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Unread 07-27-2007, 05:00 AM   #1
samjesse
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Default SIP Registrations

Hi

what is "SIP Registrations", does it affect me, I just have a VoXaLot basic and do not need webcall back and the forwarding things.

thanks
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Unread 07-27-2007, 06:10 AM   #2
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SIP registrations are only required for in-bounds, that is if you want people to be able to call your VSP provided number or you want to set up a Direct Dial-in Number (DID)

Tony.
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Unread 07-27-2007, 01:58 PM   #3
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I'm trying to call my voxalot accout from another domain but the voxalot server require authorization. Is it normal? I would receive free call from anyone.
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Unread 07-27-2007, 02:27 PM   #4
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Default change "From" domain

Quote:
Originally Posted by stefanop View Post
I'm trying to call my voxalot accout from another domain but the voxalot server require authorization. Is it normal? I would receive free call from anyone.
VoXaLot's SIP proxy responds with "407 Proxy Authentication Required" if the domain in the "From:" header is "us.voxalot.com" etc, because it thinks the caller is a VoXaLot user wishing to make a call through VoXaLot.

The SIP INVITE packet would look like this:
Code:
INVITE sip:12xxxx@us.voxalot.com SIP/2.0
Via: SIP/2.0/UDP 221.xx.xx.xx:5060;branch=zh4K500614582
From: 66xxxx <sip:66xxxx@us.voxalot.com>;tag=06810
...

You need to change the "From:" part to something else, like this:
Code:
INVITE sip:12xxxx@us.voxalot.com SIP/2.0
Via: SIP/2.0/UDP 221.xx.xx.xx:5060;branch=zh4K500614582
From: 66xxxx <sip:66xxxx@sip.pennytel.com>;tag=06810
...
For this SIP INVITE, VoXaLot will (attempt to) connect the call to VoXaLot user 12xxxx, without challenging for credentials.
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Unread 07-27-2007, 02:56 PM   #5
stefanop
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This is the response I got.

Code:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 212.109.160.59:5060;branch=z9hG4bK31c28521;rport=5060
From: "1001" <sip:1001@sip.omnianet.it>;tag=as66afa6f1
To: <sip:190475@voxalot.com>;tag=as3ca8a686
Call-ID: 0cefea390cf655587eb649534aa63d66@sip.omnianet.it
CSeq: 103 INVITE
Proxy-Authenticate: Digest realm="voxalot.com", nonce="46aa07b5c2f912c19133ae6f638842b98e7712cb"
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 64.34.173.199:5060 "Noisy feedback tells:  pid=15434 req_src_ip=212.109.160.59 req_src_port=5060 in_uri=sip:190475@64.34.163.35:5061 out_uri=sip:190475@64.34.163.35:5061 via_cnt==1"
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Unread 07-27-2007, 05:25 PM   #6
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Default

Quote:
Originally Posted by samjesse View Post
Hi

what is "SIP Registrations", does it affect me?
A very few VSPs, like MyNetFone (and Oztell, too, now, apparently) require registration even for outgoing calls, so, if you are planning to use these providers via Voxalot, you will need Premium.
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Unread 07-28-2007, 12:41 AM   #7
v164
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Default works for me

Quote:
Originally Posted by stefanop View Post
This is the response I got.

Code:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 212.xxx.xxx.xxx:5060;branch=z9hG4bK31c28521;rport=5060
From: "1001" <sip:1001@sip.omnianet.it>;tag=as66afa6f1
To: <sip:19xxxx@voxalot.com>;tag=as3ca8a686
Call-ID: 0cefea390cf655587eb649534aa63d66@sip.omnianet.it
CSeq: 103 INVITE
...

