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.