Entacall Easy Access SIP Trunk DiD Issues

[rsmith]

Another weird one, I have just started testing Entacall/Entanet basic SIP trunk service and I'm having major issues getting DDIs being recognised.

I haven't had this problem with any other provider, but basically the SIP Header contains the DDI number that has been dialled and even though I have a route set up for it, asterisk ignores it and goes to the default route.

I have copied the main content from the SIP header

Message Header
    Record-Route: <sip:87.127.240.101;lr;ftag=as70f6c22d>
    Via: SIP/2.0/UDP 87.127.240.101;branch=z9hG4bK98f8.7fb25d16.0
        Transport: UDP
        Sent-by Address: 87.127.240.101
        Branch: z9hG4bK98f8.7fb25d16.0
    Via: SIP/2.0/UDP 87.127.215.242:5060;branch=z9hG4bK3b8a8a3a;rport=5060
        Transport: UDP
        Sent-by Address: 87.127.215.242
        Sent-by port: 5060
        Branch: z9hG4bK3b8a8a3a
        RPort: 5060
    Max-Forwards: 69
    From: "07557********" <sip:07557*******@proxy.entacall.com>;tag=as70f6c22d
        SIP Display info: "07557********"
        SIP from address: sip:07557*******@proxy.entacall.com
        SIP from tag: as70f6c22d
    To: <sip:441308********@proxy.entacall.com>
    Contact: <sip:07557*******@87.127.215.242:5060>
    Call-ID: 3d588bd8288eeb3c6520190d212e3303@proxy.entacall.com
    CSeq: 102 INVITE
    User-Agent: Entanet Media Server
    Date: Tue, 05 Jul 2016 15:45:36 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces
    X-DNQ: 441308********
    Content-Type: application/sdp
    Content-Length: 393

Any thoughts would be appreciated?

Thanks

Richard

[sysadmin]

Run the call with sip debugging enabled on the Asterisk console and post the console output here.

[rsmith]

Below is the output of the SIP debug. So I called from my mobile to DDI ending in 34 but it seems to ignore the route for 34 and goes to the default route.

Any thoughts?

Thanks

Richard

<--- SIP read from UDP:87.127.240.101:5060 --->
ACK sip:441305*****3@52.31.56.45:5060 SIP/2.0
Via: SIP/2.0/UDP 87.127.240.101;branch=z9hG4bKebf1.de7796a4.0
From: "0**********9" <sip:0********9@proxy.entacall.com>;tag=as766a7e24
Call-ID: 5bf7d9021ff873e6505eba747f290c17@proxy.entacall.com
To: <sip:4413052******4@proxy.entacall.com>;tag=as74a912bd
CSeq: 102 ACK
User-Agent: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '5bf7d9021ff873e6505eba747f290c17@proxy.entacall.com' Method: ACK

<--- SIP read from UDP:enduserip:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.31.56.45:5060;branch=z9hG4bK693afe5e;rport=5060
From: <sip:405@mysipserver:5060>;tag=as73a526d3
To: "Emma" <sip:404@mysipserver:5060>;tag=3574087852
Call-ID: 0_3898411430@192.168.1.102
CSeq: 268 NOTIFY
User-Agent: Yealink SIP-T42G 29.80.23.1
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:phoneipaddress:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.31.56.45:5060;branch=z9hG4bK5dcef61e;rport=5060
From: "0*********9" <sip:0**********9@52.31.56.45>;tag=as02192c47
To: <sip:405@enduserip:5060>;tag=2210439667
Call-ID: 6da5ff4510b6293b293bad4c5d6dbeb5@52.31.56.45:5060
CSeq: 102 CANCEL
User-Agent: Yealink SIP-T42G 29.80.23.1
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:enduserip:5060 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 52.31.56.45:5060;branch=z9hG4bK5dcef61e;rport=5060
From: "0*********9" <sip:0*********9@52.31.56.45>;tag=as02192c47
To: <sip:405@enduserip:5060>;tag=2210439667
Call-ID: 6da5ff4510b6293b293bad4c5d6dbeb5@52.31.56.45:5060
CSeq: 102 INVITE
User-Agent: Yealink SIP-T42G 29.80.23.1
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:405@enduserip:5060> for address/port to send to
set_destination: set destination to enduserip:5060
Transmitting (NAT) to enduserip:5060:
ACK sip:405@enduserip:5060 SIP/2.0
Via: SIP/2.0/UDP 52.31.56.45:5060;branch=z9hG4bK5dcef61e;rport
Max-Forwards: 70
From: "0**********9" <sip:0**********9@52.31.56.45>;tag=as02192c47
To: <sip:405@enduserip:5060>;tag=2210439667
Contact: <sip:0*********9@52.31.56.45:5060>
Call-ID: 6da5ff4510b6293b293bad4c5d6dbeb5@52.31.56.45:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Content-Length: 0


---
Scheduling destruction of SIP dialog '6da5ff4510b6293b293bad4c5d6dbeb5@52.31.56.45:5060' in 6400 ms (Method: INVITE)

[sysadmin]

What is the default route set to?

[rsmith]

I actually set the default route to hang up, so it ignores the default route and only sees it based on the main inbound number for the trunk not the ddi that has been called in.

[sysadmin]

I'm confused.  You said it was going to the default route but the default route is set to hangup.   Is this correct?

You should remove the default route.  It is only there as a start point, or catchall, it isn't intended for production.    Next you should create a DDI to catch the inbound TO from SIP.   



   

[rsmith]

Ok, I have managed to work this out, after a lot of work! Basically Enta put the DDI number in a custom header called X-DNQ (don't ask me why and the current tech manager doesn't know either) but it is.

So I  was able to add the following to sark_customer_extensions_globals.conf:

[from-custom]
exten => _.,1,Goto(mainmenu,${SIP_HEADER(X-DNQ)},1)

In case anyone happens to come across the same issue!

[sysadmin]

Yes, I can see the custom header in the SIP trace... but the TO header also looks to be good?