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Avaya SIP Phone con CUCM

rpsalaza
Cisco Employee
Cisco Employee

‎05-04-2016 02:22 PM - edited ‎03-17-2019 06:49 AM

Hello

Have any of you tried to register an Avaya SIP phone on CUCM? I tried. The telephone registered well,  and it is posible to dial from it to a Cisco phone, but when  I dial from a Cisco phone, the Avaya phone doesn't ring. The Avaya phones is an 9608 SIP phone, If it worked for any of you, please share the configuration that worked.

Regards

Ricardo.

2 people had this problem
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14 Replies 14

Dennis Mink
VIP Alumni
VIP Alumni

‎05-04-2016 11:00 PM

what do the traces say of such a failed call?

Please remember to rate useful posts, by clicking on the stars below.

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rpsalaza
Cisco Employee
Cisco Employee
In response to Dennis Mink

‎05-05-2016 07:46 AM

Hello Dennis and Manish

I checked the CSS. All phones are in the same CSS. The analysis done in DNA show the following results:

When Avaya dial Cisco

  • Results Summary
    • Calling Party Information
    • Dialed Digits = 3001
    • Match Result = RouteThisPattern
    • Matched Pattern Information
    • Called Party Number = 3001
    • Time Zone =
    • Call Classification = OnNet
    • InterDigit Timeout = NO
    • Device Override = Disabled
    • Outside Dial Tone = NO

When Cisco dial Avaya

  • Results Summary
    • Calling Party Information
    • Dialed Digits = 3003
    • Match Result = RouteThisPattern
    • Matched Pattern Information
    • Called Party Number = 3003
    • Time Zone = Etc/GMT
    • Call Classification = OnNet
    • InterDigit Timeout = NO
    • Device Override = Disabled
    • Outside Dial Tone = NO

Also I attached the traces done by CUCM. I noted that the Avaya phone doesn't answer the "Invite" packet sent by CUCM.

If you have any advice, I will be very grateful 

Regards

ucm-pub.ciscolocal.com_.zip
0 Helpful

Deepak Mehta
VIP Alumni
VIP Alumni
In response to rpsalaza

‎05-05-2016 09:01 AM

Can you provide phone IP address and CUCM IP to looks through the logs.thanks

0 Helpful

rpsalaza
Cisco Employee
Cisco Employee
In response to Deepak Mehta

‎05-05-2016 09:56 AM

Hello Deepak

CUCM: 172.27.199.11

CUC: 172.27.199.12

Cisco Phone: 172.27.199.47 DN:3001

Avaya Phone: 172.27.199.51 DN:3003

Thanks

0 Helpful

Deepak Mehta
VIP Alumni
VIP Alumni
In response to rpsalaza

‎05-05-2016 12:55 PM

CUCM is sending the 404 not found to cisco phone as it is unable to find the route.

Can you ensure avaya phone has DN a partition assigned .This partition should be in CSS of the cisco phone.thanks

SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 172.27.199.47:49704;branch=z9hG4bK02d7c771
From: <sip:2001@172.27.199.47>;tag=2834a283bb4f069741e36b2b-2f232e64
To: <sip:2003@ucm-pub.ciscolocal.com>;tag=15340800
Date: Thu, 05 May 2016 13:12:50 GMT
Call-ID: 2834a283-bb4f006e-13af2e88-4264ce90@172.27.199.47
CSeq: 1000 SUBSCRIBE
Server: Cisco-CUCM10.5
Content-Length: 0


