During the conversation I hear looong DTMF long beep in random time

Im using Asterisk v13.19 with chain_dongle…
The Dongle modem is Huawei 1550.
In my dongle.conf the seeting is:

[general]

interval=5			; Number of seconds between trying to connect to devices
;-----------------------------------------------------------------------------------
[defaults]

context=from-gsm			; context for incoming calls
group=0				; calling group
rxgain=0			; increase the incoming volume; may be negative
txgain=0			; increase the outgoint volume; may be negative
autodeletesms=yes		; auto delete incoming sms
resetdongle=yes			; reset dongle during initialization with ATZ command
u2diag=-1			; set ^U2DIAG parameter on device (0 = disable everything except modem function) ; -1 not use ^U2DIAG command
usecallingpres=yes		; use the caller ID presentation or not
callingpres=allowed_passed_screen ; set caller ID presentation		by default use default network settings
disablesms=no			; disable of SMS reading from device when received
			
language=en			; set channel default language
smsaspdu=yes			; if 'yes' send SMS in PDU mode, feature implementation incomplete and we strongly recommend say 'yes'
mindtmfgap=45			; minimal interval from end of previews DTMF from begining of next in ms
mindtmfduration=80		; minimal DTMF tone duration in ms
mindtmfinterval=200		; minimal interval between ends of DTMF of same digits in ms

callwaiting=auto		; if 'yes' allow incoming calls waiting; by default use network settings
				; if 'no' waiting calls just ignored
disable=no			; OBSOLETED by initstate: if 'yes' no load this device and just ignore this section
initstate=start			; specified initial state of device, must be one of 'stop' 'start' 'remote'

dtmf=relax			; control of incoming DTMF detection, possible values:
				;   off	   - off DTMF tones detection, voice data passed to asterisk unaltered
				;              use this value for gateways or if not use DTMF for AVR or inside dialplan
				;   inband - do DTMF tones detection
				;   relax  - like inband but with relaxdtmf option
				;  default is 'relax' by compatibility reason

; dongle required settings
[dongle0]
audio=/dev/DONGLE-3G-MODEM-1		; tty port for audio connection; 	no default value
data=/dev/DONGLE-3G-MODEM-2			; tty port for AT commands; 		no default value
context=from-gsm
 

It happening that during the call in random secund of call I hear loong beep and after that just beep, till I do not hang up or start pressing the dialpad.
I started asterisk with full log, and here is the part of log:

<------------->
[Feb 25 12:34:21] DEBUG[14899] chan_sip.c:  Header  0 [  0]: 
[Feb 25 12:34:24] DEBUG[14924] at_read.c: [dongle0] receive 12 byte, used 12, free 2036, read 0, write 12
[Feb 25 12:34:24] DEBUG[14924] at_read.c: [dongle0] [
^RSSI:12
]
[Feb 25 12:34:27] DEBUG[21156][C-00000013] res_rtp_asterisk.c: Got RTCP report of 80 bytes from 2.xxx.xxx.xxx:44383
[Feb 25 12:34:28] DEBUG[14924] at_read.c: [dongle0] receive 27 byte, used 27, free 2021, read 0, write 27
[Feb 25 12:34:28] DEBUG[14924] at_read.c: [dongle0] [
^BOOT:35647360,0,0,0,87
]
[Feb 25 12:34:29] DTMF[21664][C-00000013] channel.c: DTMF begin 'A' received on Dongle/dongle0-0100000012
[Feb 25 12:34:29] DTMF[21664][C-00000013] channel.c: DTMF begin passthrough 'A' on Dongle/dongle0-0100000012
[Feb 25 12:34:29] DEBUG[14863] threadpool.c: Increasing threadpool stasis-core's size by 1
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c: Allocating new SIP dialog for 092c78576a3127e922d07fb5676ab85d@[::1]:36888 - OPTIONS (No RTP)
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c: OBPROXY: Not applying OBproxy to this call
[Feb 25 12:34:29] DEBUG[14899] acl.c: For destination '2.xxx.xxx.xxx', our source address is '192.168.50.111'.
[Feb 25 12:34:29] DEBUG[14899] netsock2.c: Splitting 'mydns.mydomain.com:36888' into...
[Feb 25 12:34:29] DEBUG[14899] netsock2.c: ...host 'mydns.mydomain.com' and port '36888'.
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c: Target address 2.xxx.xxx.xxx:52855 is not local, substituting externaddr
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c: Setting AST_TRANSPORT_UDP with address 95.xxx.xxx.xxx:36888
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c: SIP call-id changed from '092c78576a3127e922d07fb5676ab85d@[::1]:36888' to '06a1578f677b0293714cbdcc72dab3f7@95.xxx.xxx.xxx:36888'
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c: Initializing initreq for method OPTIONS - callid 06a1578f677b0293714cbdcc72dab3f7@95.xxx.xxx.xxx:36888
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  0 [ 91]: OPTIONS sip:user_myusername@2.xxx.xxx.xxx:1026;transport=UDP;rinstance=63c657500959a5a1 SIP/2.0
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  1 [ 65]: Via: SIP/2.0/UDP 95.xxx.xxx.xxx:36888;branch=z9hG4bK226a3906;rport
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  2 [ 16]: Max-Forwards: 70
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  3 [ 66]: From: "asterisk" <sip:asterisk@95.xxx.xxx.xxx:36888>;tag=as5812b742
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  4 [ 81]: To: <sip:user_myusername@2.xxx.xxx.xxx:1026;transport=UDP;rinstance=63c657500959a5a1>
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  5 [ 43]: Contact: <sip:asterisk@95.xxx.xxx.xxx:36888>
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  6 [ 61]: Call-ID: 06a1578f677b0293714cbdcc72dab3f7@95.xxx.xxx.xxx:36888
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  7 [ 17]: CSeq: 102 OPTIONS
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  8 [ 32]: User-Agent: Asterisk PBX 13.19.0
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  9 [ 35]: Date: Sun, 25 Feb 2018 11:34:29 GMT
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header 11 [ 26]: Supported: replaces, timer
[Feb 25 12:34:29] VERBOSE[14899] chan_sip.c: Reliably Transmitting (NAT) to 2.xxx.xxx.xxx:52855:
OPTIONS sip:user_myusername@2.xxx.xxx.xxx:1026;transport=UDP;rinstance=63c657500959a5a1 SIP/2.0
Via: SIP/2.0/UDP 95.xxx.xxx.xxx:36888;branch=z9hG4bK226a3906;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@95.xxx.xxx.xxx:36888>;tag=as5812b742
To: <sip:user_myusername@2.xxx.xxx.xxx:1026;transport=UDP;rinstance=63c657500959a5a1>
Contact: <sip:asterisk@95.xxx.xxx.xxx:36888>
Call-ID: 06a1578f677b0293714cbdcc72dab3f7@95.xxx.xxx.xxx:36888
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.19.0
Date: Sun, 25 Feb 2018 11:34:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id  #16
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 2.xxx.xxx.xxx:52855
**[Feb 25 12:34:29] DEBUG[21664][C-00000013] channel.c: [dongle0] DTMF char A ignored min gap 45 > 20**
**[Feb 25 12:34:29] VERBOSE[14899] chan_sip.c: **
**<--- SIP read from UDP:2.xxx.xxx.xxx:52855 --->**
SIP/2.0 200 OK
Via: SIP/2.0/UDP 95.xxx.xxx.xxx:36888;branch=z9hG4bK226a3906;rport=36888
Contact: <sip:192.168.1.134:52855>
To: <sip:user_myusername@2.xxx.xxx.xxx:1026;transport=UDP;rinstance=63c657500959a5a1>;tag=7e79ed00
From: "asterisk" <sip:asterisk@95.xxx.xxx.xxx:36888>;tag=as5812b742
Call-ID: 06a1578f677b0293714cbdcc72dab3f7@95.xxx.xxx.xxx:36888
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper rv2.8.15
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  0 [ 14]: SIP/2.0 200 OK
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  1 [ 71]: Via: SIP/2.0/UDP 95.xxx.xxx.xxx:36888;branch=z9hG4bK226a3906;rport=36888
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  2 [ 34]: Contact: <sip:192.168.1.134:52855>
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  3 [ 94]: To: <sip:user_myusername@2.xxx.xxx.xxx:1026;transport=UDP;rinstance=63c657500959a5a1>;tag=7e79ed00
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  4 [ 66]: From: "asterisk" <sip:asterisk@95.xxx.xxx.xxx:36888>;tag=as5812b742
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  5 [ 61]: Call-ID: 06a1578f677b0293714cbdcc72dab3f7@95.xxx.xxx.xxx:36888