I've just tried calling my own VoXaLot number, via SIP, and it works for me:

SEND >> 64.34.163.35:5060
Code:
INVITE sip:66xxxx@voxalot.com SIP/2.0
Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z12JUKS83L3KS5
From: 1001 <sip:1001@sip.omnianet.it>;tag=764466127
To: <sip:66xxxx@voxalot.com>
Contact: <sip:1001@221.xx.xx.yyy:5060>
Call-ID: 98KS32432@221.xx.xx.yyy
CSeq: 61224 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 194

v=0
o=1001 1687423 1687496 IN IP4 221.xx.xx.yyy
s=X-Lite
c=IN IP4 221.xx.xx.yyy
t=0 0
m=audio 29360 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

RECEIVE << 64.34.163.35:5060
Code:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z12JUKS83L3KS5
From: 1001 <sip:1001@sip.omnianet.it>;tag=764466127
To: <sip:66xxxx@voxalot.com>
Call-ID: 98KS32432@221.xx.xx.yyy
CSeq: 61224 INVITE
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 64.34.163.35:5060 "Noisy feedback tells:  pid=30944 req_src_ip=221.xx.xx.yyy req_src_port=5060 in_uri=sip:66xxxx@voxalot.com out_uri=sip:66xxxx@voxalot.com via_cnt==1"


RECEIVE << 64.34.163.35:5060
Code:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z12JUKS83L3KS5
From: 1001 <sip:1001@sip.omnianet.it>;tag=764466127
To: <sip:66xxxx@voxalot.com>
Call-ID: 98KS32432@221.xx.xx.yyy
CSeq: 61224 INVITE
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 64.34.163.35:5060 "Noisy feedback tells:  pid=30944 req_src_ip=221.xx.xx.yyy req_src_port=5060 in_uri=sip:66xxxx@voxalot.com out_uri=sip:*01066xxxx@64.34.173.199:5060 via_cnt==1"

RECEIVE << 64.34.163.35:5060
Code:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z12JUKS83L3KS5
Record-Route: <sip:64.34.173.199;lr=on;ftag=764466127>
Record-Route: <sip:64.34.163.35;lr=on;ftag=764466127>
From: 1001 <sip:1001@sip.omnianet.it>;tag=764466127
To: <sip:66xxxx@voxalot.com>;tag=1402227574
Call-ID: 98KS32432@221.xx.xx.yyy
CSeq: 61224 INVITE
Contact: <sip:66xxxx@221.xx.xx.xx:6080>
Content-Length: 0

Could you provide more information on your setup?


Which VoXaLot server is your SIP device registering with?

Have you got call forwarding active?

Could you provide a copy of the SIP INVITE message SIP device is sending?
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Unread 07-28-2007, 07:19 AM   #8
stefanop
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Hello. Here is the invite. On this number there is just the voicemail.

Code:
INVITE sip:190475@64.34.163.35:5061 SIP/2.0
Via: SIP/2.0/UDP 212.109.160.59:5060;branch=z9hG4bK2d21db3b;rport
Route: 
<sip:64.34.173.199;lr=on;ftag=as75fcf5be>,<sip:64.34.163.35;lr=on;ftag=as75fcf5be>
From: "1001" <sip:1001@omnianet.it>;tag=as75fcf5be
To: <sip:190475@voxalot.com>;tag=as2d2dbc73
Contact: <sip:1001@212.109.160.59>
Call-ID: 5478b0226d9238c54aaf9ea052283918@omnianet.it
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 309

v=0
o=root 2200 2201 IN IP4 192.168.10.65
s=session
c=IN IP4 192.168.10.65
t=0 0
m=audio 16152 RTP/AVP 0 8 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
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Unread 07-28-2007, 09:41 AM   #9
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Default Sip registration

Quote:
Originally Posted by Tony View Post
SIP registrations are only required for in-bounds, that is if you want people to be able to call your VSP provided number or you want to set up a Direct Dial-in Number (DID)

Tony.

Hi,

I wonder where I should set the SIP settings?
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Unread 07-28-2007, 09:44 AM   #10
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Default Sip registration

Quote:
Originally Posted by tipmix View Post
Hi,

I wonder where I should set the SIP settings?
Sorrry, I meant the Sip registration.
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