GXW4501 First time setup


These is my first time setting one of these up. I am using this to provide SIP service to an old Hitachi PBX that was serviced by a T1. I got the T1 light solid green and have the digital interface set to PRI_NET, T1, Slave, NI2. When I go to the dashboard my SIP trunk is blue, but my digital port 2-24 are all coming up red unavailable. I have tried so many things to get them blue. Can anyone help? Thanks in advance.


So I was able to get the digital trunk up. I found an old piece of paper that had some of the trunk settings. It was using E&M… now they are all blue, however I still can’t get calls out. I have switched from master to slave, ami or b8zs, and all with no luck. The PBX now seems like it trys to send out through the trunk but gives a “invalid” message. Before it would fail immediately after you dialed more than three digits after choosing the 9 trunk group. Now when you press nine, it gives you a dialtone, lets you enter the full number, pauses for like 3 or 4 seconds and then gives the “invalid” messsage.


Did you set incoming and outgoing routes ?
It is PBX in that matter and you need set up trunk-trunk calls.


I did setup inbound and outbound routes. I was able to get the cisco config file from the existing gateway. It seems to be using E&M Wink not E&M Immediate. I don’t know if that is the issue. Bere is the configuration

fax protocol pass-through g711ulaw
modem passthrough nse codec g711ulaw
bind control source-interface GigabitEthernet0/0/0.300
bind media source-interface GigabitEthernet0/0/0.300
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
voice class sip-profiles 101
request ANY sip-header From modify “sip:(.*@.*)()” “<sip:\1;otg=2007026778-1\2”
response ANY sip-header From modify “sip:(.*@.*)()” “<sip:\1;otg=2007026778-1\2”
request BYE sip-header Reason modify “(.)cause=3(.)” “\1cause=16\2”
voice iec syslog
voice cause-code
voice statistics type iec
voice statistics time-range since-reset
voice translation-rule 11
rule 1 /.*/ /4073513900/
voice translation-profile CAS
translate calling 11
service dsapp
param callWaiting FALSE
param callConference TRUE
param callTransfer TRUE
service default dsapp
voice-card 0/1
no watchdog
voice-card 1/0
no watchdog
license udi pid ISR4331/K9 sn FLM262509H5
license boot level uck9
log config
logging enable
logging size 200
path bootflash:CONFIG-BACKUP-
maximum 3
memory free low-watermark processor 67522
diagnostic bootup level minimal

mode none
controller T1 0/1/0
framing esf
clock source network
linecode b8zs
cablelength short 110
ds0-group 0 timeslots 1-24 type e&m-wink-start
cas-custom 0
trunk-group CAS


Setting is related to T1 establish, but not to inbound/outband routes.
If T1 is stable then problem is with routes. Settings from cisco is useless in that case


Attached is my route setup


You sure that provider sent you this ?


What do you mean? My sip provider? I have two trunks that I can use, one is Flowroute and the other is to my FreePBX as an extension. I found this config from Googling and YouTube for outbound and inbound routes


So you’re trying to keep the old Hitachi running with its phones, make calls out on the PRI which you have connected between the PRI card on the Hitachi and the 4501, then SIP Trunk from the 4501 to FreePBX which has a sip extension on it? (presumably to test the link, with the intention of setting up sip trunks to the FreePBX and route calls through that after you have the link working?)


That’s one way… or I don’t mind even just connecting to my sip provider flowroute and just provide dialing. This pbx runs the guest rooms. The front desk and office are running on the freepbx separately now. If I can bridge the two systems then greate, but the number one priority is that the guest can make calls.


without seeing your programming (the route programming you have doesnt show enough info), the level of help will also be limited. I think @Marcin is right, with the T1 lights looking good, the issue is routing.
Have you run a T1 troubleshooting test then make 1 call (pick a direction and start with that) - Hitachi to 4501. That will be inbound route on the 4501 - I would start with sending EVERY call to something you know works. Run the wireshark on the T1 and look at the signalling - its likely the route isnt set to receive precisely what the hitachi is sending for digits. Not being there to see its a guess on my part (but i’ve had to experiment with stuff like this between Panasonic and UCM, Samsung OfficeServ and UCM), but looking at the SS7 logs should give you some idea of what the hitachi is sending.


which log do I download and how?


Anyone know how to look at the SS7 logs??


Sorry, was a crazy few days at work.

Chose the SS7 signalling trace (or if you’re using a PRI link you can chose that).
Try doing a call test while the test is running, then download the logs.
It will be zipped up but they will be raw text, not a wireshark file.

On the PRI.log (txt file), you want to look for lines like “message type” - these are feedback messages that describe where the call is in its process.
2022-07-10, 17:20:30:392:148 < Message Type: SETUP (5) is an example of an incoming call that is just starting to be received by the PBX for instance. Scroll down to get more info about the call…

2022-07-10, 17:20:30:392:431 < Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
2022-07-10, 17:20:30:392:486 < User information layer 1: u-Law (34)
2022-07-10, 17:20:30:392:538 < [18 03 a9 83 81]
2022-07-10, 17:20:30:392:591 < Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0
2022-07-10, 17:20:30:392:637 < ChanSel: As indicated in following octets
2022-07-10, 17:20:30:392:687 < Ext: 1 Coding: 0 Number Specified Channel Type: 3
2022-07-10, 17:20:30:392:734 < Ext: 1 Channel: 1 Type: CPE]
2022-07-10, 17:20:30:392:793 < [6c 0c 21 83 37 30 35 37 38 39 38 39 39 30]
2022-07-10, 17:20:30:392:846 < Calling Party Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
2022-07-10, 17:20:30:392:898 < Presentation: Presentation allowed, Network provided (3) ‘7057898000’ ](inbound caller ID shows here)
2022-07-10, 17:20:30:392:953 < [70 05 c1 33 31 32 30]
2022-07-10, 17:20:30:393:005 < Called Party Number (len= 7) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) ‘3120’ ] (shows inbound DID digits sent from the central office - this is your inbound DID number for translation in the PBX for inbound route)
2022-07-10, 17:20:30:393:055 – Got ACK for N(S)=27 to (but not including) N(S)=27
2022-07-10, 17:20:30:393:101 – T200 requested to stop when not started
2022-07-10, 17:20:30:393:149 T203 requested to start without stopping first
2022-07-10, 17:20:30:393:195 – Starting T203 timer
2022-07-10, 17:20:30:393:243 – Making new call for cref 7054
2022-07-10, 17:20:30:393:309 Received message for call 0x411b7b78 on link 0xb91f4c TEI/SAPI 0/0
2022-07-10, 17:20:30:393:366 – Processing Q.931 Call Setup (codec choice - PRI is Q931, and the call setup is progressing)

For SS7, the log is a wireshark pcap. I’ve been successful in troubleshooting PRI using the pri.log and havent needed the ss7 logs.

As far as my settings for the link, its set up as a T1, with 23 channels:

I dont know how the Hitachi would be set up, but i’m guessing you had it connected to the central office before? One of the sides will need to be set as a master and the other a slave. Is that how you’ve got it set up?


I got the pri up because I found an old sheet with the settings from the installer in the 90s. It is E&M Wink. So under advances there isn’t all of the options like yours just wink interval. It comes up, now it’s just figuring out what digits the pbx is sending. Thanks for how to look at the logs. I will have a tech onsite tomorrow. Hopefully I can get it up.