[HELP] Asterisk 1.0.9 on NetBSD Can't call out!


#1

Hello all,

Just a warning! I am new to the asterisk world :smile:

I have been working on getting Asterisk to run on the Cobalt Cube 2 microserver it’s a MIPS based cpu box with no vga 2x 10/100 1x pci.
It is currently running NetBSD 2.0, No I will not use Linux it will not run as smooth on this system.
Using the pkgsrc under Netbsd I have needed to make some changes in the make file optimization I’ve posted them to the teck-pkg NetBSD list. This is for 1.0.9 asterisk. I have also used the makefile flags for optimization for low memory/embedded systems since that fits the cube best.
With the changes I have made Asterisk will compile but I am still working on the Zaptel stuff.

Since I am only a VOIP user I wanted to see if asterisk is actually working on my cube!

Although I have seemed to run into a few issues mainly I can’t call out! I read and read some more but I just can not work this one out. If anyone can tell me what to read I am quite willing to read some more

What happens is when I try dial a number with the prefix 72 (this should get me outside line on my asterisk) I get the error message
"
Looking for 7246302065 in home
Reliably Transmitting (no NAT):
SIP/2.0 484 Address Incomplete
"

I am not sure how to fix this?

I would like someone to assist me to try and get my conf files correct so I can continue testing asterisk on the cube microserver.

I will post my full sip debug log in the next post.

Thank you all for your help with this matter.


#2

I am using SJPhone (softphone) for testing I will swap over to my real phone once it works.

Here is my Sip debug.


Sip read:
INVITE sip:7246302065@192.168.0.249 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253:5060;rport;branch=z9hG4bKc0a800020000002b43a7a40100002fc90000002f
Content-Length: 337
Contact: sip:1000@192.168.0.253:5060
Call-ID: 235CB50F-9BC8-489D-B826-61B8808D3C25@192.168.0.2
Content-Type: application/sdp
CSeq: 1 INVITE
From: "1000"sip:1000@192.168.0.249;tag=92257810573
Max-Forwards: 70
To: sip:7246302065@192.168.0.249
User-Agent: SJphone/1.60.289a (SJ Labs)

v=0
o=- 3344048769 3344048769 IN IP4 192.168.0.2
s=SJphone
c=IN IP4 192.168.0.253
t=0 0
a=direction:active
m=audio 49160 RTP/AVP 8 0 3 97 98 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

11 headers, 15 lines
Using latest request as basis request
Sending to 192.168.0.253 : 5060 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.253:5060;branch=z9hG4bKc0a800020000002b43a7a40100002fc90000002f
From: "1000"sip:1000@192.168.0.249;tag=92257810573
To: sip:7246302065@192.168.0.249;tag=as4b934b4c
Call-ID: 235CB50F-9BC8-489D-B826-61B8808D3C25@192.168.0.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:7246302065@192.168.0.249
Proxy-Authenticate: Digest realm=“asterisk”, nonce="17706795"
Content-Length: 0

to 192.168.0.253:5060
Scheduling destruction of call ‘235CB50F-9BC8-489D-B826-61B8808D3C25@192.168.0.2’ in 15000 ms
Found user ‘1000’

Sip read:
ACK sip:7246302065@192.168.0.249 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253:5060;rport;branch=z9hG4bKc0a800020000002b43a7a40100002fc90000002f
Content-Length: 0
Call-ID: 235CB50F-9BC8-489D-B826-61B8808D3C25@192.168.0.2
CSeq: 1 ACK
From: "1000"sip:1000@192.168.0.249;tag=92257810573
Max-Forwards: 70
To: sip:7246302065@192.168.0.249;tag=as4b934b4c
User-Agent: SJphone/1.60.289a (SJ Labs)

9 headers, 0 lines

Sip read:
INVITE sip:7246302065@192.168.0.249 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253:5060;rport;branch=z9hG4bKc0a800020000002b43a7a4010000102b00000030
Content-Length: 337
Contact: sip:1000@192.168.0.253:5060
Call-ID: 235CB50F-9BC8-489D-B826-61B8808D3C25@192.168.0.2
Content-Type: application/sdp
CSeq: 2 INVITE
From: “1000"sip:1000@192.168.0.249;tag=92257810573
Max-Forwards: 70
To: sip:7246302065@192.168.0.249
User-Agent: SJphone/1.60.289a (SJ Labs)
Proxy-Authorization: Digest username=“1000”,realm=“asterisk”,nonce=“17706795”,uri="sip:7246302065@192.168.0.249”,response=“ec915f30d163dece370fd2a96b99e5f0”

v=0
o=- 3344048769 3344048769 IN IP4 192.168.0.2
s=SJphone
c=IN IP4 192.168.0.253
t=0 0
a=direction:active
m=audio 49160 RTP/AVP 8 0 3 97 98 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