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  6 [ 17]: CSeq: 102 OPTIONS
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  7 [ 40]: Accept: application/sdp, application/sdp
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  8 [ 19]: Accept-Language: en
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header  9 [ 81]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header 10 [ 90]: Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header 11 [ 27]: User-Agent: Zoiper rv2.8.15
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header 12 [ 34]: Allow-Events: presence, kpml, talk
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c:  Header 13 [ 17]: Content-Length: 0
[Feb 25 12:34:29] VERBOSE[14899] chan_sip.c: --- (14 headers 0 lines) ---
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c: = Looking for  Call ID: 06a1578f677b0293714cbdcc72dab3f7@95.xxx.xxx.xxx:36888 (Checking To) --From tag as5812b742 --To-tag 7e79ed00  
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c: Stopping retransmission on '06a1578f677b0293714cbdcc72dab3f7@95.xxx.xxx.xxx:36888' of Request 102: Match Found
[Feb 25 12:34:29] DEBUG[14899] chan_sip.c: Destroying SIP dialog 06a1578f677b0293714cbdcc72dab3f7@95.xxx.xxx.xxx:36888
[Feb 25 12:34:29] VERBOSE[14899] chan_sip.c: Really destroying SIP dialog '06a1578f677b0293714cbdcc72dab3f7@95.xxx.xxx.xxx:36888' Method: OPTIONS
[Feb 25 12:34:30] DEBUG[14924] at_read.c: [dongle0] receive 12 byte, used 12, free 2036, read 0, write 12
[Feb 25 12:34:30] DEBUG[14924] at_read.c: [dongle0] [
^RSSI:12
]
[Feb 25 12:34:31] DEBUG[14899] chan_sip.c: Auto destroying SIP dialog '3q91Y2TwcBVHf_kTvqM_uA..'
[Feb 25 12:34:31] DEBUG[14899] chan_sip.c: Destroying SIP dialog 3q91Y2TwcBVHf_kTvqM_uA..
[Feb 25 12:34:31] VERBOSE[14899] chan_sip.c: Really destroying SIP dialog '3q91Y2TwcBVHf_kTvqM_uA..' Method: REGISTER
[Feb 25 12:34:33] DEBUG[21156][C-00000013] res_rtp_asterisk.c: Got RTCP report of 80 bytes from 2.xxx.xxx.xxx:44383
[Feb 25 12:34:33] DEBUG[14924] at_read.c: [dongle0] receive 12 byte, used 12, free 2036, read 0, write 12
[Feb 25 12:34:33] DEBUG[14924] at_read.c: [dongle0] [
^RSSI:12
]
[Feb 25 12:34:36] DEBUG[14924] at_read.c: [dongle0] receive 12 byte, used 12, free 2036, read 0, write 12
[Feb 25 12:34:36] DEBUG[14924] at_read.c: [dongle0] [
^RSSI:12
]
**[Feb 25 12:34:37] DEBUG[21156][C-00000013] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 48 (0), at 2.xxx.xxx.xxx:44382**
**[Feb 25 12:34:37] DTMF[21156][C-00000013] channel.c: DTMF begin '0' received on SIP/user_myusername-0000000b**
**[Feb 25 12:34:37] DTMF[21156][C-00000013] channel.c: DTMF begin passthrough '0' on SIP/user_myusername-0000000b**
**[Feb 25 12:34:37] DEBUG[21664][C-00000013] at_queue.c: [dongle0] insert task with 1 commands begin with 'AT^DTMF' expected response 'OK' after head of queue**
**[Feb 25 12:34:37] DEBUG[21664][C-00000013] at_queue.c: [dongle0] write command 'AT^DTMF' expected response 'OK' length 12**
**[Feb 25 12:34:37] DEBUG[21664][C-00000013] at_queue.c: [dongle0] [AT^DTMF=1,0**
**]**
**[Feb 25 12:34:37] DEBUG[21664][C-00000013] channel.c: [dongle0] Send DTMF 0**
**[Feb 25 12:34:37] DTMF[21664][C-00000013] channel.c: DTMF begin '0' received on Dongle/dongle0-0100000012**
**[Feb 25 12:34:37] DTMF[21664][C-00000013] channel.c: DTMF begin passthrough '0' on Dongle/dongle0-0100000012**
**[Feb 25 12:34:37] DEBUG[21156][C-00000013] res_rtp_asterisk.c: Creating END DTMF Frame: 48 (0), at 2.xxx.xxx.xxx:44382**
**[Feb 25 12:34:37] DTMF[21156][C-00000013] channel.c: DTMF end '0' received on SIP/user_myusername-0000000b, duration 120 ms**
**[Feb 25 12:34:37] DTMF[21156][C-00000013] channel.c: DTMF end accepted with begin '0' on SIP/user_myusername-0000000b**
**[Feb 25 12:34:37] DTMF[21156][C-00000013] channel.c: DTMF end passthrough '0' on SIP/user_myusername-0000000b**
**[Feb 25 12:34:37] DEBUG[21664][C-00000013] channel.c: [dongle0] Got DTMF char 0**
**[Feb 25 12:34:37] DTMF[21664][C-00000013] channel.c: DTMF end '0' received on Dongle/dongle0-0100000012, duration 140 ms**
**[Feb 25 12:34:37] DTMF[21664][C-00000013] channel.c: DTMF end accepted with begin '0' on Dongle/dongle0-0100000012**
**[Feb 25 12:34:37] DTMF[21664][C-00000013] channel.c: DTMF end passthrough '0' on Dongle/dongle0-0100000012**
**[Feb 25 12:34:37] DEBUG[14924] at_read.c: [dongle0] receive 6 byte, used 6, free 2042, read 0, write 6**
**[Feb 25 12:34:37] DEBUG[14924] at_read.c: [dongle0] [**
**OK**
**]**
**[Feb 25 12:34:37] DEBUG[14924] at_response.c: [dongle0] DTMF sent successfully for call idx 1**
**[Feb 25 12:34:37] DEBUG[14924] at_queue.c: [dongle0] remove command 'AT^DTMF' expected response 'OK' real 'OK' cmd 1/1 flags 0x00 from queue**
**[Feb 25 12:34:37] DEBUG[14924] at_queue.c: [dongle0] remove task with 1 command(s) begin with 'AT^DTMF' expected response 'OK' from queue**
**[Feb 25 12:34:38] DEBUG[21156][C-00000013] res_rtp_asterisk.c: Got RTCP report of 80 bytes from 2.xxx.xxx.xxx:44383**
[Feb 25 12:34:39] DEBUG[14924] at_read.c: [dongle0] receive 12 byte, used 12, free 2036, read 0, write 12
[Feb 25 12:34:39] DEBUG[14924] at_read.c: [dongle0] [
^RSSI:12
]
[Feb 25 12:34:42] DEBUG[14924] at_read.c: [dongle0] receive 12 byte, used 12, free 2036, read 0, write 12
[Feb 25 12:34:42] DEBUG[14924] at_read.c: [dongle0] [
^RSSI:12
]

