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Anonymous
Not applicable

break dial tone

We inherited a VoIP deployment with 30+ 908 gateways that have repetitive issues. Currently, at least once a week a gateway has ports on it that are unable to break dial tone. Our current solution is a weekly reboot of every gateway. Firmware has been updated as well as changing codecs. In house wiring has been verified.

Labels (3)
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8 Replies
jayh
Honored Contributor
Honored Contributor

Re: break dial tone

I'm assuming that you're referring to analog FXS ports, if not please advise.

Is it possible that some of the connected devices are configured for pulse dialing? The TA908 series doesn't support it, DTMF only.

Re: break dial tone

If you are still looking for an answer to this, I had the same issue on TA 924s running 12.3.4 firmware. I downgraded mine to R12.3.1.E and the issue never occurred again. At the time of downgrading, Adtran support had stated the issue was present in 12.3.4 and would be fixed in a future release to it is possibly fixed by now.

Anonymous
Not applicable

Re: break dial tone

Have a client this just happened on today. 924e (3rd Gen) running version R13.2.3.E. All the configured FXS ports could not break dialtone when they started to dial. Needed to reboot to resolve.

Re: break dial tone

I opened a case RQST00004992334 with support and they are investigating it further. It appears to be a bug but has not been identified fully so their dev/eng folks need more debugs prior to the 924s ending up in this state so they can write a fix into upcoming firmware revisions.

tshea
New Contributor II

Re: break dial tone

Was there ever feedback on this? We also experience periodic FXS failure on several of our customers. The analog port won't produce dial tone and inbound calls won't ring on the customer's side yet the debug of the FXS appears normal ringing.

Reboot is usually the only solution and it's typically happening on one port at a time.

tshea
New Contributor II

Re: break dial tone

The incoming calls ring and the term side goes off hook. I hear dead silence and the call does not progress. If I reboot the ADTRAN I know for sure everything will be fine. Then weeks later it will happen again. No audio. The analog ports will like "lock up" and I've never been able to figure out why. Idk what testing to do.

ADTRAN, Inc. OS version R13.11.0.E
Mainline Version: ENM.21.52
P4 Changelist: 671378
Checksum: bf9a080c73a6810574d302196f6329ae47905491dac5e075227a3d54e2e3bd36
Built on: Wed Jun 16 11:24:51 CDT 2021
Hardware version Vesuvius v.193.1.1.2.7-3
Boot ROM version R13.7.0.B1
Built on: Thu Jul 16 22:09:40 CDT 2020
Compatibility Version: 1
Copyright (c) 1999-2021, ADTRAN, Inc.
Platform: Total Access 908 (3rd Gen), part number 4213908F1
Serial number CFG1881593
Flash: 216084480 bytes DRAM: 509648896 bytes

Concord_Pediatrics uptime is 14 weeks, 5 days, 22 hours, 50 minutes, 7 seconds

slot 0, DSP 1
DSP software version: G3.R1.5.0-R
DSP hardware version: Freescale BSC9131
Total channels: 60

System returned to ROM by Soft Reset
Current system image file is "T900G3A-R13-11-0-E.biz"
Primary boot system image file is "T900G3A-R13-11-0-E.biz"
Backup boot system image file is "9213900-2r1309hb.biz"
Primary system configuration file is "startup-config"

----

08:01:52.300 FXS.0/1 Ring state change: Ring Stage 1
08:01:52.300 FXS.0/1 Ringing
08:01:54.301 FXS.0/1 Ring state change: Ring Stage 1
08:01:54.302 FXS.0/1 Active Battery
08:01:54.803 FXS.0/1 Ring state change: Insert CID
08:01:55.392 FXS.0/1 Off Hook
08:01:55.393 FXS.0/1 Active Battery
08:02:10.087 FXS.0/1 Battery Removed
08:02:10.100 FXS.0/1 On Hook
08:02:10.840 FXS.0/1 Idle
08:02:10.841 FXS.0/1 Idle
08:02:10.952 FXS.0/1 Off Hook
08:02:10.953 FXS.0/1 Active Battery

---

2022.01.06 07:53:01 FXS.0/1 Initiating Test.
2022.01.06 07:53:01 FXS.0/1 Line Test in progress...
2022.01.06 07:53:10 FXS.0/1
2022.01.06 07:53:10 FXS.0/1 HPT Pass Loop Impedance
2022.01.06 07:53:10 FXS.0/1 FEMF Pass Tip-Ring 243.25 Ohms
2022.01.06 07:53:10 FXS.0/1 RFT Fail Tip-Ground 3718067.50 Ohms
2022.01.06 07:53:10 FXS.0/1 Ringer Fail Ring-Ground 4364147.00 Ohms
2022.01.06 07:53:10 FXS.0/1 ROH Fail REN 0.00 REN
2022.01.06 07:53:10 FXS.0/1
2022.01.06 07:53:10 FXS.0/1 Foreign Voltage
2022.01.06 07:53:10 FXS.0/1 Tip-Ground (ac) 0.00 Volts
2022.01.06 07:53:10 FXS.0/1 Ring-Ground (ac) 0.00 Volts
2022.01.06 07:53:10 FXS.0/1 Tip-Ground (dc) 0.18 Volts
2022.01.06 07:53:10 FXS.0/1 Ring-Ground (dc) 0.17 Volts
2022.01.06 07:53:10 FXS.0/1
2022.01.06 07:53:10 FXS.0/1
2022.01.06 07:53:12 FXS.0/1 Restoring service state.
2022.01.06 07:53:12 FXS.0/1 Test complete.debug int fxs 0/1

bobwolf
New Contributor

Re: break dial tone

Any update to this thread? We have several 924 3rd Gen which the same issue. Users cannot break dialtone. Inbound calls will ring and can be answered, but the outside person cannot hear the inside person. If we reboot, the issue goes away for a while. 

