Avaya 9621G SIP phones Name not showing 1

[hussainvlr]

Hi Experts ,

i have Avaya 9621G SIP version 6.x. phone also it is registered with Alcatel PABX OXE

When Alcatel phone calls to Avaya phone only the number is showing

i need to display both name & number any idea ?

But from Alcatel i can able to see the name also showing in the traces but why the name was not showing Avaya is there any bug ?

 

[kyle555]

what's your config file look like?

## DISPLAY_NAME_NUMBER specifies whether the name and/or number will be displayed for
##  incoming calls, and if both are displayed, the order in which they are displayed.
##  Value  Operation
##    0: display calling party name only
##    1: display calling party name followed by calling party number
##    2: display calling party number only 
##    3: display calling party number followed by calling party name
##  This parameter is supported by:
##       J169/J179 SIP R1.5.0;        valid values 0 through 3; the default value is 0.
##       96x1 SIP R6.2 and later;        valid values 0 through 3; the default value is 0.
##       96x1 SIP R6.0.x;                valid values 0 through 1; the default value is 0.
##       96x0 SIP R2.6.5 and later;    valid values 0 through 3; the default value is 0.
##       96x0 SIP R2.0 through R2.6.4; valid values 0 through 1; the default value is 0.
## SET DISPLAY_NAME_NUMBER 0

 

[hussainvlr]

Hi,

Thanks for your answer

I'm new to Avaya can you tell me where to check these settings in Avaya SIP phones ?

 

[kyle555]

I think SNMP polling might tell you if its enabled. Otherwise, SNMP config, and that parameter you're looking for are defined in the 46xxsettings.txt file

When a phone boots, usually DHCP provides option 242 with HTTPSRVR and perhaps HTTPDIR, or TLSSRVR and possibly TLSDIR.
If HTTPSRVR = 1.2.3.4 and HTTPDIR = site1, the phone will ask for

http://1.2.3.4/site1/46xxsettings.txt

If you used TLSSRVR and TLSDIR instead, it would ask fro

https://1.2.3.4/site1/46xxsettings.txt

The DIR parameter is optional if you have 1 settings file in the root of your http server, it'll just ask for

http://1.2.3.4/46xxsettings.txt

That contains most of parameters the endpoint uses.

 

[hussainvlr]

Hi,

Thanks for your answer

I will check and let me know the update but sure the Avaya SIP phone will show name when it is registered with. Third party PABX ?

 

[kyle555]

Yes, what I've outlined is how a phone boots up, period.
In that settings file there are configs like "avaya or 3rd party environment" and whether its 9600 or the new J100 series that are certified on Broadsoft, the various settings for avaya and non-avaya networks are a prerequisite and all in there. Unless you just configured your phone by hand from scratch, in which case, start using a settings file and you'll have more options available to play with than just those in the phone menu!

 

[hussainvlr]

Hi,

I Checked in 46XX settings in Avaya SIP phone and value is 0 but still name is not showing

Any suggestions please ?

 

[hussainvlr]

Hi,

I can able to see the name & number on Avaya SIP phones once receive the incoming call from Alcatel Phone to Avaya phone thereafter hold the call and resume back .

But i need to show both name and number while during ringing phase (before answering the call ) is there any way to sort it out ?

 

[kyle555]

try newer firmware? Are the FROM and PAI headers the same on the original invite vs on the hold/unhold?

 

[hussainvlr]

Currently the firmware for Avaya is 6.5.X so is there new version available

Can you share the link if you have ? Also how to check the FROM & PAI headers is it in 46XX settings ?

 

[hussainvlr]

After upgraded to the latest firmware (7.1.3)finally there where no option to add Proxy server details in the phone

In phone showing Press New to add proxy server

Only "Back" option is showing & there is no new option

Someone can help me to figure it out

 

[hussainvlr]

Hi,

Please find the FROM & PAI section below


----------------------utf8-----------------------
SIP/2.0 180 Ringing
[From: <sip:3501@192.168.30.254;user=phone>;tag=aa05a15e09a863a5b419599de6202bd7 [
To: <sip:3550@192.168.30.254;user=phone>;tag=386d90cf5c1d96021b706l5m4t595a118512o3b2b51543169u_T3550192.168.30.247
Call-ID: e0449064d607f8dab2711c650c063283@192.168.30.254
CSeq: 2009436317 INVITE
Via: SIP/2.0/TCP 192.168.30.254;branch=z9hG4bK62e3c0579f6f2cdae08edff25afe75fa
Allow: INVITE,ACK,BYE,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE,REFER,INFO,PRACK,PUBLISH,UPDATE
User-Agent: Avaya one-X Deskphone 7.0.1.4.6 (6)
Contact: <sip:3550@192.168.30.247:8324;transport=tcp>
Accept-Language: en
Content-Length: 0


This was after hold from Avaya SIP extension

SIP/2.0 200 OK
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
Contact: <sip:3501@192.168.30.254;transport=TCP>
User-Agent: OmniPCX Enterprise R12.0 m1.403.22.a
Session-Expires: 1800;refresher=uas
P-Asserted-Identity: "hussain mohd" <sip:3501@192.168.30.254;user=phone>

 

[kyle555]

well, I'd want to see the original invite to the avaya phone, not the 180 ringing, but I know Avaya is big on p-asserted-identity.

See if the original invite to the phone includes that.

 

[hussainvlr]

INVITE sip:3550@192.168.30.247:8324;transport=TCP;avaya-sc-enabled SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO
Supported: replaces,timer,path
User-Agent: OmniPCX Enterprise R12.0 m1.403.22.a
Session-Expires: 1800;refresher=uac
Min-SE: 900
Content-Type: application/sdp
To: "Mohd Hussan" <sip:3550@192.168.30.254;user=phone>
From: <sip:3501@192.168.30.254;user=phone>;tag=aa05a15e09a863a5b419599de6202bd7
Contact: <sip:3501@192.168.30.254;transport=TCP>
Call-ID: e0449064d607f8dab2711c650c063283@192.168.30.254
CSeq: 2009436317 INVITE
Via: SIP/2.0/TCP 192.168.30.254;branch=z9hG4bK62e3c0579f6f2cdae08edff25afe75fa

3550 --- Avaya SIP extension need to display name "hussain mohd" along with number 3501 but it is not happening pls refer the above log

 

[hussainvlr]

Sun Sep 16 17:27:47 2018 [display_ipc_in] ------------ Begin ---------------
Sun Sep 16 17:27:47 2018 neqt : 1646 Id : 1
Sun Sep 16 17:27:47 2018 INVITE
Sun Sep 16 17:27:47 2018 REQUEST URI : <> 3550@test:5060 ; user=phone
Sun Sep 16 17:27:47 2018 FROM : <> 3501@192.168.30.254:5060 ; user=phone

 

[kyle555]

Just to be clear - 3550 is the Avaya phone and that invite is the message that was presented to it?

I'd say you need the name in the FROM or PAssertedIdentity header and not the TO header if that is the case.

 

[hussainvlr]

Yes 3550 is the Avaya SIP extension

 

[kyle555]

Wouldn't that mean that the name should be in the FROM header then?

To: "Mohd Hussan" <sip:3550@192.168.30.254;user=phone>
From: <sip:3501@192.168.30.254;user=phone>;tag=aa05a15e09a863a5b419599de6202bd7

 

[hussainvlr]

Yes any advice ?

 

[janni78]

Extension 3550 is Mohd Hussan?

If 3501 calls 3550 it should show the name of extension 3501, which in this case isn't available in the From header.

"Trying is the first step to failure..." - Homer