GrandStream HT-701 ATA ignoring pulses

KarolRWT · January 29, 2025, 03:14:32 PM

Quote from: 5415551212 on January 29, 2025, 02:40:03 PMWhen you say your checking the log is that on the grandstream?
Can you post the output of that log?

There is no log on my Grandstream, I was posting results of log from Asterisk, AKA the VoIP server. Everything is as it was, either grandstream is broken, or its software related....

5415551212 · January 29, 2025, 05:10:57 PM

Quote from: KarolRWT on January 29, 2025, 03:14:32 PMThere is no log on my Grandstream, I was posting results of log from Asterisk, AKA the VoIP server. Everything is as it was, either grandstream is broken, or its software related....

I see so we don't know for sure if the problem is in the Grandstream interpreting the pulses or how its sending SIP data to the asterisk or the asterisk server config.
How many FXS ports does the grandstream have?

5415551212 · January 29, 2025, 05:25:04 PM

I took a quick peel at the manual
http://www.grandstream.com/hubfs/Product_Documentation/ht70x_usermanual_english.pdf

And you can set 'Syslog Level' to info and it will log all calls. (under Advanced Settings)
Check the grandstream log file for the incoming digits.

dsk · January 30, 2025, 02:04:48 AM

This is strange, my HT702 has accepted all of my phones. (including my BRATEK 271) I know that pulse-tone converters has trouble with some phones. Phones who short the line completely when dial is out of rest position does not have those problems.
1:Have you tested another rotary phone on the ATA?
2:Have you turned off the hook-flash?

5415551212 · January 30, 2025, 12:34:28 PM

Does the Grandsteram send the correct digits to Asterisk when DTMF is used?

KarolRWT · January 30, 2025, 02:01:58 PM

Quote from: 5415551212 on January 29, 2025, 05:10:57 PMI see so we don't know for sure if the problem is in the Grandstream interpreting the pulses or how its sending SIP data to the asterisk or the asterisk server config.
How many FXS ports does the grandstream have?

Exactly 1 FXS port

Quote from: 5415551212 on January 29, 2025, 05:25:04 PMI took a quick peel at the manual
http://www.grandstream.com/hubfs/Product_Documentation/ht70x_usermanual_english.pdf

And you can set 'Syslog Level' to info and it will log all calls. (under Advanced Settings)
Check the grandstream log file for the incoming digits.

I'll check and send the log.

Quote from: dsk on January 30, 2025, 02:04:48 AMThis is strange, my HT702 has accepted all of my phones. (including my BRATEK 271) I know that pulse-tone converters has trouble with some phones. Phones who short the line completely when dial is out of rest position does not have those problems.
1:Have you tested another rotary phone on the ATA?
2:Have you turned off the hook-flash?

1 - I did not test another rotary phone but modern one with pulse dialing setting.
2 - yes.

Quote from: 5415551212 on January 30, 2025, 12:34:28 PMDoes the Grandsteram send the correct digits to Asterisk when DTMF is used?

Yes it does send the correct digits while using DTMF.

KarolRWT · January 30, 2025, 02:45:46 PM

Here is log of calling numbers: 0000 and 1234567890, the rest of the log is logged boot up so I won/'t be sending it here.

