Sip Direct to IP 1

[IPGuru]

Our sip provider wants to send calls direct to our public IP address without any registration.

I have never needed to configure SIP like this before. is it possible if so how do i configure the Sip trunk
IPO V8.1

A Maintenance contract is essential, not a Luxury.
Do things on the cheap & it will cost you dear

 

[intrigrant]

Just set it up as any other SIP trunk but do not set "registration required".
The firewall must forward all incoming traffic ( or at least what is needed for SIP ) comming from the providers ip address to IP Office.

 

[IPGuru]

Thanks intrirant, that's what I tried but My PO does not seem to respond to the inbound call


currently configured with a default sip trunk
set itsp proxy & Istp domain
added sip uri of *,*,*,None

this is the trace of an inbound call, can you tell if the issue is me or the ISP?

2014-02-17      93291mS SIP Rx: UDP 87.127.234.38:5060 -> 192.168.99.200:5060
                    INVITE sip:44xxxxxxxxxx@yy.yy.yy.yyy SIP/2.0
                    Via: SIP/2.0/UDP 87.127.234.38:5060;branch=z9hG4bK65f101df;rport
                    Max-Forwards: 70
                    From: "07xxxxxxxxx" <sip:07xxxxxxxxx@87.127.234.38>;tag=as18e90dd1
                    To: <sip:44xxxxxxxxxx@yy.yy.yy.yy>
                    Contact: <sip:07xxxxxxxxx@87.127.234.38:5060>
                    Call-ID: 68584cbe2667d9501905b19476df0761@87.127.234.38:5060
                    CSeq: 102 INVITE
                    User-Agent: Entanet Media Server
                    Date: Mon, 17 Feb 2014 17:05:41 GMT
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                    Supported: replaces
                    X-DNQ: 44xxxxxxxxxx
                    Content-Type: application/sdp
                    Content-Length: 389
                    
                    v=0
                    o=root 930738635 930738635 IN IP4 87.127.234.38
                    s=Asterisk PBX 1.8.7.1
                    c=IN IP4 87.127.234.38
                    t=0 0
                    m=audio 13960 RTP/AVP 0 9 8 18 111 101
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:9 G722/8000
                    a=rtpmap:8 PCMA/8000
                    a=rtpmap:18 G729/8000
                    a=fmtp:18 annexb=no
                    a=rtpmap:111 G726-32/8000
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-16
                    a=silenceSupp:off - - - -
                    a=ptime:20
                    a=sendrecv

A Maintenance contract is essential, not a Luxury.
Do things on the cheap & it will cost you dear

 

[Westi]

usually the problem is the Firewall settings in the network topology which determines the NAT encapsulation.
You can use STUN to determine what setting you need to use and then turn STUN back off.

Joe W.

TeleTechs.ca
FHandw, ACSS (SME), ACIS (SME)


“This is the end of the world, make sure to buy your T-shirt before it is too late"
Original expression of my daughter

 

[IPGuru]

stun set corrctly
port forwarding ok
(even tried with IPO in DMZ)
I suspect SIP the sip provider but need to prove the IPO & this provider are incompatible.

A Maintenance contract is essential, not a Luxury.
Do things on the cheap & it will cost you dear

 

[intrigrant]

In the trace I see only a RX message which tell as much as the SIP is coming in from the provider, if you can post a longer trace either here or in the other area then that could reveil some more details. Preferrebly a trace with default settings plus SIP as a filter.

 

[IPGuru]

the reason there is only an RX message is because the IPO is not replying at all
here is a default trace with all SIP settings ticked just in case

18:21:56    1749563mS SIP Rx: UDP 87.127.240.98:5060 -> 192.168.99.200:5060
                    INVITE sip:44XXXXXXXXXX@yyy.yyy.yyy.yyy SIP/2.0
                    Via: SIP/2.0/UDP 87.127.240.98:5060;branch=z9hG4bK60aaaf24;rport
                    Max-Forwards: 70
                    From: "07XXXXXXXXX" <sip:07XXXXXXXXX@87.127.240.98>;tag=as59250b93
                    To: <sip:44XXXXXXXXXX@yyy.yyy.yyy.yyy>
                    Contact: <sip:07XXXXXXXXX@87.127.240.98:5060>
                    Call-ID: 015a9db31354eaf26f73259b09cd5f93@87.127.240.98:5060
                    CSeq: 102 INVITE
                    User-Agent: Entanet Media Server
                    Date: Mon, 17 Feb 2014 18:21:57 GMT
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                    Supported: replaces
                    X-DNQ: 44XXXXXXXXXX
                    Content-Type: application/sdp
                    Content-Length: 389
                    
