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SIP CONFIGURATION PLEASE HELP 4

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n1k05

n1k05

Technical User
Sep 23, 2008
16
GR
Hello
im connecting through lan2 on a router with static ip and no firewall and nat.The lan2 has static ip as well.My short code is
4N
DIAL
N
51[4]
and the ars 51[4] has the following short code
N;
DIAL
N"@voipbusterip"
When im trying to call i get this from monitor
473403mS SIP Trunk: 18:Tx
INVITE sip:00302109211180@194.120.0.198 SIP/2.0
Via: SIP/2.0/UDP 81.92.50.203:5060;rport;branch=z9hG4bKed26e79dc7a31127cfc0f42401ad70fa
From: "wemlinessa" <sip:wemlinessa@sip1.voipbuster.com >;tag=a1d6cd2e9fda19ac
To: <sip:00302109211180@194.120.0.198>
Call-ID: bbd7090cc93ac130aad15e4c00dc35cb@81.92.50.203
CSeq: 1941792214 INVITE
Contact: "wemlinessa" <sip:wemlinessa@81.92.50.203:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Type: application/sdp
Content-Length: 277

v=0
o=UserA 3501642207 3239865548 IN IP4 81.92.50.203
s=Session SDP
c=IN IP4 81.92.50.203
t=0 0
m=audio 49154 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=fmtp:18 annexb = no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
473403mS SIP Tx: UDP 81.92.50.203:5060 -> 194.120.0.198:5060
INVITE sip:00302109211180@194.120.0.198 SIP/2.0
Via: SIP/2.0/UDP 81.92.50.203:5060;rport;branch=z9hG4bKed26e79dc7a31127cfc0f42401ad70fa
From: "wemlinessa" <sip:wemlinessa@sip1.voipbuster.com >;tag=a1d6cd2e9fda19ac
To: <sip:00302109211180@194.120.0.198>
Call-ID: bbd7090cc93ac130aad15e4c00dc35cb@81.92.50.203
CSeq: 1941792214 INVITE
Contact: "wemlinessa" <sip:wemlinessa@81.92.50.203:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Type: application/sdp
Content-Length: 277

v=0
o=UserA 3501642207 3239865548 IN IP4 81.92.50.203
s=Session SDP
c=IN IP4 81.92.50.203
t=0 0
m=audio 49154 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=fmtp:18 annexb = no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
473485mS SIP Rx: UDP 194.120.0.198:5060 -> 81.92.50.203:5060
SIP/2.0 400 Bad request
Via: SIP/2.0/UDP 81.92.50.203:5060;rport;branch=z9hG4bKed26e79dc7a31127cfc0f42401ad70fa
To: <sip:00302109211180@194.120.0.198>
Contact: sip:00302109211180@194.120.0.198:5060
Call-ID: bbd7090cc93ac130aad15e4c00dc35cb@81.92.50.203
CSeq: 1941792214 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

473485mS SIP Trunk: 18:Rx
SIP/2.0 400 Bad request
Via: SIP/2.0/UDP 81.92.50.203:5060;rport;branch=z9hG4bKed26e79dc7a31127cfc0f42401ad70fa
To: <sip:00302109211180@194.120.0.198>
Contact: sip:00302109211180@194.120.0.198:5060
Call-ID: bbd7090cc93ac130aad15e4c00dc35cb@81.92.50.203
CSeq: 1941792214 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

473486mS SipDebugInfo: SIPDialog TXN: Decoding of message Failed 2032
473487mS SipDebugInfo: SIP Line (18): Error in decoding packet
475404mS SipDebugInfo: Timer 1 callback
475404mS SipDebugInfo: Dialog has been deleted

From the telephone i get a message 'waiting for the line'.I connected my laptop directly to the router and i could make a call with voipbuster.
From monitor i also get this

784184mS SipDebugInfo: SIPTrunks: Make Target voip, line group id is 1 and ip 194.120.0.198
784184mS SipDebugInfo: SIP Line (18) cannot find a suitable SIP URI to dial out

The sip uri im using is with use authentication and line group 18 like the number of my sip line.
I dont know the dial plan of voipbuster and i can not find even a mail to communicate with them.Does anyone know whta dial plan voipbuster follows?
Stun shows blocking firewall even though im connecting from a static ip to a router with static ip and no firewall at all.Thats why im using none in network topology and i created an ip route like that
0.0.0.0.
0.0.0.0.
routerip
Please...ideas?
 
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You're using sip1.voipbuster.com better try to use sip.voipbuster.com or 194.221.62.198

On the sip line make sure you've ticked
"Registration Required"
"In Service"
"Use Offerers Codec"
Fill in the ITSP Domain; sip.voipbuster.com
ITSP IP; 194.221.62.198
Leave the rest default.
Add a SIP URI in the next tab leave it on
"Use Authentication Name" change the in/out going group ID to a non used number lets say 99

Make a ShortCode like
Code; ?
Feature; Dial
Telephone Num; .
LineGr ID; Your ARS

Code; N;
Feature; Dial
Telephone Num; N"@sip.voipbuster.com"
LineGr ID; 99 (see your SIP URI)

You IP route looks OK

In the System tab LAN1 or 2 on the Network Topology tab

Make the Stun 0.0.0.0
Firewall "Open Internet"
And your Public IP
The Public port must be on 5060.