03622234.000 |06:12:50.277 |SdlSig |SIPNotifyReq |wait |SIPHandler(1,100,79,1) |Notifier(1,100,124,181) |1,100,14,43436.1963^172.27.199.47^SEP2834A283BB4F |[T:N-H:0,N:1,L:0,V:0,Z:0,D:0] --TransType=1 --TransSecurity=0 PeerAddr = 172.27.199.47:49704 addrList:
03622234.001 |06:12:50.277 |AppInfo |//SIP/Stack/Info/0x0/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 7 (SIP_APPLICATION_SUBSCRIBE_NOTIFY_MSG), for event 204 (UNDEFINED)
03622234.002 |06:12:50.277 |AppInfo |//SIP/Stack/Info/0x0/ccsip_spi_process_app_subscribe_event: Event [SIPSPI_EV_CC_NOTIFY] received in State [SUBSCRIBE_STATE_IDLE]
03622234.003 |06:12:50.277 |AppInfo |//SIP/Stack/Info/0x0xdfeeb48/sipSPIPutSCBInSubIDTable: SCB with key key 44147 already added to SubIDTable
03622234.004 |06:12:50.277 |AppInfo |//SIP/Stack/Info/0x0/ccsip_new_tcb: Created TCB: 0xc337960
03622234.005 |06:12:50.277 |AppInfo |//SIP/Stack/Info/0x0/sipSPIUpdateTCBParent: TCB 0xc337960's parent is 0xdfeeb48
03622234.006 |06:12:50.277 |AppInfo |//SIP/Stack/Error/0x0xdfeeb48/sipSPIGetDestUrlFromContainer: no target to send request

pwoll
Level 1
Level 1
In response to Deepak Mehta

‎07-11-2016 02:18 PM

I have the same issue.  Was this ever successfully resolved?

0 Helpful

miket
Level 5
Level 5
In response to pwoll

‎10-25-2016 08:07 PM

Has anyone got the solution

0 Helpful

Manish Gogna
Cisco Employee
Cisco Employee
In response to rpsalaza

‎05-06-2016 12:27 AM

In addition to the nice analysis by Deepak I can see that the DN 3003 is assigned PCP_Base_PT partition as per the digit analysis in the traces. As a test remove any partition from this 3003 DN and try calling it again. If it still does not work then it could be something else causing the issue.

Manish

0 Helpful

Manish Gogna
Cisco Employee
Cisco Employee

‎05-05-2016 12:24 AM

Hi Ricardo,

If basic settings like CSS/partition and any overlapping dial plan is checked then as Dennis said you should get detailed callmanager service traces to see the digit analysis and why the call is failing.

https://supportforums.cisco.com/document/126666/collecting-cucm-traces-cucm-862-tac-sr

Manish

0 Helpful

miket
Level 5
Level 5

‎10-25-2016 11:09 AM

Have you got an answer because I have this exact problem

0 Helpful

pwoll
Level 1
Level 1
In response to miket

‎10-25-2016 11:48 AM

Unfortunately, there was never a successful solution found.  Mr Salazar found that the Avaya phone works using CME and I confirmed that.  So, one possibility is to use CME as a gateway to CUCM, but that is a non-optimum solution.  I escalated the case with Cisco TAC and have comparison captures between CME and CUCM, but TAC indicates that the Avaya is not accepting the SIP request from CUCM.  They recommend obtaining Avaya support.  However, TAC also indicated that there is little option to modify the SIP request in CUCM unless there is a SIP trunk involved, so even if we had info from Avaya, the chance of resolving this is low.  

0 Helpful

Vishal Gupta.
Level 1
Level 1

‎11-01-2019 02:41 AM - edited ‎11-01-2019 02:46 AM

Hello Ricardo,

 

Though it is a very old post but i think i have a solution for your issue, thanks to one of my colleague for this.

We faced exactly the similar issue with Incoming calls to Avaya when dialed from a Cisco IP Phone and the culprit was one of the SIP Header being sent out in an Invite message to Avaya from CUCM. Which was "Alert-Info: <file://Bellcore-dr1/>"

 

I can find the same Header in the logs shared by you.

