SIP Line IPOffice 500

[HBPSYS]

I have purchased a SIP Line for my IP office 500 and have setup up a SIP Trunk. The SIP has registered with the provisor as I can see this in system Status and I can cam make a call out through the SIP Trunk. When I try to make call in to the registered No. which is a london number I just get an engaged tone. I have setup an incoming call router which just has the line group ID that refers to my SIP URI.

Has anyone get any ideas that could help me as I'm totally stuck. Many Thanks.

Also I have enable UDP Port 5060 straight through my firewall to the LAN 1 IP on the Avaya IP Office

 

[Bas1234]

post a monitor trace here.

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

 

[HBPSYS]

Ok thanks what filters should I tick on the monitor as I dont use it often

 

[Bas1234]

Only the SIP part

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

 

[HBPSYS]

Hi Many Thanks I enabled the sip logs it shows in the monitor trace when I make a outgoing call. When I make a call in there is nothing in the monitor at all.

Here is some of the monitor log
a=sendrecv
7722330mS SIP Trunk: 17:Rx
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.1.8:5060;branch=z9hG4bK29c8209060d80fb91b32c9971ef368df;received=88.96.247.242;rport=10357
From: "84415672" <sip:84415672@sip.voiceflex.com >;tag=70a7f499f20d0f2e
To: <sip:07767491241@sip.voiceflex.com>;tag=as462fcd76
Call-ID: 0a7a5a74bb6e2ed05383d0ec57948b7c@10.0.1.8
CSeq: 1579066102 INVITE
User-Agent: VoiceFlex
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:07767491241@146.101.248.200>
Content-Type: application/sdp
Content-Length: 317

v=0
o=root 14313 14313 IN IP4 146.101.248.200
s=session
c=IN IP4 146.101.248.200
t=0 0
m=audio 29944 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSuppff - - - -
a=ptime:20
a=sendrecv
7722333mS SipDebugInfo: SIPDialog TXN : Decoding of message Succeded 1
7722333mS SipDebugInfo: Find End Point 17.3131.0 382 SIPTrunk Endpoint (f53fea3c) Sip CallId 0a7a5a74bb6e2ed05383d0ec57948b7c@10.0.1.8
7722333mS SipDebugInfo: 17.3131.0 382 SIPTrunk Endpoint(f53fde18) Process SIP response dialog f53fde18, method INVITE, CodeNum 183 in state SIPDialog::INV_PROV_RESP_RCVD(0)
7722334mS SipDebugInfo: 17.3131.0 382 SIPTrunk Endpoint(f53fde18) UpdateSIPCallState SIPDialog::INV_PROV_RESP_RCVD(5) -> SIPDialog::SESSION_IN_PROGRESS(41)
7722334mS SipDebugInfo: 17.3131.0 382 SIPTrunk Endpoint(f53fde18) UpdateSDPState SIPDialog::OFFER_SENT(1) -> SIPDialog::ANSWER_RCVD(4)
7722335mS SipDebugInfo: 17.3131.0 382 SIPTrunk Endpoint(f53fde18) SendCMMessageFromSDP: 9 attribute fields
7722341mS SipDebugInfo: 17.3131.0 382 SIPTrunk Endpoint(f53fde18) UpdateSDPState SIPDialog::ANSWER_RCVD(4) -> SIPDialog::COMPLETE(7)
7726075mS SipDebugInfo: Timer 4 callback
7727557mS SipDebugInfo: 17.3131.0 -1 SIPTrunk Endpoint(f53fde18) Terminating dialog f53fde18, state SIPDialog::SESSION_IN_PROGRESS(41) for cause 16
7727558mS SipDebugInfo: 17.3131.0 -1 SIPTrunk Endpoint(f53fde18) SendSIPRequest: CANCEL SENT TO 146.101.248.200 5060
7727558mS SIP Trunk: 17:Tx
CANCEL sip:07767491241@sip.voiceflex.com SIP/2.0
Via: SIP/2.0/UDP 10.0.1.8:5060;rport;branch=z9hG4bK29c8209060d80fb91b32c9971ef368df
From: "84415672" <sip:84415672@sip.voiceflex.com >;tag=70a7f499f20d0f2e
To: <sip:07767491241@sip.voiceflex.com>
Call-ID: 0a7a5a74bb6e2ed05383d0ec57948b7c@10.0.1.8
CSeq: 1579066102 CANCEL
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

7727559mS SIP Tx: UDP 10.0.1.8:5060 -> 146.101.248.200:5060
CANCEL sip:07767491241@sip.voiceflex.com SIP/2.0
Via: SIP/2.0/UDP 10.0.1.8:5060;rport;branch=z9hG4bK29c8209060d80fb91b32c9971ef368df
From: "84415672" <sip:84415672@sip.voiceflex.com >;tag=70a7f499f20d0f2e
To: <sip:07767491241@sip.voiceflex.com>
Call-ID: 0a7a5a74bb6e2ed05383d0ec57948b7c@10.0.1.8
CSeq: 1579066102 CANCEL
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

