DIDLogic SIP trunk not working (although trunk is responding with RX/TX both SIP/2.0 200 OK)

[g18c]

I have logged in to DID logic SIP accounts, and created my new line, set the password and restricted to my WAN IP.

With STUN has automatically detected FULL CONE NAT. All outbound packets are allowed on the firewall, and I am natting 5060 UDP to my IP Office for inbound packets to the WAN.

I have installed SIP trunk licenses, and made a new SIP line:

SIP Line -> ITSP Domain Name: sip.uk.didlogic.net
Transport -> ITSP Proxy Address: sip.uk.didlogic.net
Transport -> Use Network Topology Info = LAN1
Transport -> Explicit DNS Servers = 0.0.0.0
Transport -> Call Route via Registrar = Yes
SIP URI (channel 1) -> Local URI = Use Credentials User Name
SIP URI (channel 1) -> Contact = Use Credentials User Name
SIP URI (channel 1) -> Display Name = Use Credentials User Name
SIP URI (channel 1) -> PAI = None
SIP URI (channel 1) -> Outgoing Group = 2
SIP URI (channel 1) -> Max calls per channel = 5
SIP URI (channel 1) -> Registration = (entered under sip credentials)
VoIP -> All settings default
SIP Credentials -> Username = DID Logic Account Number
SIP Credentials -> Authentication Name = DID Logic Account Number
SIP Credentials -> Password = Alphanumeric Password the same as set to the trunk under DID logic control panel
SIP Credentials -> Registration required = yes

When I try and dial I get a solid dial tone and status is showing trunk out of service.

 16:07:54  854726332mS PRN: DNS - recommended address change: query in progress
 
 16:07:59  854551330mS SIP Tx: UDP 192.168.1.2:5060 -> 176.74.184.153:5060
                    OPTIONS sip:sip.uk.didlogic.net SIP/2.0
                    Via: SIP/2.0/UDP 94.229.1.5:5060;rport;branch=x1041aab484a63805b6efa1e47e9d9558a03f3dc
                    From: <sip:sip.uk.didlogic.net>;tag=dde7e8118df6d017
                    To: <sip:sip.uk.didlogic.net>
                    Call-ID: 3a269dab19a2d7dec7f8d0521acf1295
                    CSeq: 1466294847 OPTIONS
                    Contact: <sip:94.229.1.5:5060;transport=udp>
                    Max-Forwards: 70
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
                    Supported: timer
                    User-Agent: IP Office 9.0.4.0 build 965
                    Content-Length: 0
                    
 16:07:59  854551609mS SIP Rx: UDP 176.74.184.153:5060 -> 192.168.1.2:5060
                    SIP/2.0 200 OK
                    Via: SIP/2.0/UDP 94.229.1.5:5060;rport=5060;branch=x1041aab484a63805b6efa1e47e9d9558a03f3dc
                    From: <sip:sip.uk.didlogic.net>;tag=dde7e8118df6d017
                    To: <sip:sip.uk.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.bd7a
                    Call-ID: 3a269dab19a2d7dec7f8d0521acf1295
                    CSeq: 1466294847 OPTIONS
                    Content-Length: 0
					
When I try and dial the SIP outbound group (2) I get:

16:11:43  854775192mS CMTARGET:     Problem with Line Id: 17 - check_DNS 1 OperationalTest: 0, IP address: 176.74.184.153

Anything I have done obviously wrong? Any further traces or diagnostics to enable to work out what is going on?

 

[g18c]

Hi guys, been pulling my hair out on this one... any one kind enough to point me in the right direction of finding more info, any specific trace options to set or look out for any specific code in the logs?

I cant see anything other than 200 OK both for RX and TX packets, and only about trunk being down and not the reason why.

Any pointers much appreciated.

 

[Gunnaro]

Try setting the ITSP's to the public IP address of the provider and make an IP route to that. If you don't know it, try a ping from your computer to sip.uk.didlogic.net
When done, do a Ping from SSA to see if it takes.

Also check that the RTP ports are open in your firewall for the Provider IP.

Kind regards

Gunnar
______________________________________
Mille viae ducunt homines per saecula Romam

 

[g18c]

Many thanks for the tip, I changed the RTP ports to 54000 - 56048 (i noticed Manager runs on ports within the default RTP range so wasnt happy about putting that on the internet) and made sure NAT was OK.