Send it back with a reboot when it's up wait e few seconds then it must work.


Greetzzz...Bas

y1pzZTEUdok1vrI5cLb3FdPX4PgTPlSONkb5WPjz0x50etSujaMSmhdRCbOx9vASnrRNzzXv0IxNQA

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
Upvote 0 Downvote
Make sure you add the ARS code to the Main ARS (50)
If you want to dial out with the 8 over VoipBuster then use the Code 8N; (don't forget the ; )


y1pzZTEUdok1vrI5cLb3FdPX4PgTPlSONkb5WPjz0x50etSujaMSmhdRCbOx9vASnrRNzzXv0IxNQA

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
Upvote 0 Downvote
Make sure your IP route points to the internal ip of you Router.

y1pzZTEUdok1vrI5cLb3FdPX4PgTPlSONkb5WPjz0x50etSujaMSmhdRCbOx9vASnrRNzzXv0IxNQA

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
Upvote 0 Downvote
Forgot to add for the SIP line

Make sure you point Use Network Topology Info: to the right LAN other wise it will never go out.

y1pzZTEUdok1vrI5cLb3FdPX4PgTPlSONkb5WPjz0x50etSujaMSmhdRCbOx9vASnrRNzzXv0IxNQA

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
Upvote 0 Downvote
I don't like writing....:p

y1pzZTEUdok1vrI5cLb3FdPX4PgTPlSONkb5WPjz0x50etSujaMSmhdRCbOx9vASnrRNzzXv0IxNQA

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
Upvote 0 Downvote
oh yes the Taker it is the same issue
 
Upvote 0 Downvote
I guess it wont work till stun gives a result..
The company has a linux server for nat and firewall,in order to operate throuh nat in lan1 we need to direct all the traffic in 5060 port to the ip of the ipoffice right? Is there anything else considering nat configuration and firewall?
Thank you guys for your prompt replies
 
Upvote 0 Downvote
Voipbuster won't use STUN.
Have you tried the following like i posted before?

Make sure you point Use Network Topology Info: to the right LAN other wise it will never go out.

y1pzZTEUdok1vrI5cLb3FdPX4PgTPlSONkb5WPjz0x50etSujaMSmhdRCbOx9vASnrRNzzXv0IxNQA

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
Upvote 0 Downvote
i tried all the tips you sent me but in vain... i get all the time receivemessage bad request.Right now the it manager gives me internet access through lan1 and lan1 i use in network topology.So the problem becomes a network problem haha..If stun wont work for voipbuster why do they give an ip for stun?
Im sure that you all are more experienced than me in sip but still think that this blocking firewall in stun is the problem..Because the it manager wants spefication of that can you please tell me in general what do we ask for nat and firewall?
 
Upvote 0 Downvote
Normally you must be able to make outgoing call over SIP, or you have a provider who's blocking 5060.
Incomming calls could give problems NAT/Firewall.

y1pzZTEUdok1vrI5cLb3FdPX4PgTPlSONkb5WPjz0x50etSujaMSmhdRCbOx9vASnrRNzzXv0IxNQA

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
Upvote 0 Downvote
turn off the "Run stun on startup". Enter your public IP address and leave port as 5060. SEt the Firewall type as Full Cone NAT.

Enter 0.0.0.0 in Stun.

5060 is all that needs to be open and forwarded to the IPO but that is for incoming calls only.

Firewall needs to allow outgoing 5060 requests from IPO but you are already getting responses so this is ok.
 
Upvote 0 Downvote
ITS WORKING(only outgoing yet!!)
TheTaker and Bas really thank you. Its working with none network topology,iproute 0.0.0.0,0.0.0.0,router ip,shortcode
4n,dial,4n,51[ars] and ars 4n;,dial,n"@sip1.voipbuster.com".
I didnt try incoming yet because the customer was ready to kill me with all these reboots.
Thanx again guys
 
Upvote 0 Downvote
Glad that you've got it working.

For incomming calls make sure on the firewall port 5060 is open.
In the ICR you only need to fill in the voipbuster name so don't use @sip.voipbuster.com just the name.


Greetzzz...Bas

y1pzZTEUdok1vrI5cLb3FdPX4PgTPlSONkb5WPjz0x50etSujaMSmhdRCbOx9vASnrRNzzXv0IxNQA

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
Upvote 0 Downvote
Hi N1k05,

just joined voipbuster, but struggling to make it work on a Avaya 406v2 with 4.2 software. Any help would be appreciated. A copy of your config would be even better? thanks tom.

tom@pcsgib.com
 
Upvote 0 Downvote
@tomboy10

Better start your own thread.

y1pzZTEUdok1vrI5cLb3FdPX4PgTPlSONkb5WPjz0x50etSujaMSmhdRCbOx9vASnrRNzzXv0IxNQA

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
Upvote 0 Downvote
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