7727559mS SipDebugInfo: 17.3131.0 -1 SIPTrunk Endpoint(f53fde18) UpdateSIPCallState SIPDialog::SESSION_IN_PROGRESS(41) -> SIPDialog::CANCEL_SENT(6)
7727606mS SIP Rx: UDP 146.101.248.200:5060 -> 10.0.1.8:5060
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.0.1.8:5060;branch=z9hG4bK29c8209060d80fb91b32c9971ef368df;received=88.96.247.242;rport=10357
From: "84415672" <sip:84415672@sip.voiceflex.com >;tag=70a7f499f20d0f2e
To: <sip:07767491241@sip.voiceflex.com>;tag=as462fcd76
Call-ID: 0a7a5a74bb6e2ed05383d0ec57948b7c@10.0.1.8
CSeq: 1579066102 INVITE
User-Agent: VoiceFlex
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

7727606mS SIP Trunk: 17:Rx
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.0.1.8:5060;branch=z9hG4bK29c8209060d80fb91b32c9971ef368df;received=88.96.247.242;rport=10357
From: "84415672" <sip:84415672@sip.voiceflex.com >;tag=70a7f499f20d0f2e
To: <sip:07767491241@sip.voiceflex.com>;tag=as462fcd76
Call-ID: 0a7a5a74bb6e2ed05383d0ec57948b7c@10.0.1.8
CSeq: 1579066102 INVITE
User-Agent: VoiceFlex
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

7727608mS SipDebugInfo: SIPDialog TXN : Decoding of message Succeded 1
7727608mS SipDebugInfo: Find End Point 17.3131.0 -1 SIPTrunk Endpoint (f53fea3c) Sip CallId 0a7a5a74bb6e2ed05383d0ec57948b7c@10.0.1.8
7727608mS SipDebugInfo: 17.3131.0 -1 SIPTrunk Endpoint(f53fde18) Process SIP response dialog f53fde18, method INVITE, CodeNum 487 in state SIPDialog::CANCEL_SENT(0)
7727609mS SipDebugInfo: 17.3131.0 -1 SIPTrunk Endpoint(f53fde18) ExtractContactFromMessage: cannot get Contact Header 2012
7727609mS SipDebugInfo: 17.3131.0 -1 SIPTrunk Endpoint(f53fde18) SendSIPRequest: ACK SENT TO 146.101.248.200 5060
7727610mS SIP Trunk: 17:Tx
ACK sip:07767491241@sip.voiceflex.com SIP/2.0
Via: SIP/2.0/UDP 10.0.1.8:5060;rport;branch=z9hG4bK29c8209060d80fb91b32c9971ef368df
From: "84415672" <sip:84415672@sip.voiceflex.com >;tag=70a7f499f20d0f2e
To: <sip:07767491241@sip.voiceflex.com>;tag=as462fcd76
Call-ID: 0a7a5a74bb6e2ed05383d0ec57948b7c@10.0.1.8
CSeq: 1579066102 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

7727610mS SIP Tx: UDP 10.0.1.8:5060 -> 146.101.248.200:5060
ACK sip:07767491241@sip.voiceflex.com SIP/2.0
Via: SIP/2.0/UDP 10.0.1.8:5060;rport;branch=z9hG4bK29c8209060d80fb91b32c9971ef368df
From: "84415672" <sip:84415672@sip.voiceflex.com >;tag=70a7f499f20d0f2e
To: <sip:07767491241@sip.voiceflex.com>;tag=as462fcd76
Call-ID: 0a7a5a74bb6e2ed05383d0ec57948b7c@10.0.1.8
CSeq: 1579066102 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

7727611mS SipDebugInfo: 17.3131.0 -1 SIPTrunk Endpoint(f53fde18) UpdateSIPCallState SIPDialog::CANCEL_SENT(6) -> SIPDialog::FINAL(40)
7727611mS SipDebugInfo: 17.3131.0 -1 SIPTrunk Endpoint(f53fde18) #1 0
7727612mS SIP Rx: UDP 146.101.248.200:5060 -> 10.0.1.8:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.8:5060;branch=z9hG4bK29c8209060d80fb91b32c9971ef368df;received=88.96.247.242;rport=10357
From: "84415672" <sip:84415672@sip.voiceflex.com >;tag=70a7f499f20d0f2e
To: <sip:07767491241@sip.voiceflex.com>;tag=as462fcd76
Call-ID: 0a7a5a74bb6e2ed05383d0ec57948b7c@10.0.1.8
CSeq: 1579066102 CANCEL
User-Agent: VoiceFlex
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:07767491241@146.101.248.200>
Content-Length: 0

7727613mS SIP Trunk: 17:Rx
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.8:5060;branch=z9hG4bK29c8209060d80fb91b32c9971ef368df;received=88.96.247.242;rport=10357
From: "84415672" <sip:84415672@sip.voiceflex.com >;tag=70a7f499f20d0f2e
To: <sip:07767491241@sip.voiceflex.com>;tag=as462fcd76
Call-ID: 0a7a5a74bb6e2ed05383d0ec57948b7c@10.0.1.8
CSeq: 1579066102 CANCEL
User-Agent: VoiceFlex
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:07767491241@146.101.248.200>
Content-Length: 0