I also changed the password to one that didn't contain special characters, and it now works for outbound

I cant dial in however, i have the incoming group setup, but the DID provider just hangs up immediately - any ideas on that one and what to check? Any settings on the SIP trunk to allow DID?

 

[Gunnaro]

Without seeing the full picture, it will be a lot of guessing.

Do you see anything at all in the trace?

Could be that the incoming number is different than what you have configured, or nothing at all.

How is your URI setup?

Kind regards

Gunnar
______________________________________
Mille viae ducunt homines per saecula Romam

 

[g18c]

There was an issue with the config on provider side, traffic is now hitting the PBX on inbound calls.

However it is showing 404 not found.

 05:43:20    1011160mS Sip: 17.1025.1 -1 SIPTrunk Endpoint(f4e99bb0) Present Call, no match (112035192500) from URI in To header or (112035192500) from request URI
 
 05:43:20    1011160mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 176.74.184.153:5060 to 176.74.184.153:5060
 
 05:43:20    1011161mS SIP Call Tx: 17
                    SIP/2.0 404 Not Found
                    Via: SIP/2.0/UDP 176.74.184.153;branch=z9hG4bK8911.e3ba9c92.0
                    Via: SIP/2.0/UDP 72.251.241.166;rport=5060;branch=z9hG4bK8911.0f3c1016.0
                    Via: SIP/2.0/UDP 95.154.254.93:5080;branch=z9hG4bK60c59df5;rport=5080
                    Record-Route: <sip:176.74.184.153;lr=on;ftag=as4688c221;vst=AAAAAAIHBAYAdQYLHAcHAG4LAB8cAw8YNjcuMTQ0>
                    Record-Route: <sip:72.251.241.166;lr=on;ftag=as4688c221;vsf=AAAAAAAAAAAAAAAAAAAAAAAOBwADAAUAAAEFAAgFDDUwODA->
                    From: 971501705209 <sip:971501705209@72.251.241.166>;tag=as4688c221
                    Call-ID: 79fa74625c438a59440d6385586b6797@95.154.254.93
                    CSeq: 102 INVITE
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
                    Supported: timer
                    Server: IP Office 9.0.4.0 build 965
                    Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
                    To: "112035192500" <sip:112035192500@94.29.67.100>;tag=b09f563dd5a4a825
                    Content-Length: 0

I have setup the incoming call route for trunk group 3 (for which the SIP trunk is set to), to go direct to my extension.

Getting very close!

I can see from searches amriddle01 has suggested others who get 404 errors need a SIP URI with a * for all 3 fields (Local URI, Contact and Display) in addition to the incoming call route.

Does this mean I need to make a second new SIP URI (with *) in addition to the existing SIP URI (with Use Credentials) below:

SIP Line -> ITSP Domain Name: sip.uk.didlogic.net
Transport -> ITSP Proxy Address: sip.uk.didlogic.net
Transport -> Use Network Topology Info = LAN1
Transport -> Explicit DNS Servers = 0.0.0.0
Transport -> Call Route via Registrar = Yes
SIP URI (channel 1) -> Local URI = Use Credentials User Name
SIP URI (channel 1) -> Contact = Use Credentials User Name
SIP URI (channel 1) -> Display Name = Use Credentials User Name
SIP URI (channel 1) -> PAI = None
SIP URI (channel 1) -> Outgoing Group = 2
SIP URI (channel 1) -> Max calls per channel = 5
SIP URI (channel 1) -> Registration = (entered under sip credentials)
VoIP -> All settings default
SIP Credentials -> Username = DID Logic Account Number
SIP Credentials -> Authentication Name = DID Logic Account Number
SIP Credentials -> Password = Alphanumeric Password the same as set to the trunk under DID logic control panel
SIP Credentials -> Registration required = yes

 

[g18c]

Or in other words, two URIs or just one URI?

