Avaya IP Office unable to respond sip invite from Call manager8.5,but IP Office able to call CUCM

[farazshamsi]

Hi,

I am facing the problem to integrate Call manager with Avaya IP Office with SIP Trunk and Avaya IP Office able to Call cisco Call manager and but Call manager unable to call to Avaya IP Office Below are the Call flow and we isolate gateway from this scenario and make direct sip trunk between Call manager and Avaya IP OFfice (6.0)

Cisco Phone----- Call manager-----------SIP----Avaya IP Office ----------Avaya Phone

A



As per cisco TAC engineer Avaya IP Office unable to respond initial sip invite from cisco Call manager. Please let me know from Avaya Side

 

[derfloh]

Do you have a */*/* URI at the IP office side? Do you see messages in SysMon?

 

[amriddle01]

The other question is have to done this before?
My point being you haven't conveyed how it's programmed, it could be poor programming or a slight tweak...

 

[farazshamsi]

Avaya Monitor Log

2BF760F2@172.16.26.11
1997852054mS Sip: ~SipTrunkEndpoint 17.30592.1 -1 SIPTrunk Endpoint
1997857246mS Stun: Info: media ports will not be mapped with STUN.
1997857246mS Sip: License, Valid 1, Available 5, Consumed 1
1997857247mS Sip: CheckLineMonitors on SIP Endpoint - KEY & LAMP for SIP Trunk!
1997857248mS Sip: 17.30593.1 -1 SIPTrunk Endpoint(f54cedb0) SendSIPResponse: INVITE code 100 SENT TO 172.16.26.11 5060
1997857250mS Sip: 17.30593.1 -1 SIPTrunk Endpoint(f54cedb0) SendSIPResponse: INVITE code 503 SENT TO 172.16.26.11 5060
1997857282mS Sip: Find End Point 17.30593.1 -1 SIPTrunk Endpoint (f552eec8) Sip CallId CF2D3DF1-CA8511E5-8DB5E4A2-

 

[amriddle01]

You need to select the SIP options in the filter settings of monitor, that race isn't any good really.
You still need to tell us how it's programmed, impossible to help otherwise.

P.s I've connected many different systems to the IPO using SIP, the IPO side is always the easiest and most flexible

 

[farazshamsi]

This is long outstanding issue so I appreciate if anyone let me know from Avaya Side. This is intail scenaria where we implemented but found that 503 service is unavailable due to SDP invite missing, but when we intiate sdp invite along with SIP early offer for Avaya IP Office we get 486 Busy here. I am trying to make simple and isolate gateway in between and make direct SIP Trunk between cisco CallManager and Avaya IP Office so please let me know the step. Below is the Cisoc TAC verdict and Logs, But from the Avaya Side so I donot have support available so please Avaya Engineer help me in this regard.

Cisco TAC Verdict:

I went through the logs shared by you. You can find my analysis for the successful call below. You can also find the analysis for the failed call as mentioned by Stuti. Please note that we are doing Early Offer on the SIP leg in both cases, and the call is rejected by Avaya in the failed call and it is answered with a 200 OK in the successful call.

Please check with the Avaya Engineers as to why the call is being rejected even after sending SDP in the INVITE.

Even after making the changes on the CUCM and the GW to ensure that we do a fast-start on the H323 leg and send SDP in the INVITE to the Avaya IP Office, Stuti has mentioned that Avaya is disconnecting the call with a 486 User Busy here.

Logs


FAILED CALL from 2810 to 2023


Sent:
INVITE sip:2023@192.168.39.30:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.26.11:5060;branch=z9hG4bK2DD3DF8
Remote-Party-ID: <sip:2810@172.16.26.11>;party=calling;screen=yes;privacy=off
From: <sip:2810@172.16.26.11>;tag=677D76C0-944
To: <sip:2023@192.168.39.30>
Date: Mon, 01 Feb 2016 17:04:32 GMT
Call-ID: B2B367A3-C83C11E5-8B88EA1F-4AB1664@172.16.26.11
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800

Cisco-Guid: 0006994241-0226881898-0083938305-3232251723
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1454346272
Contact: <sip:2810@172.16.26.11:5060>
Expires: 180
Allow-Events: telephone-event


Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 294

v=0
o=CiscoSystemsSIP-GW-UserAgent 1527 4071 IN IP4 172.16.26.11
s=SIP Call
c=IN IP4 172.16.26.11
t=0 0
m=audio 16596 RTP/AVP 18 101 19
c=IN IP4 172.16.26.11
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20

4778839: Feb 1 22:04:32.660: //452438/006AB9410500/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.26.11:5060;branch=z9hG4bK2DD3DF8
From: <sip:2810@172.16.26.11>;tag=677D76C0-944
To: <sip:2023@192.168.39.30>;tag=427fc7f3f50ce6e1
Call-ID: B2B367A3-C83C11E5-8B88EA1F-4AB1664@172.16.26.11
CSeq: 101 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO
Content-Length: 0


4778840: Feb 1 22:04:32.664: //452438/006AB9410500/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 172.16.26.11:5060;branch=z9hG4bK2DD3DF8
From: <sip:2810@172.16.26.11>;tag=677D76C0-944
To: <sip:2023@192.168.39.30>;tag=427fc7f3f50ce6e1
Call-ID: B2B367A3-C83C11E5-8B88EA1F-4AB1664@172.16.26.11
CSeq: 101 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO
Content-Length: 0







SUCCESSFUL CALL from 4661 to 2016– H323 Fast Start and Avaya accepts the call

Cisco phone---CME---Gatekeeper(H323)------(H323|)Gateway---SIP----AVAYA IP OFFICE---AVAYA PHONE


Incoming Dial-peer=1452 and Outgoing Dial-peer=2010
087562: Feb 3 14:09:46.013: //40004/A88186FDA387/H323/setup_ind: callingNumber[4661] calledNumber[2016]

H323 SETUP (Fast Start):

087557: Feb 3 14:09:45.997: H225.0 INCOMING PDU ::=

value H323_UserInformation ::=
{
h323-uu-pdu
{
h323-message-body setup :
{
protocolIdentifier { 0 0 8 2250 0 4 }
sourceAddress
{
h323-ID : {"Gulistan-e-Jauhar-2"}
}
sourceInfo
{
vendor
{
vendor
{
t35CountryCode 181
t35Extension 0
manufacturerCode 18
}
productId '436973636F47617465776179'H
versionId '32'H
}
gateway
{
protocol
{
voice :
{
supportedPrefixes
{

{
prefix dialedDigits : "1#"
}
}
}, h323 :
{
supportedPrefixes
{
}
}
}
}
mc FALSE
undefinedNode FALSE
}
activeMC FALSE
conferenceID 'A88186FDC98511E5A387B5F7F4174E04'H
conferenceGoal create : NULL
callType pointToPoint : NULL
sourceCallSignalAddress ipAddress :
{
ip 'C0A83821'H
port 57540
}
callIdentifier
{
guid 'A8822325C98511E5A389B5F7F4174E04'H
}
fastStart
{
'0000000D4001800B05000100C0A83821679500'H,
'400000060401004D4001801215000100C0A83821...'H
}
mediaWaitForConnect FALSE
canOverlapSend FALSE
multipleCalls TRUE
maintainConnection TRUE
symmetricOperationRequired NULL
}
h245Tunneling TRUE
nonStandardControl
{

{
nonStandardIdentifier h221NonStandard :
{
t35CountryCode 181
t35Extension 0
manufacturerCode 18
}
data 'C0010018800006000400000003'H
}
}
}
}




########GW sent INVITE with SDP


087629: Feb 3 14:09:46.097: //40005/A88186FDA387/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:2016@192.168.39.30:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.26.11:5060;branch=z9hG4bK4D625FD
Remote-Party-ID: "Mohsin Ali -OM" <sip:4661@172.16.26.11>;party=calling;screen=no;privacy=off
From: "Mohsin Ali -OM" <sip:4661@172.16.26.11>;tag=5DA94B8-10E6
To: <sip:2016@192.168.39.30>
Date: Wed, 03 Feb 2016 09:09:46 GMT
Call-ID: B438C5E7-C98C11E5-A19EE4A2-2BF760F2@172.16.26.11
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2827060989-3380941285-2743580151-4095168004
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1454490586
Contact: <sip:4661@172.16.26.11:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 294

v=0
o=CiscoSystemsSIP-GW-UserAgent 8509 3617 IN IP4 172.16.26.11
s=SIP Call
c=IN IP4 172.16.26.11
t=0 0
m=audio 19276 RTP/AVP 18 101 19
c=IN IP4 172.16.26.11
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20


