Name not display

[mdhussainvlr]

I have OXE R12 registered with Avaya SIP phones but when I calling from OXE to Avaya SIP Phone the name was not showing only it displays number

Is there any possibility to show both name and number on Avaya SIP phone

[mdhussainvlr]

Hi,
Has anybody did this one ?

[mdhussainvlr]

After putting the hold and resume the call in Avaya SIP extension then only name displaying

Is there any possibility to display name also during ringing phase ?

[cavagnaro]

Do a sip trace and check ask what fields are used by Avaya...then change OXE sip trunk according to what Avaya needs. We are not Avaya experts

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[mdhussainvlr]

----------------------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>


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[cavagnaro]

So it is p-asserted identity. Now modify your trunk configuration at OXE to include such parameter


Enviado de meu E6633 usando o Tapatalk

[haroun]

Do a sip trace and check ask what fields are used by Avaya...then change OXE sip trunk according to what Avaya needs. We are not Avaya experts

Enviado de meu E6633 usando o Tapatalk

----------------------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>


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200.OK response

The P-Asserted-ID header of the 200.OK response contains the identity of the connected party. This header can contain Tel URI.
see the p-asserted id parameters at Ext sip gateway

[mdhussainvlr]

I was not using any SIP trunk only connectIng the Avaya SIP phone with OXE


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[mdhussainvlr]

While calling from Avaya SIP to NOE phone during ringing phase the name
And number of NOE was showing in Avaya Sip extension

Also find the logs for the same

utf8-----------------------
SIP/2.0 180 Ringing
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
Contact: sip:192.168.30.254
User-Agent: OmniPCX Enterprise R12.0 m1.403.22.a
P-Asserted-Identity: "hussain mohd" <sip:3501@192.168.30.254;user=phone>
Content-Type: application/sdp
To: <sip:3501@192.168.30.254>;tag=71108ebf442ce10b2ac64f9b007c8ee2
From: <sip:3550@192.168.30.254>;tag=386d9c8630229df3j1w4d1a623p5b2q5i492s4i4k4l2l5y3t33_F3550192.168.30.247
Call-ID: 21_386d9c8615a248b15k30241y3e5q2v1v3a6o2g1y3d4a1d1r6512_I3550192.168.30.247
CSeq: 33 INVITE
Via: SIP/2.0/TCP 192.168.30.247:23941;branch=z9hG4bK21_386d9c86721e4bef1p67v3z5y6q4o2w172t4ea3j4s379723y_I3550192.168.30.247
Content-Length: 220
v=0
o=OXE 1537271999 1537271999 IN IP4 192.168.30.254
s=abs
c=IN IP4 192.168.30.220
t=0 0
m=audio 32576 RTP/AVP 8 120
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:120 telephone-event/8000


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[cavagnaro]

Well you are doing tests on a non validated set. There are no settings or help to go.
It is what you get to work


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