404 Error


#1

Hi All,

Here you are the SIP debugging info from my X-lite phone, it’s catched while I was trying to call my FWD account:

SEND TIME: 15368463
SEND >> 192.168.0.1:5060
INVITE sip:727867@fwd.pulver.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5061;rport;branch=z9hG4bKEE1289F8C38543F7A7AC2BAB571A8A00
From: XXXXX sip:test@192.168.0.1:5061;tag=3667980216
To: sip:727867@fwd.pulver.com
Contact: sip:test@192.168.0.10:5061
Call-ID: F58A95EE-C49E-4B8E-8D5E-C0A03C5A18D5@192.168.0.10
CSeq: 64192 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
Content-Length: 305

v=0
o=test 15368379 15368463 IN IP4 192.168.0.10
s=X-Lite
c=IN IP4 192.168.0.10
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

RECEIVE TIME: 15368481
RECEIVE << 192.168.0.1:5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.10:5061;rport;branch=z9hG4bKEE1289F8C38543F7A7AC2BAB571A8A00;received=192.168.0.10
From: XXXXX sip:test@192.168.0.1:5061;tag=3667980216
To: sip:727867@fwd.pulver.com;tag=as7595564e
Call-ID: F58A95EE-C49E-4B8E-8D5E-C0A03C5A18D5@192.168.0.10
CSeq: 64192 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:727867@192.168.0.1
Proxy-Authenticate: Digest realm=“asterisk”, nonce="02de2778"
Content-Length: 0

SEND TIME: 15368483
SEND >> 192.168.0.1:5060
ACK sip:727867@fwd.pulver.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5061;rport;branch=z9hG4bKEE1289F8C38543F7A7AC2BAB571A8A00
From: XXXXX sip:test@192.168.0.1:5061;tag=3667980216
To: sip:727867@fwd.pulver.com;tag=as7595564e
Contact: sip:test@192.168.0.10:5061
Call-ID: F58A95EE-C49E-4B8E-8D5E-C0A03C5A18D5@192.168.0.10
CSeq: 64192 ACK
Max-Forwards: 70
Content-Length: 0

SEND TIME: 15368492
SEND >> 192.168.0.1:5060
INVITE sip:727867@fwd.pulver.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5061;rport;branch=z9hG4bKC0B376B63CC64054A1D921AABEE6B99B
From: tony sip:test@192.168.0.1:5061;tag=3667980216
To: sip:727867@fwd.pulver.com
Contact: sip:test@192.168.0.10:5061
Call-ID: F58A95EE-C49E-4B8E-8D5E-C0A03C5A18D5@192.168.0.10
CSeq: 64193 INVITE
Proxy-Authorization: Digest username=“test”,realm=“asterisk”,nonce=“02de2778”,response=“073e27359071308ee82c0b457b209e94”,uri="sip:727867@fwd.pulver.com"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
Content-Length: 305

v=0
o=test 15368379 15368463 IN IP4 192.168.0.10
s=X-Lite
c=IN IP4 192.168.0.10
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

RECEIVE TIME: 15368495
RECEIVE << 192.168.0.1:5060
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.10:5061;rport;branch=z9hG4bKC0B376B63CC64054A1D921AABEE6B99B;received=192.168.0.10
From: XXXX sip:test@192.168.0.1:5061;tag=3667980216
To: sip:727867@fwd.pulver.com;tag=as7595564e
Call-ID: F58A95EE-C49E-4B8E-8D5E-C0A03C5A18D5@192.168.0.10
CSeq: 64193 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:727867@192.168.0.1
Content-Length: 0

SEND TIME: 15368497
SEND >> 192.168.0.1:5060
ACK sip:727867@fwd.pulver.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5061;rport;branch=z9hG4bKC0B376B63CC64054A1D921AABEE6B99B
From: XXXX sip:test@192.168.0.1:5061;tag=3667980216
To: sip:727867@fwd.pulver.com;tag=as7595564e
Contact: sip:test@192.168.0.10:5061
Call-ID: F58A95EE-C49E-4B8E-8D5E-C0A03C5A18D5@192.168.0.10
CSeq: 64193 ACK
Max-Forwards: 70
Content-Length: 0

SIP.CONF

[test]
type=friend
username=test
secret=xxxxx
context=default
nat=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw

EXTENSIONS.CONF

[default]
exten => test, 1, Dial(SIP/test@192.168.0.10:5061)

I think the configuration is OK, but I permanently get 404 error and actually I can make any Outbound SIP calls , as well I can’t receive any :frowning:

Any suggestions please?


#2

Have you got canreinvite=no configured somewhere in sip.conf? If you haven’t got that in the relevant place(s), it’s not going to work.