7727614mS SipDebugInfo: SIPDialog TXN : Decoding of message Succeded 1
7727615mS SipDebugInfo: Find End Point 17.3131.0 -1 SIPTrunk Endpoint (f53fea3c) Sip CallId 0a7a5a74bb6e2ed05383d0ec57948b7c@10.0.1.8
7727615mS SipDebugInfo: 17.3131.0 -1 SIPTrunk Endpoint(f53fde18) Process SIP response dialog f53fde18, method CANCEL, CodeNum 200 in state SIPDialog::FINAL(0)
7732610mS SipDebugInfo: Timer 4 callback
7732610mS SipDebugInfo: 17.3131.0 -1 SIPTrunk Endpoint(f53fde18) Completed ... removing Dialog of CallId: 0a7a5a74bb6e2ed05383d0ec57948b7c@10.0.1.8 and State: SIPDialog::FINAL(40)
7732610mS SipDebugInfo: 17.3131.0 -1 SIPTrunk Endpoint(f53fde18) SIPDialog destructor ... f53fde18
7732611mS SipDebugInfo: ~SipTrunkEndpoint 17.3131.0 -1 SIPTrunk Endpoint
7732614mS SipDebugInfo: Timer 11 callback
7732614mS SipDebugInfo: Dialog has been deleted
7800555mS SipDebugInfo: SIPTrunks: Make Target voip, line group id is 0 and ip 146.101.248.200
7800555mS SipDebugInfo: SIP Line (17) cannot find a suitable SIP URI to dial out

 

[Bas1234]

Is your public IP address OK?
System > Lan 1 or 2 > Network Topology

If your using Lan 2 put it on "Open Internet"

Do you use a Cisco router?

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

 

[Bas1234]

If your using Lan 1 or 2 put it on "Open Internet"
If your using Lan 2 put it on turn of the Firewall

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

 

[HBPSYS]

I have tried it through both Lan 1 and 2 but did the same thing. I am now back to using LAN 1 and behind a Watch Guard X550E but it has got a static NAT rule through to the IP address of the IP Office LAN 1 for UDP 5060 which is the voiceflex SIP Port

 

[Bas1234]

What is there behind the WatchGuard how is that configured?

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

 

[HBPSYS]

The watchguard goes straight in to a dlink switch which is the same switch the phone system is plugged into. The other end of the watchguard there is a NO NAT Router. Im not to sure if it is a firewall issue as I can make an outgoing call so I think it must have registered with the SIP providor to be able to do this

 

[Bas1234]

Is it a SIP trunk you're using from voiceflex?

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

 

[Pepp77]

We use Voiceflex SIP trunks in about 15 of our customer sites.

We always use LAN 2 where we can to keep the calls off the local network, but have a couple set up to use LAN 1.

Have you enabled STUN on the LAN Network Topology tab using 146.101.248.221 as the STUN server IP address?

 

[HBPSYS]

Thats good then its good that someone else has got this working I have tried the default STUN IP and Voiceflex's one but I will try it again now to see as I have made alot of changed since Thanks.

 

[HBPSYS]

Just a quick one when you RUN the STUN Server by the RUN STUN button is this real time or do you need to save the config back to the system?

 

[Pepp77]

Run Stun and then once its finished running resend the config back to the control unit. Also ensure Run STUN at startup is ticked.

 

[Bas1234]

This is real time, it will come back with the "right" settings but it the provider doesn't use STUN the untick it and leave the ip addr 0.0.0.0.

Try to add an nother SIP URI and change the 3 "Use Authentication Name" to your external number.

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

 

[HBPSYS]

I dont belive it it is now working!! I think in the end it was a combernation of making sure I have A NAT entry through my firewall to my Phone system for UDP Port 5060 and also Making sure I had the right Stun server IP in on the LAN 1 card. Pepp77 Thanks so much for all your help Im new to this forum and if there is any way of giving you points like on experts exchange then I will do. Your Avaya Knowledge is Top Class Once again thanks.

 

[HBPSYS]

Sorry I jumped the gun a bit I think we have made progress as I got a call come through but I then hung up and then rang the number again and now it just goes dead after about 15 Secs and disconnects me. It now does this every time

 

[Bas1234]

Some provider have the problem that it's not possible to call out over SIP to the same SIP number, it better to use you mobile or call out over ISDN/Alog

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

 

[HBPSYS]

I think I have found something the firt time I put the voiceflex Stun servers IP address in the firewall went to unknown and the Public IP address wne to 0.0.0.0 and that when I managed to get a SIP call in. Then when it stopped workin again a recieved the config again and in the STUN section it has now put in port restricted cone NAT and My public IP address and it showing an error saying communication is not possible unless is supported on the same IP address as the ITSP. Any ideas ? And Sorry BAS1234 and have just noticed that most of the posts were from you so Thanks goes out to you as well.