

[2203]
type=friend
host=dynamic
secret=mypassword
context=default
qualify=300
nat=yes