Call IDs:

From Cisco IP Phone to CUCM
2834a283-bb4f001b-1b6e1abd-4299945b


From CUCM to AVAYA Phone
507a3900-72b143e3-7979-bc71bac

 

Incoming Invite msg from Cisco IP Phone to CUCM:

 

03619294.002 |06:00:19.534 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 172.27.199.47 on port 49704 index 923 with 1518 bytes:
[100940,NET]
INVITE sip:3003@ucm-pub.ciscolocal.com;user=phone SIP/2.0
Via: SIP/2.0/TCP 172.27.199.47:49704;branch=z9hG4bK30a1b550
From: "3001" <sip:3001@ucm-pub.ciscolocal.com>;tag=2834a283bb4f06890aee3aff-0b1653a7
To: <sip:3003@ucm-pub.ciscolocal.com>
Call-ID: 2834a283-bb4f001b-1b6e1abd-4299945b@172.27.199.47
Max-Forwards: 70
Date: Thu, 05 May 2016 13:00:23 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP8861/10.2.2
Contact: <sip:c898b91c-ec05-2600-5582-e13e42c3ff31@172.27.199.47:49704;transport=tcp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "3001" <sip:3001@ucm-pub.ciscolocal.com>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 408
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 26171 0 IN IP4 172.27.199.47
s=SIP Call
t=0 0
m=audio 26986 RTP/AVP 102 9 124 0 8 116 18 101
c=IN IP4 172.27.199.47
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:124 ISAC/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

 

 

Outgoing Invite from CUCM to Avaya Phone:

 

03619371.001 |06:00:19.541 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 172.27.199.51 on port 1029 index 1063
[100942,NET]
INVITE sip:3003@172.27.199.51:1029;transport=tcp;avaya-sc-enabled SIP/2.0
Via: SIP/2.0/TCP 172.27.199.11:5060;branch=z9hG4bK7be763281518
From: <sip:3001@172.27.199.11>;tag=44102~4aca1d00-fcbd-60df-8ac5-ac9f97ba2d7d-20639160
To: <sip:3003@172.27.199.11>
Date: Thu, 05 May 2016 13:00:19 GMT
Call-ID: 507a3900-72b143e3-7979-bc71bac@172.27.199.11
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Send-Info: conference, x-cisco-conference
Alert-Info: <file://Bellcore-dr1/>
Contact: <sip:3001@172.27.199.11:5060;transport=tcp>
Remote-Party-ID: <sip:3001@172.27.199.11;x-cisco-callback-number=3001>;party=calling;screen=yes;privacy=off
Max-Forwards: 69
Content-Length: 0

 

In order to resolve this,

1. We configured a SIP Normalization Script (Screenshot Attached) to remove the Alert-Info Header from the INVITE message.

2. Applied the Normalization script to a SIP Profile.

3. Applied the SIP Profile to Avaya Phone.

                 Issue Resolved :)

 

Regards,

Vishal

617 KB

mfernandezm
Level 1
Level 1
In response to Vishal Gupta.

‎12-24-2019 08:55 AM

 

Vishal:

 

We found your response to Ricardo´s problem and tried to apply your script to our CUCM Ver 8.6.2, but now we don´t know how to bind or link it to the Phone Configuration.

 

First, we used your script to create a Normalization Script and then, applied it to a new Trunk Configuration (evidences A).

Then we created a new SIP Profile to be used with the Avaya Phones, based on the Third Party SIP Device template; this new Profile is invoked from the SIP Phone Configuration form (evidences B).

 

I think we are missing some step, because I don´t see a link between this 2 procedures, and the Avaya phones are not responding the way they should.

 

I appreciate your help a lot.

 

 

Avaya Phone in CUCM 862.zip
0 Helpful

Vishal Gupta.
Level 1
Level 1
In response to mfernandezm

‎01-05-2020 07:08 PM

Hello,

 

My CUCM version was 10.5 and this version have an option to add Normalization Script to SIP profile too apart from SIP Trunk, as can be seen in the attached screenshot.

 

So in my case, there was no SIP Trunk involved, instead a Script was applied to SIP Profile and Profile was applied on Third Party SIP device configuration page.

 

Even I just verified at my end and came to know that CUCM 8.6 does not have a provision/an option to apply Normalization script directly to SIP Profile.

I will update on this thread, in case I come across any workaround for CUCM 8.6 as well.

 

Regards,

Vishal

66 KB
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