I see these two line that starting to sending the DTMF:

[Feb 25 12:34:29] DTMF[21664][C-00000013] channel.c: DTMF begin 'A' received on Dongle/dongle0-0100000012
[Feb 25 12:34:29] DTMF[21664][C-00000013] channel.c: DTMF begin passthrough 'A' on Dongle/dongle0-0100000012

Thx for any kind of help.

Anyone with any idea?

Could be talk-off, unless DTMF is in band end to end.

A problem with any system that decodes DTMF and regenerates it (the preferred way with SIP, and essential with low bit rate codecs), the tone decoder can think that speech sounds are digits, encode them as such and have the digit regenerated on the outgoing side, rather than the speech passed through.

Thx for replay.
I understand the logic, but how can I prevent this?
Is it any posibility in config?

Assuming SIP (and with details based on chan_sip).

Ensure that you are using a simple codec, like ulaw or alaw

Set dtmfmode to inband on all legs.

Make sure that all other SIP entities (phones, ITSPs, etc.) do the same.

Hmm thx…
Im using gsm codec because of bitrate.
It consumping ~30kbps up adn down till ulaw or alaw aprox 80kbps up and down.
Do you think that this can be the proboblem?
Btw I tried in dongle.conf to put:
dtmf=inband
Till now I usedwith option:
dtmf=relax

In sip.conf I have:

[general]
dtmfmode=inband
...
...
[username]
context=from-sip-phone
host = dynamic
type = friend
;dtmfmode=rfc2833 
...
...

In user section I disable dtmfmode=rfc2833 row to see would it help.

Any other suggestion?

You could try and break DTMF on the incoming side, by deliberately misconfiguring but you are likely to get silences instead of tones, if this is talk off

And how can I do that?