                    v=0
                    o=root 322836865 322836865 IN IP4 87.127.240.98
                    s=Asterisk PBX 1.8.7.1
                    c=IN IP4 87.127.240.98
                    t=0 0
                    m=audio 18918 RTP/AVP 0 9 8 18 111 101
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:9 G722/8000
                    a=rtpmap:8 PCMA/8000
                    a=rtpmap:18 G729/8000
                    a=fmtp:18 annexb=no
                    a=rtpmap:111 G726-32/8000
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-16
                    a=silenceSupp:off - - - -
                    a=ptime:20
                    a=sendrecv
 18:21:56    1749567mS SIP Call Rx: phone
                    INVITE sip:44XXXXXXXXXX@yyy.yyy.yyy.yyy SIP/2.0
                    Via: SIP/2.0/UDP 87.127.240.98:5060;branch=z9hG4bK60aaaf24;rport
                    Max-Forwards: 70
                    From: "07XXXXXXXXX" <sip:07XXXXXXXXX@87.127.240.98>;tag=as59250b93
                    To: <sip:44XXXXXXXXXX@yyy.yyy.yyy.yyy>
                    Contact: <sip:07XXXXXXXXX@87.127.240.98:5060>
                    Call-ID: 015a9db31354eaf26f73259b09cd5f93@87.127.240.98:5060
                    CSeq: 102 INVITE
                    User-Agent: Entanet Media Server
                    Date: Mon, 17 Feb 2014 18:21:57 GMT
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                    Supported: replaces
                    X-DNQ: 44XXXXXXXXXX
                    Content-Type: application/sdp
                    Content-Length: 389
                    
                    v=0
                    o=root 322836865 322836865 IN IP4 87.127.240.98
                    s=Asterisk PBX 1.8.7.1
                    c=IN IP4 87.127.240.98
                    t=0 0
                    m=audio 18918 RTP/AVP 0 9 8 18 111 101
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:9 G722/8000
                    a=rtpmap:8 PCMA/8000
                    a=rtpmap:18 G729/8000
                    a=fmtp:18 annexb=no
                    a=rtpmap:111 G726-32/8000
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-16
                    a=silenceSupp:off - - - -
                    a=ptime:20
                    a=sendrecv
 18:21:57    1750051mS SIP Rx: UDP 87.127.240.98:5060 -> 192.168.99.200:5060
                    INVITE sip:44XXXXXXXXXX@yyy.yyy.yyy.yyy SIP/2.0
                    Via: SIP/2.0/UDP 87.127.240.98:5060;branch=z9hG4bK60aaaf24;rport
                    Max-Forwards: 70
                    From: "07XXXXXXXXX" <sip:07XXXXXXXXX@87.127.240.98>;tag=as59250b93
                    To: <sip:44XXXXXXXXXX@yyy.yyy.yyy.yyy>
                    Contact: <sip:07XXXXXXXXX@87.127.240.98:5060>
                    Call-ID: 015a9db31354eaf26f73259b09cd5f93@87.127.240.98:5060
                    CSeq: 102 INVITE
                    User-Agent: Entanet Media Server
                    Date: Mon, 17 Feb 2014 18:21:57 GMT
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                    Supported: replaces
                    X-DNQ: 44XXXXXXXXXX
                    Content-Type: application/sdp
                    Content-Length: 389
                    
                    v=0
                    o=root 322836865 322836865 IN IP4 87.127.240.98
                    s=Asterisk PBX 1.8.7.1
                    c=IN IP4 87.127.240.98
                    t=0 0
                    m=audio 18918 RTP/AVP 0 9 8 18 111 101
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:9 G722/8000
                    a=rtpmap:8 PCMA/8000
                    a=rtpmap:18 G729/8000
                    a=fmtp:18 annexb=no
                    a=rtpmap:111 G726-32/8000
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-16
                    a=silenceSupp:off - - - -
                    a=ptime:20
                    a=sendrecv
 18:21:57    1750054mS SIP Call Rx: phone
                    INVITE sip:44XXXXXXXXXX@yyy.yyy.yyy.yyy SIP/2.0
                    Via: SIP/2.0/UDP 87.127.240.98:5060;branch=z9hG4bK60aaaf24;rport
                    Max-Forwards: 70
                    From: "07XXXXXXXXX" <sip:07XXXXXXXXX@87.127.240.98>;tag=as59250b93
                    To: <sip:44XXXXXXXXXX@yyy.yyy.yyy.yyy>
                    Contact: <sip:07XXXXXXXXX@87.127.240.98:5060>
                    Call-ID: 015a9db31354eaf26f73259b09cd5f93@87.127.240.98:5060
                    CSeq: 102 INVITE
                    User-Agent: Entanet Media Server
                    Date: Mon, 17 Feb 2014 18:21:57 GMT
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                    Supported: replaces
                    X-DNQ: 44XXXXXXXXXX
                    Content-Type: application/sdp
                    Content-Length: 389
                    