########AYAVA IP Office sends 100 Trying and 180 Ringing:

Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.26.11:5060;branch=z9hG4bK4D625FD
From: "Mohsin Ali -OM" <sip:4661@172.16.26.11>;tag=5DA94B8-10E6
To: <sip:2016@192.168.39.30>;tag=3de5460035407978
Call-ID: B438C5E7-C98C11E5-A19EE4A2-2BF760F2@172.16.26.11
CSeq: 101 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO
Content-Length: 0


087637: Feb 3 14:09:46.125: //40005/A88186FDA387/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.26.11:5060;branch=z9hG4bK4D625FD
From: "Mohsin Ali -OM" <sip:4661@172.16.26.11>;tag=5DA94B8-10E6
To: <sip:2016@192.168.39.30>;tag=3de5460035407978
Call-ID: B438C5E7-C98C11E5-A19EE4A2-2BF760F2@172.16.26.11
CSeq: 101 INVITE
Contact: "2016" <sip:2016@192.168.39.30:5060;transport=udp>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO
Content-Length: 0



####Avaya IP Office completes the call with a 200 OK with G729

088290: Feb 3 14:09:49.477: //40005/A88186FDA387/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 172.16.26.11:5060;branch=z9hG4bK4D625FD
From: "Mohsin Ali -OM" <sip:4661@172.16.26.11>;tag=5DA94B8-10E6
To: <sip:2016@192.168.39.30>;tag=3de5460035407978
Call-ID: B438C5E7-C98C11E5-A19EE4A2-2BF760F2@172.16.26.11
CSeq: 101 INVITE
Contact: "2016" <sip:2016@192.168.39.30:5060;transport=udp>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 228

v=0
o=UserA 865089734 1243411614 IN IP4 192.168.39.30
s=Session SDP
c=IN IP4 192.168.39.30
t=0 0
m=audio 49156 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


#####The call was disconnected by the Ayavay end

089011: Feb 3 14:09:57.049: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:4661@172.16.26.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.39.30:5060;rport;branch=z9hG4bKb065fe4ca94a3892fc20499471dc6241
From: "2016" <sip:2016@192.168.39.30>;tag=3de5460035407978
To: "Mohsin Ali -OM" <sip:4661@172.16.26.11>;tag=5DA94B8-10E6
Call-ID: B438C5E7-C98C11E5-A19EE4A2-2BF760F2@172.16.26.11
CSeq: 102 BYE
Contact: "2016" <sip:2016@192.168.39.30:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO
Content-Length: 0

 

[amriddle01]

Either tell us how it's programmed as requested and give us more detailed traces or indeed "ask your Avaya engineer"

 

[farazshamsi]

Hi,

I will share sys Monitor Logs with detail SIP trace and I will share the URI at the Avaya IP OFfice so might be suffice enough to find the actual root cause.

Anyway thanks

 

[farazshamsi]

Hi,

I am trying to make Simple SIP Trunk between Call manager and Avaya IP Office.I have made SIP Trunk On Cisco Call Manager and I need to receive this incoming sip trunk at Avaya IP OFFice so What configuration need to be done incoming sip Trunk at Avaya IP Office

Any Input would be highly appreciated

 

[derfloh]

Do you have a */*/* URI at the IP office side?

 

[amriddle01]

I am trying to make Simple SIP Trunk between Call manager and Avaya IP Office.I have made SIP Trunk On Cisco Call Manager and I need to receive this incoming sip trunk at Avaya IP OFFice so What configuration need to be done incoming sip Trunk at Avaya IP Office

You are asking for help with a connection you have already made, why won't you tell us what you've already done?
Then we can tell you where you have gone wrong....

 

[farazshamsi]

hi,

This is running envoirnment where I can receive call from the Cisco Gateway To Avaya IP Office, but unable to receive call from Calll manager to Gateway to Avaya IP Office where I receive intial 503 service is unavailable. I will try to make simple make sip trunk only between Call manager and Avaya IP Office so I am asking should I make configuration for for receiving this call from sip trunk. We are receiving all call from cisco gateway to IP Office. I hope i can clear to you.

Fuurthermore, I have enclosed URI Screen shot of Avaya IP Office.

 

[farazshamsi]

Hi,

Please anyone let me know.

 

[derfloh]

You have open questions to answer to get some tips.