QuoteHT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_rcvreq: Received SIP request NOTIFY
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SigCtrl::processSigMwi, Voice mail for acct 0: 0/0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 2
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2] NTP :Days:45685 Seconds:70684.403 Elapsed:139090.0 sec Stall: 15.9 sec Skew:190651.5 sec Dispersion: 656.1 usec Frequency: 0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:ATACtrl::processPhoneOffHook on port 0:0, status = CALL_IDLE/CALL_IDLE, reg'd:1, allow calls w/o reg:1 ,sigReferred:0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_rcvreq: Received SIP request OPTIONS
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 2
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:Call(1)::Call, Creating Call object 1 at port 0:0, caller 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:Call(1)::run, Dialing 2222
HT701 [00:0B:82:58:62:18] [1.0.8.2]:RTP::QoS: Layer 3 DSCP for RTP set to 46
HT701 [00:0B:82:58:62:18] [1.0.8.2]:RTP::QoS: Layer 3 DSCP for RTCP set to 46
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_snd_message: Transaction 7 , state=0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 2
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPClientTransaction::sendRequest: Request 7 is sent
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_rcv4xx: Received 401 response for transaction 7(INVITE)
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 2
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 2
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPTransaction::waitForResponse: Request 7 got status code 401
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_snd_message: Transaction 8 , state=0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 2
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_rcv1xx: Received 100 response for transaction 8 (INVITE)
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 2
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_rcv1xx: Received 183 response for transaction 8 (INVITE)
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPTransaction::waitForResponse: Request 8 got status code 100
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active call dialogs: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPTransaction::waitForResponse: Request 8 got status code 183
HT701 [00:0B:82:58:62:18] [1.0.8.2]:ATACtrl::processSigEarlyMedia on port 0:0, status = CALL_DIALED/CALL_IDLE
HT701 [00:0B:82:58:62:18] [1.0.8.2]:RTP::setSDP on port 0:0, current sdp: (nil), new sdp: 0x132810
HT701 [00:0B:82:58:62:18] [1.0.8.2]:EarlyMedia::run, Start Early Media on port 0:0, disableLEC 1, hasInfo =0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:GSDSP::stop RTP on port 0:0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:GSDSP::RTP stopped on port 0:0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:RTP::start on port 0:0, SRTP status NO_SRTP
HT701 [00:0B:82:58:62:18] [1.0.8.2]:GSDSP::startRTP at port 0:0, RTP encoder 0@20, VAD 0, disableLEC:1 ptevt:101
HT701 [00:0B:82:58:62:18] [1.0.8.2]:GSDSP::startRTP, JB min =120, JB max =1000, silence =0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:GSDSP::startRTP, Start RTP on port 0:0, socket 26, sndrcv 3, remote 83.168.69.15:10970
HT701 [00:0B:82:58:62:18] [1.0.8.2]:GSDSP::startRTP started on port 0:0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_rcv5xx: Received 503 response for transaction 8(INVITE)

syslogs time : 01-30 20:46:00
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active call dialogs: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:Call(2)::run, Dialing 1222222222
HT701 [00:0B:82:58:62:18] [1.0.8.2]:RTP::QoS: Layer 3 DSCP for RTP set to 46
HT701 [00:0B:82:58:62:18] [1.0.8.2]:RTP::QoS: Layer 3 DSCP for RTCP set to 46
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Deleting call dialog (1)
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPDialog::~SIPDialog: Cleaning in-dialog in-transaction
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPDialog(1)::~SIPDialog: Cleaning in-dialog out-transaction
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPDialog(1)::~SIPDialog, Deleting SDP in the dialog
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 2
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_snd_message: Transaction 9 , state=0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 2
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPClientTransaction::sendRequest: Request 9 is sent
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_rcv4xx: Received 401 response for transaction 9(INVITE)
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 2
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 2
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPTransaction::waitForResponse: Request 9 got status code 401
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_snd_message: Transaction 10 , state=0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 2
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_rcv1xx: Received 100 response for transaction 10 (INVITE)
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 2
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_rcv1xx: Received 183 response for transaction 10 (INVITE)
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPTransaction::waitForResponse: Request 10 got status code 100
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active call dialogs: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:ATACtrl::processSigEarlyMedia on port 0:0, status = CALL_DIALED/CALL_IDLE
HT701 [00:0B:82:58:62:18] [1.0.8.2]:RTP::setSDP on port 0:0, current sdp: (nil), new sdp: 0x133f40
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPTransaction::waitForResponse: Request 10 got status code 183
HT701 [00:0B:82:58:62:18] [1.0.8.2]:EarlyMedia::run, Start Early Media on port 0:0, disableLEC 1, hasInfo =0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:GSDSP::stop RTP on port 0:0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:GSDSP::RTP stopped on port 0:0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:RTP::start on port 0:0, SRTP status NO_SRTP
HT701 [00:0B:82:58:62:18] [1.0.8.2]:GSDSP::startRTP at port 0:0, RTP encoder 0@20, VAD 0, disableLEC:1 ptevt:101
HT701 [00:0B:82:58:62:18] [1.0.8.2]:GSDSP::startRTP, JB min =120, JB max =1000, silence =0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:GSDSP::startRTP, Start RTP on port 0:0, socket 27, sndrcv 3, remote 83.168.69.15:16744
HT701 [00:0B:82:58:62:18] [1.0.8.2]:GSDSP::startRTP started on port 0:0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_rcv5xx: Received 503 response for transaction 10(INVITE)
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active call dialogs: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPTransaction::waitForResponse: Request 10 got status code 503
HT701 [00:0B:82:58:62:18] [1.0.8.2]:EarlyMedia::stop, Stop Early Media on port 0:0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active call dialogs: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:GSDSP::stop RTP on port 0:0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:GSDSP::RTP stopped on port 0:0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:ATACtrl::call, cannot make the call, statusCode = 503, chan status = CALL_RINGING, emergency call =0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:ATACtrl::processCallFailed on port 0:0, status = CALL_RINGING/CALL_IDLE stCode:503 canConf:1 ,sigReferred:0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:ATACtrl::processCallFailed Loop current disconnect on port 0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:Call(2)::~Call, Deleting Call object 2 at port 0:0, callCount=0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:ATACtrl::processPhoneOnHook on port 0:0, status = CALL_ENDING/CALL_IDLE ,sigReferred:0
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::cb_rcvreq: Received SIP request OPTIONS
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active call dialogs: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 2
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active call dialogs: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Deleting call dialog (2)
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPDialog::~SIPDialog: Cleaning in-dialog in-transaction
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPDialog(2)::~SIPDialog: Cleaning in-dialog out-transaction
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPDialog(2)::~SIPDialog, Deleting SDP in the dialog
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 1
HT701 [00:0B:82:58:62:18] [1.0.8.2]:SIPStack(0)::run: Active transactions: 1
HT701 [00:0B:82:58:62:18] [1.

