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J100s No Dial Tone or Ring Back

sheldoom

Vendor
Dec 12, 2006
365
US
Installing a new system SE R11.1.3.1 on virtual machine. We are migrating J139s and 9600s from CM to IP Office. Phones are not getting dial tone when going offhook and not hearing ring back on internal calls. Phones are working fine and can make and receive calls, just no dial tone or ring back. Local dial tone is enabled.

Thank you!!
 
Yes, both local dial tone and secondary dial tone are enabled
 
Sorry not what I asked, where did you enable local tones?
 
I did find that for one phone that does get dial tone, when I look at the phone is SSA is has Media Stream - Best Effort, where phones that don't have dial tone have Media Stream - RTP
 
Where did you enable local tones

What the network connection between the phone and system?

What are you using to provision the phones? DES? DHCP? or Manul?

I assume you factory reset the phones?

Are they running the right version?
 
Local dial tone is enabled under System > Telephone > Tone and Music
This is a multi plant manufacturing operation so connectivity varies between plants and I don't have that info
Phones are provisioned using customer's DHCP servers and some plants have there own DHCP server, but we checked that all scopes are good.
All phones were factory reset.
Firmware is 4.0.10.3.2

After further checking it appears the J129s do have dial tone, but the J139s and 9600s do not.
 
Are the phones displaying as (SIP Feature) (Is SSA or Monitor) What is the product ID for them?

1731024811987.png
 
J139s display SIP Feature in SSA

Telephone Type: Avaya J139 (SIP Feature)
User-agent SIP header: Avaya J139 IP Phone 4.0.10.3.2 c81feaa2ee45
Part number is - 700513916 for the new J139s, some of them are reused from the CM
 
Unless you have done something odd in the config then it would be the networks causing this.

I think it would be good to understand how the network is setup, might be network related.

Have you got Keepalives set? Under VOIP on the LAN interface for the phones.

This is my default Keepalive settings.

1731032012797.png
 
Menniss, mine is set the same as they are using AT&T SIP trunks and this is now the Application Notes say to configure keepalives.

1731070625245.png
 
check if there is a user shortcode that starts dialing something when going offhook.

Check in SSA on an extension trace.

If you have an analog port see if you have dialtone on that, just fishing here but ...
 
Installing analog ports today, but they will be on an IP500 SE expansion

I added Local Tones on the H.323 Extensions, but this didn't help. I'm wondering if I should tweak the security System > VoIP > VoIP Security. I set this up per the Application Notes for A&T SIP trunks with BE.
 
I tried disabling Media Security and the extension level as to not impact the SIP trunks and this didn't help, still no dial tone on J139s and 9600s.
 
check this out on your system tab>> telephony .
1731237095056.png

if this is already configured. so try to change the locale from the system tab >> system. as follow.

1731237200911.png
 
Is this a know workaround for this issue? Something you've done before to get this working?
 
yes, i did this for a customer and worked fine with no issues.
you should try nothing to lose.
if this didn't work, you can change the config back.
 
Figured it out. Per the Application notes for AT&T SIP with BE the RTP ports need to be changed. I changed it back to default and dial tone work now and the SIP trunks are also working. The AT&T engineer told me that they access any RTP port from 6000 - 65000.

1731344147250.png
 
That’s bullshit in my opinion. Every end of a SIP connection tells the other end at what port it expects to receive. Even AT&T should be able to handle thisto…
 

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