We have tried upgrading everytime a new firmware comes out, but no fix. Current version is 14.1.1.HA

Re: break dial tone

I have confirmed a simliar issue and was able to debug the fxs port. 

Customer using a 900e (3rd Gen) running R14.1.1HAE and was able to reproduce the intermittent issue where the DSP doesn't detect the digits being dialed. 

 

14:45:29.503 TONESERVICES.EVENTS fxs 0/2 - empty - DialTone Generation: Request resource
14:45:29.503 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.1 - DialTone Generation: DSP channel allocated for the resource
14:45:29.504 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.1 - DialTone Generation: constructed
14:45:29.504 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.1 - DialTone Generation: starting
14:45:29.504 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.1 - DialTone Generation: TDM map
14:45:29.504 TONESERVICES.EVENTS fxs 0/2 - empty - Tone Detection: Request resource
14:45:29.505 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.1 - Tone Detection: DSP channel allocated for the resource
14:45:29.505 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.1 - Tone Detection: constructed
14:45:29.505 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.1 - Tone Detection: starting
14:45:29.505 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.1 - Tone Detection: TDM map
14:45:45.536 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.1 - DialTone Generation: stopping
14:45:45.536 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.1 - DialTone Generation: TDM unmap
14:45:45.537 RTP.CHANNEL fxs 0/2 - Dsp 0/1.1 - DialTone Generation: releasing RTP resource
14:45:45.537 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.1 - DialTone Generation: release
14:45:45.810 SB.CCM isMappable:
14:45:45.810 SB.CCM  :  Call Struct 0x0x5162b210 :   Call-ID = 128
14:45:45.810 SB.CCM  :  Org Acct = aaln/2    Dst Acct = aaln/2
14:45:45.811 SB.CCM  :  Org Port ID = MgcpPhone 0/2   Dst Port ID = unknown 0/0
14:45:45.811 SB.CCM isMappable: Call Connection Type is TDM_TO_RTP
14:45:45.811 RTP.CHANNEL allocateForInterface 0/1.1, status = 1, allocatedForInterface = 0
14:45:45.811 SB.CCM isMappable: Reserving RTP Channel 0/1.1
14:45:45.812 SB.CCM isMappable: Creating SDP Offer
14:45:45.813 SB.CCM updateOfferWithEndpointConfig: DTMF(NTE 101), VAD(off), ptime(20)
14:45:45.814 SB.CCM translateOffer: offer codec list: PCMU G729
14:45:45.814 SB.CCM getEndpointCodecList: Applying dynamic codec list to endpoint config
14:45:45.815 SB.CCM getEndpointCodecList: Applying dynamic codec list to endpoint config
14:45:45.815 SB.CCM translateOffer: revised offer codec list: PCMU
14:45:45.815 SB.CCM translateOffer: codec list after answerer: PCMU
14:45:45.816 SB.CCM translateOffer: DTMF signaling: answerer has no restrictions configured, passing offer(NTE 101) through
14:45:45.816 SRTP.SDES NEGOTIATION SrtpNegotiation - SDP before offer filter
14:45:45.816 SRTP.SDES NEGOTIATION     m=audio 0 RTP/AVP 0 101
14:45:45.817 SRTP.SDES NEGOTIATION SrtpNegotiation - SDP after offer filter
14:45:45.817 SRTP.SDES NEGOTIATION     m=audio 0 RTP/AVP 0 101
14:45:45.817 SRTP.SDES NEGOTIATION Invoked pass-through offer translator
14:45:45.817 SRTP.SDES NEGOTIATION SrtpNegotiation - SDP after offer translator
14:45:45.818 SRTP.SDES NEGOTIATION     m=audio 0 RTP/AVP 0 101
14:45:45.818 SB.CCM translateOffer: success
14:45:45.818 MEDIA.MANAGER Allocating media port.
</>

 

..Compared to a working sample where it does detect the digits in the beginning.

12:52:45.645 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.2 - Tone Detection: received digit (1) event
12:52:45.645 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.2 - DialTone Generation: stopping
12:52:45.645 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.2 - DialTone Generation: TDM unmap
12:52:45.645 RTP.CHANNEL fxs 0/2 - Dsp 0/1.2 - DialTone Generation: releasing RTP resource
12:52:45.646 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.2 - DialTone Generation: release
12:52:45.914 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.2 - Tone Detection: received digit (3) event
12:52:46.184 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.2 - Tone Detection: received digit (6) event
12:52:46.454 TONESERVICES.EVENTS fxs 0/2 - Dsp 0/1.2 - Tone Detection: received digit (0) event

 

I'm rolling firmware back to R13.2.2.E since previous thread says it was detected on R13.2.3 and the customer has another location where they are also running 13.2.2 and claim it's 'solid' for years.