 

[g18c]

OK so i have it working now but need to check everything is OK:

SIP URI (channel 1) -> Local URI = *
SIP URI (channel 1) -> Contact = *
SIP URI (channel 1) -> Display Name = *
SIP URI (channel 1) -> PAI = None
SIP URI (channel 1) -> Incoming Group = 2
SIP URI (channel 1) -> Outgoing Group = 20
SIP URI (channel 1) -> Max calls per channel = 5
SIP URI (channel 1) -> Registration = (entered under sip credentials)

SIP URI (channel 2) -> Local URI = Use Credentials User Name
SIP URI (channel 2) -> Contact = Use Credentials User Name
SIP URI (channel 2) -> Display Name = Use Credentials User Name
SIP URI (channel 2) -> PAI = None
SIP URI (channel 2) -> Incoming Group = 20
SIP URI (channel 2) -> Outgoing Group = 2
SIP URI (channel 2) -> Max calls per channel = 5
SIP URI (channel 2) -> Registration = (entered under sip credentials)

I also setup the incoming call route, for channel 2 with presented number to map to the extension.

It works OK, but should i be putting the above 2 channels in per trunk or can i just use 1 with * / * / * for local, contact and display?

 

[Gunnaro]

This is good, you dont need any ICR for the "channel 2", that's your outbound.

Well done, you made it!

PS: be careful posting public IP's, you have a firewall and I assume it has something else than default pwd? And you have not made any other port forwarding I hope...

Kind regards

Gunnar
______________________________________
Mille viae ducunt homines per saecula Romam

 

[g18c]

Thanks yes that is not my real IP and there is a firewall with only RTP packets forwarded.

I have the UK SIP trunk working, but the USA SIP number is throwing an error:

 14:44:47     201360mS Sip: 17.1032.1 -1 SIPTrunk Endpoint(f4e84240) SetRfc2833TxPayload: use RFC2833 for dtmf 
 14:44:47     201361mS Sip: 17.1032.1 -1 SIPTrunk Endpoint(f4e84240) SetRemoteRTPAddress to 72.251.228.147:17372 
 14:44:47     201363mS CMCallEvt:    CREATE CALL:14 (f57bc9a8)
 14:44:47     201363mS CMCallEvt:    0.1033.0 -1 BaseEP: NEW CMEndpoint f4e7bc48 TOTAL NOW=2 CALL_LIST=0
 14:44:47     201365mS CMLineRx: v=0
            CMSetup
            Line: type=SIPLine 17 Call: lid=17 id=1032 in=1
            Called[13452612101] Type=Default (100) Reason=CMDRdirect  SndComp Calling[94479735424211@72.251.228.147] Type=Unknown Plan=Default 
            BC: CMTC=Speech CMTM=Circuit CMTR=64 CMST=Default CMU1=ALaw
            IE CMIEFastStartInfoData (6) 6 item(s)
            IE CMIERespondingPartyNumber (230)(P:100 S:100 T:0 N:100 R:4) number=94479735424211@72.251.228.147
            IE CMIEDeviceDetail (231) LOCALE=aru HW=15 VER=9 class=CMDeviceSIPTrunk type=0 number=17 channel=1 features=0x0 rx_gain=32 tx_gain=32 ep_callid=1032 ipaddr=192.168.111.2 apps=0 loc=999 em_loc=999 features2=0x0
 14:44:47     201366mS CD: CALL: 17.1032.1 BState=Idle Cut=1 Music=0.0 Aend="Line 17" (0.0) Bend="" [] (0.0) CalledNum=13452612101 () CallingNum=94479735424211@72.251.228.147 () Internal=0 Time=2 AState=Idle
 14:44:47     201366mS CMCallEvt:    17.1032.1 14 SIPTrunk Endpoint: StateChange: END=A CMCSIdle->CMCSDialInitiated
 14:44:47     201366mS CMTARGET:     17.1032.1 14 SIPTrunk Endpoint: LOOKUP CALL ROUTE: type=100 called_party=13452612101 sub= calling=94479735424211@72.251.228.147 dir=in complete=1 ses=0
 14:44:47     201368mS CMLOGGING:     CALL:2015/01/0314:44,00:00:00,000,94479735424211@72.251.228.147,I,13452612101,13452612101,,,,0,,""n/a,0
 14:44:47     201368mS CD: CALL: 17.1032.1 BState=Idle Cut=0 Music=0.0 Aend="Line 17" (0.0) Bend="" [] (0.0) CalledNum=13452612101 () CallingNum=94479735424211@72.251.228.147 () Internal=0 Time=5 AState=Dialling
 14:44:47     201369mS CD: CALL: 17.1032.1 Deleted
 14:44:47     201369mS CMLineTx: v=0
            CMReleaseComp
            Line: type=SIPLine 17 Call: lid=17 id=1032 in=1
            Cause=1, Unallocated (unassigned) number
 14:44:47     201369mS Sip: 17.1032.1 -1 SIPTrunk Endpoint(f4e85840) received CMReleaseComp
 14:44:47     201369mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 176.74.184.153:5060 to 176.74.184.153:5060
 14:44:47     201370mS SIP Call Tx: 17
                    SIP/2.0 404 Not Found
                    Via: SIP/2.0/UDP 176.74.184.153;branch=z9hG4bK5aea.86e43696.0
                    Via: SIP/2.0/UDP 192.241.183.87;rport=5060;branch=z9hG4bK5aea.b868ab13.0
                    Via: SIP/2.0/UDP 72.251.228.147:5080;branch=z9hG4bK683c5f03;rport=5080
                    Record-Route: <sip:176.74.184.153;lr=on;ftag=as27b03c42;vst=AAAAAAcAAgEBdggLBx8DdAgaHwoBFxYBNy4xNDQ->
                    Record-Route: <sip:192.241.183.87;lr=on;ftag=as27b03c42;vsf=AAAAAAAAAAAAAAAAAAAAAAAGCxwcBwUfHAMAHR8MADo1MDgw>
                    From: 94479735424211 <sip:94479735424211@192.241.183.87>;tag=as27b03c42
                    Call-ID: 1119dfeb20d97ab9671423064e33d3de@72.251.228.147:5080
                    CSeq: 102 INVITE
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
                    Supported: timer
                    Server: IP Office 9.0.4.0 build 965
                    Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
                    To: "13452612101" <sip:13452612101@94.2.66.10>;tag=f1d59b64a8bcc683
                    Content-Length: 0

