Avaya 9608 SIP phone registration

[astroshark]

Hi,

This is my first post about something that is probably quite simple so here goes.

Has anyone ever successfully registered an Avaya 9608 sip phone on Asterisk?
they fall under the 91xx sip family so I have software upgraded them to SIP 6.3
46xx settings file is running ok from a http server, i just can't get them to talk to Asterisk.

[midhatsalah]

Hey,

Did u get to solve this issue, as i am facing the same dilemma. Please let me know how u solved it and share it.

[yardietraxx]

I have successfully deployed a few of them with FreePBX 5.211.65-12 with Asterisk 11.9.0

[astroshark]

yardietraxx,

I'd be really interested in your setup so i can see where I'm going wrong.
did you manually config your phones via craft, or use a 46xx settings file?

I don't need it to make external calls yet, just want to be able to make extension to extension calls.

I have a domain controller running DHCP and my Asterisk server all connected with various other pcs via a cisco managed switch that I believe I have setup correctly.

The phone powers ok and is issued an IP address I get no errors until i try and log the phone in, where it stops, and will not talk to Asterisk. nothing in the log to say it's connecting, just sits there.

I haven't played with it for a while and need to get this resolved, (I have stuff lying around everywhere!

[pet12]

Hi,

i have the same problem.
yardietraxx - could you write some about your configuration? What i have to do to register an Avaya 9608 sip phone on Asteriks?

[gourou5169]

Hi everyone,

I tried too and have the same problem after entering the username and the password (sip account and sip password from sip.conf) the phone try to Acquire Services .... infinitely.

Here is the tcpdump of sip exchange :

Code: Select all

REGISTER sip:192.168.200.16 SIP/2.0
From: <sip:10@192.168.200.16>;tag=386dbf5c32bb683d3t2n5d3m4q3ra336l6l5320584j2i4v1634_F100.0.0.0
To: <sip:10@192.168.200.16>
Call-ID: 1_386dbf5cadebff12s5j5n6g1e2d5f1y3x3d4g2l3m6q6h4i5210_R100.0.0.0
CSeq: 1 REGISTER
Max-Forwards: 70
Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK1_386dbf5d-47b3bbd92m231n451q1256v5m1559j6qd3o181p5n_R100.0.0.0
Supported: eventlist,feature-ref,replaces,tdialog
Allow: INVITE,ACK,BYE,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE,REFER,INFO,PUBLISH,UPDATE
User-Agent: Avaya one-X Deskphone 6.4.1.25 (25)
Contact: <sip:10@0.0.0.0;transport=udp;avaya-sc-enabled>;q=1;expires=900;+avaya.gmtoffset="0:00";+avaya.js-ver="1.0";+avaya.model="9608";+avaya.sn="12WZ273608WY";+avaya.firmware="S96x1_SALBR6_4_1r25_V4r83.tar";+sip.instance="<urn:uuid:00000000-0000-1000-8000-7038eed21b34>";reg-id=1
Content-Length:     0

09:44:30.307449 IP ipbx.wisp-e.fr.sip > AVXD21B34.wisp-e.com.1025: SIP, length: 612
E.... ..@.O~.......m.....l.MSIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK1_386dbf5d-47b3bbd92m231n451q1256v5m1559j6qd3o181p5n_R100.0.0.0;received=192.168.200.109
From: <sip:10@192.168.200.16>;tag=386dbf5c32bb683d3t2n5d3m4q3ra336l6l5320584j2i4v1634_F100.0.0.0
To: <sip:10@192.168.200.16>;tag=as7bb9cc80
Call-ID: 1_386dbf5cadebff12s5j5n6g1e2d5f1y3x3d4g2l3m6q6h4i5210_R100.0.0.0
CSeq: 1 REGISTER
Server: X-Lite release 1103m
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="733087dd"
Content-Length: 0


Anyone ? any advices ?

[david55]

Trace is incomplete or the phone doesn't know any passwords at all. The normal sequence is REGISTER (no authentication) - 401 - REGISTER (authentication) - OK.

[mvenkatesh431]

Guys, did you ever get this working? I'm stuck aquiring services too.
My phone is using latest avaya sip firmware 6.5 and i am using Asterisk 11.7.

My avaya 9608 sending registration to asterisk and then asterisk challenging that request. Please look in to below trace

Code: Select all

<--- SIP read from UDP:192.168.1.234:1027 --->
REGISTER sip:192.168.1.100 SIP/2.0
From: <sip:200@192.168.1.100>;tag=557836d65854b665m4vcu1j1x3b4s5551t2e3w526k5z2c6t_F2000.0.0.0
To: <sip:200@192.168.1.100>
Call-ID: 5_557836d61f797b1b6f1s14154q5up29286l1n6c6w1347166x38_R2000.0.0.0
CSeq: 5 REGISTER
Max-Forwards: 70
Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK5_557836d6-509080bdz201e1t435hp433920x642t3s483d5sx_R2000.0.0.0
Supported: eventlist,feature-ref,replaces,tdialog
Allow: INVITE,ACK,BYE,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE,REFER,INFO,PUBLISH,UPDATE
User-Agent: Avaya one-X Deskphone 6.5.0.17 (17)
Contact: <sip:200@0.0.0.0;transport=udp;avaya-sc-enabled>;q=1;expires=300;avaya-actions="presence.initiate-pubsub,presence.redirect";+avaya.gmtoffset="+5:30";+avaya.js-ver="1.0";+avaya.model="9608";+avaya.sn="13N503301973";+avaya.firmware="S96x1_SALBR6_5_0r17_V4r83.tar";+sip.instance="<urn:uuid:00000000-0000-1000-8000-3475c7e34eac>";reg-id=1
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.1.234:1027 (NAT)

<--- Transmitting (NAT) to 192.168.1.234:1027 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK5_557836d6-509080bdz201e1t435hp433920x642t3s483d5sx_R2000.0.0.0;received=192.168.1.234;rport=1027
From: <sip:200@192.168.1.100>;tag=557836d65854b665m4vcu1j1x3b4s5551t2e3w526k5z2c6t_F2000.0.0.0
To: <sip:200@192.168.1.100>;tag=as7b141705
Call-ID: 5_557836d61f797b1b6f1s14154q5up29286l1n6c6w1347166x38_R2000.0.0.0
CSeq: 5 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="21ae18d1"
Content-Length: 0



Why Avaya sending contact as 200@0.0.0.0 ? Is this causing the issue ?

Thank you in advance.

[david55]

No. That will cause it to break once it receives and reacts correctly to the 401. At the moment it isn't getting far enough to look at anything but the authentication data.

[mvenkatesh431]

Guys it is working by using TCP instead of UDP. I used TCP in the 46xxsettings file and turn off Avaya environment in the same file.

SET ENABLE_AVAYA_ENVIRONMENT 0
SET SIP_CONTROLLER_LIST "192.168.18.100:5060;transport=tcp"

Voicemail also working fine, But MWI is not working. But i don't know why it is not working with UDP.

[astroshark]

Mvenkatesh431 it would be really good if you can share your setup and settings.
For the life of me i cannot get this working in asterisk
Funny though it all works fine out of the box in 3cx..

i cant get the phones to register at all.