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Spectralink 8440 Implementation

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TN94z

Technical User
Apr 15, 2010
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We are in the process of implementing the 8440 sets for testing. I have searched the 3 other closed threads and actually used the documents to help setup the node and pbx portions. My problem comes in with getting the 8440 on the network to grab an IP, so I can browse to it and set it up.

I downloaded some guides from Spectralink and just want to make sure I am looking at the right process. From reading the guides, it seems that I must edit some .cfg files with an xml editor to get the phones to connect to the wlan and download it's config files from a provisioning server. is this correct? The reason I ask is because a fellow technician setup some SIP phones a while back but could not remember having to go through all of that. The site he worked on has since moved to Cisco so he no longer has any of their documentation to look back through.

Any help would be greatly appreciated. I am currently at the point of configuring the phones.
 
I have made some progress here. After speaking with Spectralink, the documents are misleading. So I now have the phone on the network and configured via the gui interface, but can't get the line to register. I am working on that now, but wanted to update the post since I can't delete.
 
I have gotten a little bit further. I have the SIP 8440 handset receiving calls but I still get a fast busy when I try to call out. I'll be working on that issue tomorrow.
 
Hi,

Google for this doc, which will help you on CS1K.

Application Notes for Configuring Spectralink 84-Series SIP Telephone Version 4.7.0 with Avaya Communication Server 1000 Release 7.6 - Issue 1.0

-- Sample TN

On phones a must:
UXTY SIPL
MCCL Yes (always)
SIPN 0 0=3rd party; 1=Nortel SIP phone
SIP3 1 1=3rd party; 0=Avaya RED phone
SIPU XXXX (make same as extension)
NDID 1250 (Node ID)
Zone 10
SCPW (make same as extension)
Key 0 SCR XXXX ENTER
Key 1 HOT U (Press enter and it will add the UAPR)+(ext) CLID
----------------------------------------------------------------------------------------------
REQ: new
TYPE:----- uext
TN 164-X-X-X
CDEN 8D
CTYP XDLC
CUST 0
UXTY-----SIPL
MCCL-----YES
SIPN------ 0
SIP3-------1
FMCL 0
TLSV 0
SIPU-----Extension
NDID -----1250 Node ID
SUPR NO
UXID
NUID
NHTN
ZONE -----10
CUR_ZONE 00010
MRT
ERL 0
ECL 0
FDN
TGAR----- 0
LDN NO
NCOS------5
SGRP 0
RNPG 0
SCI 0
SSU
LNRS 16
XLST
SCPW ------Extension
CLS CTD FBD WTA LPR MTD FNA HTD TDD HFA CRPD
MWA LMPN RMMD AAD IMD XHD IRD NID OLD VCE DRG1
POD SLKD CCSD SWD LNA CNDA
CFTD SFD MRD DDV CNID CDCA MSID DAPA BFED RCBD
ICDD CDMD LLCN MCTD CLBD AUTU
GPUD DPUD DNDD CFXD ARHD FITD CLTD ASCD
CPFA CPTA ABDD CFHD FICD NAID BUZZ AGRD MOAD
AHA DDGA NAMA
DRDD EXR0
USMD USRD ULAD RTDD RBDD RBHD PGND FLXD FTTC MCBN
VOLA VOUD CDMR PRED RECD MCDD T87D SBMD ELMD
MSNV FRA PKCH MWTD DVLD CROD ELCD VMSA
CPND_LANG ENG
HUNT
PLEV 02
PUID
UPWD
DANI NO
AST
IAPG 0
AACS NO
ITNA NO
DGRP
MLWU_LANG 0
MLNG ENG
DNDR 0
KEY 0 SCR XXXX enter
CPND
NAME Test
XPLN 5
DISPLAY_FMT FIRST,LAST
Key 01 HOT U enter (222XXXX MARP 0)
Key 02 MSB  MSB key is used for Make Set busy feature on SIP endpoint
Key 03 CWT CWT key is used for Call Waiting feature on SIP endpoint
04
05
06
07
08
09
10
11
12
13
14
15
16
17 TRN
18 AO6
19 CFW 16
20 RGA
21 PRK
22 RNP
23
24 PRS
25 CHG
26 CPN
27
28
29
30
31
 
Yeah, that is the document I used during configuration.
 
So, we got the calls coming in by changing from TCP to UDP. Now I have to look into why my SIP handset puts the call on hold whenever the called number goes to their voicemail...
 
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