#3

yes, I have canreinvite=no, but still no success


#4

Have you read this page and followed what it says?

voip-info.org/wiki/view/Asterisk … ect+to+FWD


#5

Thanks, I did read it, but still there is no clear answer why my current Asterisk setup doesn’t work. I don’t see why I have to define FWD as a peer in the SIP.CONF of my Asterisk. Simply I want to use that FWD account for Asterisk testing . Isn’t that possible and why? Theoritically anybody using SIP phone ( hardware or software) should be able to call my Asterisk and vice versa! That’s why I am wondering what’s strange with my current Asterisk setup. In my understanding Asterisk must accept all incoming calls no matter who or where are originating them and to be able to call any other SIP phone on the net as well? I haven’t put any restrictions anywhere in the Asterisk configs.


#6

I can’t see that you’re likely to get it to work unless you configure FWD in sip.conf. Part of the reason is because your Asterisk will have to register with FWD’s server before it’s allowed to place calls through it. I don’t know anything about FWD, but this is how most ITSPs work.

What’s the problem with configuring it in sip.conf, anyway?


#7

There is no problem to configure FWD in the SIP.CONF.
I think you didn’t get what I wanted to say. I don’t want to register to the FWD and make calls through it! Just wanted to test calls between the FWD account and the X-lite behind my Asterisk. The problem is not related to the FWD, Gizmo or whatever registration in my sip.conf.
The problem is that I can’t dial out or receive any calls on the X-lite phone. I always get 404 error when I call any SIP number and also nobody can reach me. You have my configuration files and you can see there is nothing wrong, also I opened all required UDP ports on my firewall, but no success to get it working :frowning:


#8

Oh, ok. I understand you now.

Well, i can’t see there’s nothing wrong with your config files, because i don’t know the whole story. And (hopefully) you’ve only posted part of your sip.conf. There’s obviously something wrong with your configuration somewhere - otherwise it would work. Because x-lite does work with Asterisk, and Asterisk does work with FWD - so all that’s left is your configuration or some firewall problem (which is still probably configuration).

That looks like you’re using NAT, but haven’t got nat=yes and externip set up. Is that right? If so, that’s probably the problem.


#9

My Asterisk is running on the NAT/Router gateway, so it has two interfaces - one internal to the LAN 192.168.0.1/24 and one external to the Interent. You remember my post about Asterisk and NAT, we have discussed there where is possbile or not to have Asterisk running on the NAT/Router gateway.
So according this setup I don’t use NAT, because the Asterisk is acting as a proxy server between my LAN and the X-lite on it and the external world, isn’t it? That’s why NAT=no and don’t have externip configured in the NAT section of SIP.CONF.


#10

Ah, i think i can see what you’re doing now. I’m not that familiar with the format of the output from sip debug, so i didn’t spot it earlier.

It looks to me like you’re dialling the following string on the x-lite:

727867@fwd.pulver.com

but you haven’t got an extension called that configured in extensions.conf - therefore you get the 404 response.

You can only dial extensions that actually exist. Asterisk won’t try and interpret the extension string and see if it can make something else out of it other than an extension. It’s a pbx, not a sip proxy.

So if you want to be able to make a call to 727867@fwd.pulver.com, you’ll have to configure an extension for that purpose - e.g.

exten => 123456,1,Dial(SIP/727867@fwd.pulver.com)

and then dial that extension (123456 in this example) on the x-lite.


#11

I did what you told me, I put the extension for FWD in the extension.conf. But still not working and X-lite reports the following error: “503 Service not availbale”

Btw I noticed something in my CLI:

Dec 26 08:42:23 WARNING[18962]: channel.c:2520 ast_request: No channel type regi
stered for '(SIP’
Dec 26 08:42:23 NOTICE[18962]: app_dial.c:1011 dial_exec_full: Unable to create
channel of type ‘(SIP’ (cause 66 - Channel not implemented)
Dec 26 08:45:11 WARNING[18978]: channel.c:2520 ast_request: No channel type registered for '(SIP’
Dec 26 08:45:11 NOTICE[18978]: app_dial.c:1011 dial_exec_full: Unable to create channel of type ‘(SIP’ (cau
se 66 - Channel not implemented)

Probably this could give an idea of what’s going on with the Asterisk setup?


#12

Strewth! This is such a simple thing to do, i can’t believe it’s taken us this long to work it out! :wink:

Please select and paste the relevant lines from extensions.conf .


#13

Here you are what I have in my extensions.conf:

[general]

static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[default]

exten => test,1,Dial(SIP/test@192.168.0.10)
exten => 123456,1,Dial,(SIP/727867@fwd.pulver.com)


#14

For a start, take the comma out from between the “Dial” and the “(”.


#15

Hi ,

I have the sampe issue with my sip server, have you solved your problem, pleas help.

Regards,
Rami