[Asterisk-Users] SIP problems

[Primoz Kragelj]

Hi,
thanks a lot. After adding username, password and changing
extensions.conf my very basic * setup works excellent. 

So, next milestone is establishing connection from my office to my home
server where my * resides...

Thanks and Regards,
  Primoz



-----Original Message-----
From: Bellows, Jared [mailto:jmbell at byu.edu] 
Sent: 2. maj 2005 22:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SIP problems

It looks like you haven't defined a "tina" extension.  You have the
"tina" SIP account set to be extension "1000".  If you want to dial
extension "tina" change "1000" to "tina". 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Primoz
Kragelj
Sent: Monday, May 02, 2005 12:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] SIP problems

Hi all,

I'm newbie to VOIP/SIP/asterisk... and I having problems with SIP on
local network. I have Freebsd server 5.3 running asterisk and two x-lite
clients.

I added following lines to sip.conf
[tina]
type=friend
host=dynamic
dtmfmode=inband
context=sip

[primozz]
type=friend
host=dynamic
dtmfmode=inband
context=sip

And following to extensions.conf
[sip]
exten => 1000,1,Dial,SIP/tina
exten => 2000,1,Dial,SIP/primozz

*CLI> sip show users
Username         Secret           Accountcode     Def.Context     ACL
NAT
primozz                                           sip             No
RFC35
tina                                              sip             No
RFC35


I have X-Lite clinet on Win XP and while trying to make call to "tina" I
got 404 error - not found. Same for vice versa...Both users are local.
>From debug below following line:
To: <sip:tina at 192.168.1.3>;tag=as1283188b
is very strange to me. Instead od 192.168.1.3 there should be
192.168.1.1.

Do I need to put some ware static IP for each client ?



And following is debug from asterisk:

Peer audio RTP is at port 192.168.1.3:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e
(gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe
(gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for tina in sip
list_route: hop: <sip:primozz at 192.168.1.3:5060>
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.1.3:5060;branch=z9hG4bKBD00F479A53B4C97B6FF319C9D6B06CA
From: Primoz <sip:primozz at 192.168.1.3>;tag=1716760483
To: <sip:tina at 192.168.1.3>;tag=as1283188b
Call-ID: B5CEF7FC-640F-42FD-B93E-FFDE1F9EE6F8 at 192.168.1.3
CSeq: 22324 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:tina at 192.168.1.12>
Content-Length: 0


 to 192.168.1.3:5060


Sip read:
ACK sip:tina at 192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.3:5060;rport;branch=z9hG4bKBD00F479A53B4C97B6FF319C9D6B06CA
From: Primoz <sip:primozz at 192.168.1.3>;tag=1716760483
To: <sip:tina at 192.168.1.3>;tag=as1283188b
Contact: <sip:primozz at 192.168.1.3:5060>
Call-ID: B5CEF7FC-640F-42FD-B93E-FFDE1F9EE6F8 at 192.168.1.3
CSeq: 22324 ACK
Max-Forwards: 70
Content-Length: 0


Thanks for help !

Regards,
  Primoz

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