                    v=0
                    o=root 322836865 322836865 IN IP4 87.127.240.98
                    s=Asterisk PBX 1.8.7.1
                    c=IN IP4 87.127.240.98
                    t=0 0
                    m=audio 18918 RTP/AVP 0 9 8 18 111 101
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:9 G722/8000
                    a=rtpmap:8 PCMA/8000
                    a=rtpmap:18 G729/8000
                    a=fmtp:18 annexb=no
                    a=rtpmap:111 G726-32/8000
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-16
                    a=silenceSupp:off - - - -
                    a=ptime:20
                    a=sendrecv

there are a few items that spike my curiosity but I wont say which as I don't want to predispose you to any particular answer.


A Maintenance contract is essential, not a Luxury.
Do things on the cheap & it will cost you dear

 

[mforrence]

I had a similar situation recently - I deleted the SIP trunk, sent config, made new SIP trunk, sent config and IPO began responding to invites again. I chalked it up to IPO not liking me changing addresses, etc.

 

[IPGuru]

Possible although at present I am testing @ home with test kit so it has been deleted & added a few times now

A Maintenance contract is essential, not a Luxury.
Do things on the cheap & it will cost you dear

 

[intrigrant]

I suppose it is this : 18:21:56 1749567mS SIP Call Rx: phone
It shows as it try to call in as a phone instead as a Trunk.
Tell them to buy Sippy or something like that, it is a lot more proffessional and reliable, asterisks is nice for playing with a SIP softswitch but definetely NOT for a serious provider.

 

[IPGuru]

Exactly my thoughts so far.
I will be checking traces on working systems tomorrow to confirm

I don't think Asterix is the problem (I am sure other providers are using it) but this provider does things strange, it is the same one I mentioned elsewhere that is sending the DDI as a X-dnq message on a circuit using registration, this is there suggested "Fix".


A Maintenance contract is essential, not a Luxury.
Do things on the cheap & it will cost you dear

 

[redphone]

Verify the LAN VoIP settings has SIP Trunks Enable checked. This can be overlooked and cause IP Office to ignore the INVITE.

 

[intrigrant]

Hey, it is IPGuru, he would never make such a beginners error.
Still, Asterisk is not really PRO in my opinion....

 

[IPGuru]

Thanks for the vote of Confidence, You would be amazed about how many "Beginners errors" I can make, Just I usually spot them before coming here (Asking for help instead of giving it is embarrassing )

I also dont reject any suggestions no matter how obvious, those are the ones that usually get missed.

Sip Trunks enabled
Firewall None
topology (using stun)
Voip All enabled (yes I know about the security but this is in a sand-pit environment)
Line deleted, system Reboot, recreated No Change

I am fairly convinced that Entanet & IP Office cannot be configured to work correctly together, unless anyone else knows better?




A Maintenance contract is essential, not a Luxury.
Do things on the cheap & it will cost you dear

 

[IPGuru]

Finally seem to have resolved the problems

Association method needs to be "Via" Header Hostpart...

The provider also has a number of IP addresses used as the ITSP Proxy (I have seen found 3 so far). these need to be entered in the ITSP Proxy Field separated by spaces.
entering the url (proxy.entacall.com) does not appear to work.



A Maintenance contract is essential, not a Luxury.
Do things on the cheap & it will cost you dear

 

[tlpeter]

The URL does not work when you do not have entered a working DNS server in the IPO or in the SIP settings.
Can you ping it and resolve it?

BAZINGA!

I'm not insane, my mother had me tested!

 

[IPGuru]

DNS servers have been set in the IPO Peter,
Pinging the url gives yet another IP Address that I have not yet seen as the origin of a call.

the ITSP now tells me that have a lot of addresses that could potentially be used.

One door opens another one closes ...


A Maintenance contract is essential, not a Luxury.
Do things on the cheap & it will cost you dear

 

[StubbedmytoeonIPO]

Put username and password in but don't check registration both can be garbage.