I have added additional URI channels to the SIP trunk, so the UK channels are on groups 1 and 10, and the US are on 2 and 20. The both go to the same SIP registrar.

The incoming call routes are set the same, everything is identical except the number on the incoming call route which is set as 13452612101. The UK number is the same and works fine, the US one doesn't.

Any idea why I am getting Reason: Q.850;cause=1;text="Unallocated (unassigned) number" please?

 

[Gunnaro]

You lost me there.
Do you have two trunks or one Trunk with a mix of US and UK DDI's?

Unallocated would normaly mean that the number is not in the ICR, so have you put I'm the correct Line Grp ID?

Kind regards

Gunnar
______________________________________
Mille viae ducunt homines per saecula Romam

 

[g18c]

OK fixed it fully Your comment helped me think everything over properly.

I noticed that the registration for the channel was being ignored for the inbound URI (i saw it flashing up on the wrong trunk as i had tried first by separate trunks but for some reason the calls were always sent to the first trunk so it was going to the wrong incoming group - maybe didlogic don't like multiple registrations?). I removed all separate incoming groups, and just made a shared one. So i setup as follows and now works perfectly

SIP URI (channel 1) -> Local URI = Use Credentials User Name
SIP URI (channel 1) -> Contact = Use Credentials User Name
SIP URI (channel 1) -> Display Name = Use Credentials User Name
SIP URI (channel 1) -> PAI = None
SIP URI (channel 1) -> Incoming Group = 20
SIP URI (channel 1) -> Outgoing Group = 2
SIP URI (channel 1) -> Max calls per channel = 5
SIP URI (channel 1) -> Registration = (entered under sip credentials UK SIP account)

SIP URI (channel 2) -> Local URI = Use Credentials User Name
SIP URI (channel 2) -> Contact = Use Credentials User Name
SIP URI (channel 2) -> Display Name = Use Credentials User Name
SIP URI (channel 2) -> PAI = None
SIP URI (channel 2) -> Incoming Group = 20
SIP URI (channel 2) -> Outgoing Group = 2
SIP URI (channel 2) -> Max calls per channel = 5
SIP URI (channel 2) -> Registration = (entered under sip credentials US SIP account)

SIP URI (channel 3) -> Local URI = *
SIP URI (channel 3) -> Contact = *
SIP URI (channel 3) -> Display Name = *
SIP URI (channel 3) -> PAI = None
SIP URI (channel 3) -> Incoming Group = 100
SIP URI (channel 3) -> Outgoing Group = 100
SIP URI (channel 3) -> Max calls per channel = 5
SIP URI (channel 3) -> Registration = None

I then create the DIDs, under Incoming Call Route, set the incoming group 100, followed by the DID number, then extension destination.

