Incoming calls from Analog Trunk Issue

[musle1]

Hello,

Thanks for taking the time to read my issue with incoming calls, the story like this:

We have 2 PBX connected together through SIP Trunk , first Avaya IP Office 500 V2, second Elastix PBX, i can make calls from any PBX to another without any problems.
4 Analog trunks lines (gateway devices support GSM SIM ) are connected to Elastix ports and i can make In and Out calls .

What i need is to transfer all coming calls from Elastix to reach Avaya PBX then transfer the calls to any destination needed. I've made all necessary configuration in both PBX, but the number still ringing without entering to avaya.

What i configured like this :
1- All incoming calls to Elastix transferring to the Sip trunk that reach Avaya IP Office .
2- Configured incoming route in IP office with correct Line group ID , Incoming number : Plank ( also i've tried * ) , Incoming CLI: Plank , destination to our IVR.
3- Ensured that the calling is reached IP office by Monitor screen.

Some log from IP Office Monitor when i'm trying to call :

13:00:47 69093104mS SIP Rx: UDP 192.168.9.192:5060 -> 192.168.9.220:5060
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.192:5060;branch=z9hG4bK67018cf9;rport
Max-Forwards: 70
From: "10" <sip:[email protected]>;tag=as70b92207
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:506 0
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(11.20.0)
Date: Wed, 07 Dec 2016 11:01:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 1186632178 1186632178 IN IP4 192.168.9.192
s=Asterisk PBX 11.20.0
c=IN IP4 192.168.9.192
t=0 0
m=audio 13368 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
13:00:47 69093108mS CMCallEvt: 0000000000000000 0.3522.0 -1 BaseEP: NEW CMEndpoint f17c0998 TOTAL NOW=1 CALL_LIST=0
13:00:47 69093109mS Stun: Info: Line 17: Not using STUN for media in this case.
13:00:47 69093111mS SIP Tx: UDP 192.168.9.220:5060 -> 192.168.9.192:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.9.192:5060;branch=z9hG4bK67018cf9;rport
From: "10" <sip:[email protected]>;tag=as70b92207
Call-ID: [email protected]:506 0
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UP DATE
Supported: timer
Server: IP Office 9.1.2.0 build 91
To: <sip:[email protected]:5060>;tag=dac97c119882a0a3
Content-Length: 0

13:00:47 69093112mS SIP Tx: UDP 192.168.9.220:5060 -> 192.168.9.192:5060
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.9.192:5060;branch=z9hG4bK67018cf9;rport
From: "10" <sip:[email protected]>;tag=as70b92207
Call-ID: [email protected]:506 0
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UP DATE
Supported: timer
Server: IP Office 9.1.2.0 build 91
Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
To: <sip:[email protected]:5060>;tag=dac97c119882a0a3
Content-Length: 0

13:00:47 69093115mS SIP Rx: UDP 192.168.9.192:5060 -> 192.168.9.220:5060
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.192:5060;branch=z9hG4bK67018cf9;rport
Max-Forwards: 70
From: "10" <sip:[email protected]>;tag=as70b92207
To: <sip:[email protected]:5060>;tag=dac97c119882a0a3
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:506 0
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(11.20.0)
Content-Length: 0

Please note that the Elasix PBX has IP : 192.168.9.192
IP Office IP : 192.168.9.220

Thank all for support .

[morri234]

I see you are using port 5060. This seems to be where something goes wrong. Suggest you review what ports are actually used for this piece of your setup and if each port in the process is configured properly.