dsk · January 30, 2025, 03:58:59 PM

Here are my setup. It that works on callcentric.

KarolRWT · January 30, 2025, 04:32:45 PM

Quote from: dsk on January 30, 2025, 03:58:59 PMHere are my setup. It that works on callcentric.

Thanks! I'll reconfigure my ATA tomorrow (it's 10:30PM) and I'll let you know!

5415551212 · January 30, 2025, 06:35:11 PM

Wow thats really odd, whne you dial 0000 it gets 2222
I wonder if its defective.

KarolRWT · January 31, 2025, 04:00:08 AM

Quote from: 5415551212 on January 30, 2025, 06:35:11 PMWow thats really odd, whne you dial 0000 it gets 2222
I wonder if its defective.

yeah, weird thing is when i dial 1234567890, it dials then 1222222222, so "1" is correct and rest is interpreted as "2".
Or sometimes as "1" in some cases numbers got interpreted as "1" and not "2" like usual in my case, it stopped now but i guess it's just my ATA thing

KarolRWT · February 01, 2025, 03:03:10 PM

Reconfigured the ATA, still the same thing happens...

TelePlay · February 01, 2025, 05:55:56 PM

Quote from: KarolRWT on February 01, 2025, 03:03:10 PMReconfigured the ATA, still the same thing happens...

What does it show if you reversed the dial number order, dial 0 first, then 9 etc.

0987654421

KarolRWT · February 02, 2025, 03:13:14 PM

Quote from: TelePlay on February 01, 2025, 05:55:56 PMWhat does it show if you reversed the dial number order, dial 0 first, then 9 etc.

0987654421

this is what i got:

QuoteHT701 [00:0B:82:58:62:18] [1.0.8.2]:Call(1)::run, Dialing 42222222

basically it skipped 2 and 1... weird

countryman · February 02, 2025, 04:42:25 PM

Since it recognizes DTMF tones correctly, this does not seem to be a software problem but related to the hardware at the "front end", dealing with the line current.
"Bad capacitor" is what anyone who is half educated with vintage electronics (like me) would shout - but it could be anything else as well. If a visual inspection does not reveal anything peculiar... good luck.

News:

GrandStream HT-701 ATA ignoring pulses