Please let me know if anything is not to best practice so i can learn from this

 

[g18c]

Sorry copy and paste error, here is what i did:

SIP URI (channel 1) -> Local URI = Use Credentials User Name
SIP URI (channel 1) -> Contact = Use Credentials User Name
SIP URI (channel 1) -> Display Name = Use Credentials User Name
SIP URI (channel 1) -> PAI = None
SIP URI (channel 1) -> Incoming Group = 1
SIP URI (channel 1) -> Outgoing Group = 1
SIP URI (channel 1) -> Max calls per channel = 5
SIP URI (channel 1) -> Registration = (entered under sip credentials UK SIP account)

SIP URI (channel 2) -> Local URI = Use Credentials User Name
SIP URI (channel 2) -> Contact = Use Credentials User Name
SIP URI (channel 2) -> Display Name = Use Credentials User Name
SIP URI (channel 2) -> PAI = None
SIP URI (channel 2) -> Incoming Group = 2
SIP URI (channel 2) -> Outgoing Group = 2
SIP URI (channel 2) -> Max calls per channel = 5
SIP URI (channel 2) -> Registration = (entered under sip credentials US SIP account)

SIP URI (channel 3) -> Local URI = *
SIP URI (channel 3) -> Contact = *
SIP URI (channel 3) -> Display Name = *
SIP URI (channel 3) -> PAI = None
SIP URI (channel 3) -> Incoming Group = 100
SIP URI (channel 3) -> Outgoing Group = 100
SIP URI (channel 3) -> Max calls per channel = 5
SIP URI (channel 3) -> Registration = None

 

[Gunnaro]

If this works as intended, it should be OK
(When you write Channel 1, 2, 3, do you mean different SIP Trunks (lines) or have you put every URI in the same SIP Trunk?

BTW: You wrote that you Port forwarded the RTP.
A good SIP trunks does not need port forwarding, it should be NAT'ed.

When I wrote Open RTP, I meant "If RTP does not reach the IPO, make a rule/service that allow those port to pass on to the destination (NAT), but only from your SIP Providers IP address".

Kind regards

Gunnar
______________________________________
Mille viae ducunt homines per saecula Romam

 

[g18c]

It was my first time to setup so a little knowledge is dangerous

Channel 1 to 3 are all within the same SIP trunk. Maybe channel is not the right word, but i only have 1 SIP trunk.

With multiple SIP trunks, seemed it was still matching on a different trunk?! I think its to do with the way IP Office is matching the URI and there was an ambiguity.

I have used NAT masquerade behind a firewall and STUN in IP Office (Full Cone Nat), when i saw port forward, any RTP or signalling packets hitting the external IP are forwarded through.

Signalling is setup so that incoming packets map through to IP Office from the providers SBC.

RTP comes from any address as the provider has multiple RTP streams and these change rapidly... so as of now RTP inbound is mapped also (within the range they mentioned) - not sure if this is something to be concerned about (since signalling is locked by IP).

I could spend some extra time and work out the firewall 1:1 NAT, but it is working, in fact it is working very well didlogic quality is actually excellent.

 

[Gunnaro]

The thing I'm worried about is that (probably) within your RTP-range, you find the ports for managing/sniffing/controlling the IPO, (5077x-5081x'ish). Some of these are UDP, others TCP.

On the IPO you can move the RTP range out of that area by reducing the span.
Provider should have given you an idea of what they expect you to "broadcast" on, typically between 40000 and 65000.

My providers send traffic on only a single IP, not a random range.
Sounds strange to me, but maybe I'm just a spoilt tech

Kind regards

Gunnar
______________________________________
Mille viae ducunt homines per saecula Romam

 

[g18c]

Already checked the service ports, by default they indeed are within the normal RTP ranges, this provider sends RTP streams on a much smaller range. on which the admin ports are not running.

Appreciate all the help and bouncing ideas of.