[Asterisk-bsd] Help with Asterisk on Netbsd 2.0 on MIPS arch.

Tue Dec 20 00:30:47 CST 2005

Hello all,

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

I have been working on getting Asterisk to run on the Cobalt Cube 2
microserver it's a MIPS based cpu.
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.
With the changes I made Asterisk will not compile at all. 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 seem 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.

I have attached a "sip debug" log called "asterisk sip debug.txt".
If you need to see my sip.conf and my extensions.conf please let me know.

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.

Thank you all for your help with this matter,

Brendan
-------------- next part --------------

Sip read:
INVITE sip:7246302065 at 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 at 192.168.0.253:5060>
Call-ID: 235CB50F-9BC8-489D-B826-61B8808D3C25 at 192.168.0.2
Content-Type: application/sdp
CSeq: 1 INVITE
From: "1000"<sip:1000 at 192.168.0.249>;tag=92257810573
Max-Forwards: 70
To: <sip:7246302065 at 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 at 192.168.0.249>;tag=92257810573
To: <sip:7246302065 at 192.168.0.249>;tag=as4b934b4c
Call-ID: 235CB50F-9BC8-489D-B826-61B8808D3C25 at 192.168.0.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:7246302065 at 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 at 192.168.0.2' in 15000 ms
Found user '1000'

Sip read:
ACK sip:7246302065 at 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 at 192.168.0.2
CSeq: 1 ACK
From: "1000"<sip:1000 at 192.168.0.249>;tag=92257810573
Max-Forwards: 70
To: <sip:7246302065 at 192.168.0.249>;tag=as4b934b4c
User-Agent: SJphone/1.60.289a (SJ Labs)

9 headers, 0 lines

Sip read:
INVITE sip:7246302065 at 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 at 192.168.0.253:5060>
Call-ID: 235CB50F-9BC8-489D-B826-61B8808D3C25 at 192.168.0.2
Content-Type: application/sdp
CSeq: 2 INVITE
From: "1000"<sip:1000 at 192.168.0.249>;tag=92257810573
Max-Forwards: 70
To: <sip:7246302065 at 192.168.0.249>
User-Agent: SJphone/1.60.289a (SJ Labs)
Proxy-Authorization: Digest username="1000",realm="asterisk",nonce="17706795",uri="sip:7246302065 at 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 at 192.168.0.249>;tag=92257810573
To: <sip:7246302065 at 192.168.0.249>;tag=as4b934b4c
Call-ID: 235CB50F-9BC8-489D-B826-61B8808D3C25 at 192.168.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:7246302065 at 192.168.0.249>
Content-Length: 0

 to 192.168.0.253:5060

Sip read:
ACK sip:7246302065 at 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 at 192.168.0.2
CSeq: 2 ACK
From: "1000"<sip:1000 at 192.168.0.249>;tag=92257810573
Max-Forwards: 70
To: <sip:7246302065 at 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 at 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 at 192.168.0.2
CSeq: 10 OPTIONS
From: <sip:1000 at 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 at 192.168.0.249>;tag=93242112147
To: <sip:192.168.0.249:5060>;tag=as2c4a2d76
Call-ID: E356C4F8-ED84-4B2A-81A5-503B2E3458ED at 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 at 192.168.0.2'