12 headers, 15 lines
Using latest request as basis request
Sending to 192.168.0.253 : 5060 (non-NAT)
Found user '1000’
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.253:49160
Found description format PCMA
Found description format PCMU
Found description format GSM
Found description format iLBC
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x8050e (gsm|ulaw|alaw|g729|ilbc|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x40e (gsm|ulaw|alaw|ilbc)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Looking for 7246302065 in home
Reliably Transmitting (no NAT):
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 192.168.0.253:5060;branch=z9hG4bKc0a800020000002b43a7a4010000102b00000030
From: "1000"sip:1000@192.168.0.249;tag=92257810573
To: sip:7246302065@192.168.0.249;tag=as4b934b4c
Call-ID: 235CB50F-9BC8-489D-B826-61B8808D3C25@192.168.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:7246302065@192.168.0.249
Content-Length: 0

to 192.168.0.253:5060

Sip read:
ACK sip:7246302065@192.168.0.249 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253:5060;rport;branch=z9hG4bKc0a800020000002b43a7a4010000102b00000030
Content-Length: 0
Call-ID: 235CB50F-9BC8-489D-B826-61B8808D3C25@192.168.0.2
CSeq: 2 ACK
From: "1000"sip:1000@192.168.0.249;tag=92257810573
Max-Forwards: 70
To: sip:7246302065@192.168.0.249;tag=as4b934b4c
User-Agent: SJphone/1.60.289a (SJ Labs)

9 headers, 0 lines
Destroying call '235CB50F-9BC8-489D-B826-61B8808D3C25@192.168.0.2’
blackbox*CLI> exit

Sip read:
OPTIONS sip:192.168.0.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253:5060;rport;branch=z9hG4bKc0a800020000001043a7a40b00007c8300000032
Content-Length: 0
Call-ID: E356C4F8-ED84-4B2A-81A5-503B2E3458ED@192.168.0.2
CSeq: 10 OPTIONS
From: sip:1000@192.168.0.249;tag=93242112147
Max-Forwards: 70
To: sip:192.168.0.249:5060

8 headers, 0 lines
Looking for 192.168.0.249:5060 in default
Dec 16 12:27:24 NOTICE[574]: pbx.c:1331 pbx_extension_helper: Cannot find extension context 'default’
Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.253:5060;branch=z9hG4bKc0a800020000001043a7a40b00007c8300000032
From: sip:1000@192.168.0.249;tag=93242112147
To: sip:192.168.0.249:5060;tag=as2c4a2d76
Call-ID: E356C4F8-ED84-4B2A-81A5-503B2E3458ED@192.168.0.2
CSeq: 10 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:192.168.0.249
Accept: application/sdp
Content-Length: 0

to 192.168.0.253:5060
Destroying call ‘E356C4F8-ED84-4B2A-81A5-503B2E3458ED@192.168.0.2’



#3

Heres my sip.conf


; Examples:
;
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the ‘s’ extension
;
;
;register => 2345:password@sip_proxy/1234
;
; Register 2345 at sip provider ‘sip_proxy’. Calls from this provider connec
; extension 1234 in extensions.conf default context, unless you define
; unless you configure a [sip_proxy] section below, and configure a context.
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.co
; Tip 2: Use separate type=peer and type=user sections for SIP providers
; (instead of type=friend) if you have calls in both direct

register => 0731290104:xxxxxxxxxx@sip.internode.on.net/0731290104

;externip = 200.201.202.203 ; Address that we’re going to put in outbound SI
; if we’re behind a NAT

                            ; The externip and localnet is used
                            ; when registering and communicating with other
                            ; that we're registered with
                            ; You may add multiple local networks.  A reason
                            ; are:

;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network

;-------------------------------------------------------------------------------

[nodephone]
type=friend
secret=xxxxxxxxxxxxxxx
username=0731290104
dtmfmode=rfc2833
qualify=yes
context=from-nodephone
fromuser=0731290104
fromdomain=sip.internode.on.net
canreinvite=no
host=sip.internode.on.net
disallow=all
allow=g729

[1000]
username=1000
type=friend
context=home
secret=xxxxxx
host=dynamic
port=5060
qualify=yes


#4

Here is my extensions.conf


TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:pass@provider

;
; Any category other than “General” and “Globals” represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a ‘_’
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXXXXXX would match normal 7 digit dialings,
; while _1NXXNXXXXXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension.
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred. One may include another
; context in the current one as well, optionally with a
; date and time. Included contexts are included in the order
; they are listed.
;
;[context]
;exten => someexten,priority,application(arg1,arg2,…)
;exten => someexten,priority,application,arg1|arg2…
;

[macro-outgoingnodephone]
exten => s,1,SetCallerID(MYNUMBER)
exten => s,2,SetCIDName(MYNAME)
exten => s,3,Dial,SIP/${ARG1}@nodephone|60|
exten => s,4,Congestion

[from-nodephone]
exten => 0731290104,1,SetCIDName,“NP - ${CALLERIDNAME}” ; Add NP infront of the callerid name
exten => 0731290104,2,SetCIDNum,“72${CALLERIDNUM}” ; Add 72 infront of calleridnum
exten => 0731290104,3,Dial,${INCOMING}|60|r ; Dial the internal extension for 60 seconds
exten => 0731290104,4,contention ; No answer play contention signal

[nodephone-numbers]
exten => _72,1,Marco(outgoingnodephone,${EXTEN:2})

[home]
include => nodephone-numbers