Hello all
im trying to install my first sip line but in vain.My customer gave me a static ip and a router with nat and firewall disabled so im using the lan2 interface and none for network topology.I dont use stun (anyway there is always a blocking firewall and i cant ping the stun server,normal?).Is it right that all the devices are connected in lan1 and internet in lan2?is there any interconnection between 2 interfaces or i should connect internet in lan1?
Right now the line seems to register from monitor but all my invite requests end up with bad request receive message and i can not see the digits i dialled in the from header...
Im pretty sure that im doing something reaally wrong so anyone who could help?
Thanx
 

sorry... can not see the digits i dialled in the TO header

if your LAN2 is behind Nat then you need to open Port 5060 to the LAN2 IP address.  You do not necessarily need Stun on Startup, but do enter your public IP address and port 5060 is the network settings.
You should be able to ping the STUN server (are you pinging localling on your PC or using Ping in SSA)
LAN1 does not need internet access for SIP.
You need a default route 0.0.0.0/255.255.255.255 via your router on LAN2.

Once you have all this you can create your Trunk and valid URI, along with SC to dial out but you are better off checking the Help file for this.

Post your trace from Monitor (SIP, ERROR & Print only) for help.


 

i ping the stun server from my computer with no response.Right now i dont use the stun thing at all and the line is  registered succesfully (i get rx ok when i send tx register)
Here is the monitor capture when im dialling


96314306mS SIP Trunk: 18:Tx
                    INVITE Tel:+ SIP/2.0
                    Via: SIP/2.0/UDP 81.92.50.203:5060;rport;branch=z9hG4bK12166cde2ae2c2a16aca3b63ee3e05ba
                    From: "wemlinessa" <sip:wemlinessa@sip1.voipbuster.com >;tag=140d50fcab213f24
                    To: Tel:+
                    Call-ID: 616b79a3a1f3f7cbac3a884f21ab610d@81.92.50.203
                    CSeq: 1157301279 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: 276
                    
                    v=0
                    o=UserA 749719227 4210276907 IN IP4 81.92.50.203
                    s=Session SDP
                    c=IN IP4 81.92.50.203
                    t=0 0
                    m=audio 49152 RTP/AVP 0 18 8 101
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:18 G729/8000
                    a=rtpmap:8 PCMA/8000
                    a=fmtp:18 annexb = no
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-15
  96314306mS SIP Tx: UDP 81.92.50.203:5060 -> 194.120.0.198:5060
                     INVITE Tel:+ SIP/2.0
                     Via: SIP/2.0/UDP 81.92.50.203:5060;rport;branch=z9hG4bK12166cde2ae2c2a16aca3b63ee3e05ba
                     From: "wemlinessa" <sip:wemlinessa@sip1.voipbuster.com >;tag=140d50fcab213f24
                     To: Tel:+
                     Call-ID: 616b79a3a1f3f7cbac3a884f21ab610d@81.92.50.203
                     CSeq: 1157301279 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: 276
                     
                     v=0
                     o=UserA 749719227 4210276907 IN IP4 81.92.50.203
                     s=Session SDP
                     c=IN IP4 81.92.50.203
                     t=0 0
                     m=audio 49152 RTP/AVP 0 18 8 101
                     a=rtpmap:0 PCMU/8000
                     a=rtpmap:18 G729/8000
                     a=rtpmap:8 PCMA/8000
                     a=fmtp:18 annexb = no
                     a=rtpmap:101 telephone-event/8000
                     a=fmtp:101 0-15
  96314307mS SipDebugInfo: 18.6407.0 1661 SIPTrunk Endpoint(f560d6b4) UpdateSIPCallState SIPDialog::INITIAL(0) -> SIPDialog::INVITE_SENT(1)
  96314308mS SipDebugInfo: 18.6407.0 1661 SIPTrunk Endpoint(f560d6b4) UpdateSDPState SIPDialog::IDLE(0) -> SIPDialog::OFFER_SENT(1)
  96314308mS CD: CALL: 0.6405.0 BState=Idle Cut=0 Music=0.0 Aend="Prokopis(132)" (20.1) Bend="Line 18" [Line 18] (0.0) CalledNum=4 () CallingNum=132 (Prokopis) Internal=0 Time=1023 AState=Dialling
  96314310mS CMMap: a=8.8 b=0.0 B0
  96314382mS 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=z9hG4bK12166cde2ae2c2a16aca3b63ee3e05ba
                     To: <Tel:+>
                     Contact: sip:194.120.0.198:5060
                     Call-ID: 616b79a3a1f3f7cbac3a884f21ab610d@81.92.50.203
                     CSeq: 1157301279 INVITE
                     Server: (Very nice Sip Registrar/Proxy Server)
                     Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
                     Content-Length: 0
                     
  96314382mS SIP Trunk: 18:Rx
                    SIP/2.0 400 Bad request
                    Via: SIP/2.0/UDP 81.92.50.203:5060;rport;branch=z9hG4bK12166cde2ae2c2a16aca3b63ee3e05ba
                    To: <Tel:+>
                    Contact: sip:194.120.0.198:5060
                    Call-ID: 616b79a3a1f3f7cbac3a884f21ab610d@81.92.50.203
                    CSeq: 1157301279 INVITE
                    Server: (Very nice Sip Registrar/Proxy Server)
                    Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
                    Content-Length: 0
                    
  96314384mS SipDebugInfo: SIPDialog TXN: Decoding of message Failed 2032
  96314384mS SipDebugInfo: SIP Line (18): Error in decoding packet
  96314610mS CMMap: a=8.8 b=0.0 B1

disable Use Tel in the Trunk settings

Your short code is also wrong,
should be in format
N;
N"@sip1.voipbuster.com"

teluri was disabled,i create the short code as you told me and i m getting no rx messages now..no registration?Shall i create a short code at ars as well? Is it normal that there are no digits displayed in the to header? thanx for your prompt replies

100254331mS SIP Trunk: 18:Tx
                    INVITE sip:@sip1.voipbuster.com SIP/2.0
                    Via: SIP/2.0/UDP 81.92.50.203:5060;rport;branch=z9hG4bK0f2e3a09555fefe68bd05bc7a90560cb
                    From: "wemlinessa" <sip:wemlinessa@sip1.voipbuster.com >;tag=0b662ec27b0e818b
                    To: <sip:@sip1.voipbuster.com>
                    Call-ID: 2f857ac74474a3ec63f86ac795515361@81.92.50.203
                    CSeq: 1784114473 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: 276
                    
                    v=0
                    o=UserA 473367281 1587980570 IN IP4 81.92.50.203
                    s=Session SDP
                    c=IN IP4 81.92.50.203
                    t=0 0
                    m=audio 49152 RTP/AVP 0 18 8 101
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:18 G729/8000
                    a=rtpmap:8 PCMA/8000
                    a=fmtp:18 annexb = no
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-15
 100254331mS SIP Tx: UDP 81.92.50.203:5060 -> 194.120.0.198:5060
                     INVITE sip:@sip1.voipbuster.com SIP/2.0
                     Via: SIP/2.0/UDP 81.92.50.203:5060;rport;branch=z9hG4bK0f2e3a09555fefe68bd05bc7a90560cb
                     From: "wemlinessa" <sip:wemlinessa@sip1.voipbuster.com >;tag=0b662ec27b0e818b
                     To: <sip:@sip1.voipbuster.com>
                     Call-ID: 2f857ac74474a3ec63f86ac795515361@81.92.50.203
                     CSeq: 1784114473 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: 276
                     
                     v=0
                     o=UserA 473367281 1587980570 IN IP4 81.92.50.203
                     s=Session SDP
                     c=IN IP4 81.92.50.203
                     t=0 0
                     m=audio 49152 RTP/AVP 0 18 8 101
                     a=rtpmap:0 PCMU/8000
                     a=rtpmap:18 G729/8000
                     a=rtpmap:8 PCMA/8000
                     a=fmtp:18 annexb = no
                     a=rtpmap:101 telephone-event/8000
                     a=fmtp:101 0-15

okay, reverse up, what short code had you been using

with this short code
4n
n
dial
i get bad request...but at least i get a response haha

try 4N;
N"@sip1.voipbuster.com"
Dial Trunk Group Outgoing Unique Trunk Group that your URI is in.

<or N"@ipaddress of provider">

i tried this
N"@sip1.voipbuster.com"
i tried this
N"@sip1.voipbuster.com:user=phone"
i tried this
N"@ipadress"
nop
can you please explain why i can not see the digigs that i dialled
?

are you sending 4N to ARS or direct to the URI Trunk?

If using ARS:
4N
N
ARS Table

ARS then should be:

N;
N"@sip1.voipbuster.com"
Trunk Number set in URI Tab of SIP Line

If not using ARS then use the short code in my earlier post.

You are not seeing digits because you are not sending them correctly via a valid short code method.  Always post the monitor output with each post.

 

Have a look here so you know what you're doing.

http://members.optusnet.com.au/hagoo/iporesource/Example%20SIP%20configurations%20for%20Release%204%20by%20Mr%20IPO.pdf

Greetzzz...Bas

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It works! Now if only I could remember what I did...
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