The modem drops the call immediately after answering

[merclangrat]

I tried to call using it to a real modem - when the call comes and the opposite side picks up, the call is dropped.
I used PR #1 on Debian 10 KVM virtual machine.

First, I thought that's because there was no audio card - I added a (virtualized) one. But maybe this is still the case.

This is the full log:

13:05:13.276 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
13:05:13.277 sip_endpoint.c .Creating endpoint instance...
13:05:13.277 pjlib .select() I/O Queue created (0x56067802bf50)
13:05:13.278 sip_endpoint.c .Module "mod-msg-print" registered
13:05:13.278 sip_transport.c .Transport manager created.
13:05:13.278 pjsua_core.c .PJSUA state changed: NULL --> CREATED
13:05:13.278 sip_endpoint.c .Module "mod-pjsua-log" registered
13:05:13.278 sip_endpoint.c .Module "mod-tsx-layer" registered
13:05:13.278 sip_endpoint.c .Module "mod-stateful-util" registered
13:05:13.278 sip_endpoint.c .Module "mod-ua" registered
13:05:13.278 sip_endpoint.c .Module "mod-100rel" registered
13:05:13.278 sip_endpoint.c .Module "mod-pjsua" registered
13:05:13.278 sip_endpoint.c .Module "mod-invite" registered
13:05:13.285 alsa_dev.c ..ALSA driver found 0 devices
13:05:13.285 alsa_dev.c ..ALSA initialized
13:05:13.285 pjlib ..select() I/O Queue created (0x56067804f758)
13:05:13.289 sip_endpoint.c .Module "mod-evsub" registered
13:05:13.289 sip_endpoint.c .Module "mod-presence" registered
13:05:13.289 sip_endpoint.c .Module "mod-mwi" registered
13:05:13.289 sip_endpoint.c .Module "mod-refer" registered
13:05:13.289 sip_endpoint.c .Module "mod-pjsua-pres" registered
13:05:13.289 sip_endpoint.c .Module "mod-pjsua-im" registered
13:05:13.289 sip_endpoint.c .Module "mod-pjsua-options" registered
13:05:13.289 pjsua_core.c .1 SIP worker threads created
13:05:13.289 pjsua_core.c .pjsua version 2.11.1 for Linux-4.19.0.27/x86_64/glibc-2.28 initialized
13:05:13.289 pjsua_core.c .PJSUA state changed: CREATED --> INIT
13:05:13.289 pjsua_aud.c Setting null sound device..
13:05:13.289 pjsua_aud.c .Opening null sound device..
13:05:13.289 pjsua_core.c SIP UDP socket reachable at 172.19.39.6:5060
13:05:13.289 udp0x560678054920 SIP UDP transport started, published address is 172.19.39.6:5060
13:05:13.289 pjsua_core.c PJSUA state changed: INIT --> STARTING
<913.289992> call: process: msg 18 --> 0
13:05:13.290 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
13:05:13.290 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
13:05:13.290 pjsua_acc.c Adding account: id=sip:21056@milena.mithril
13:05:13.290 pjsua_acc.c .Account sip:21056@milena.mithril added with id 0
calling sip:21831@milena.mithril
13:05:13.290 pjsua_call.c Making call with acc #0 to sip:21831@milena.mithril
13:05:13.290 pjsua_media.c .Call 0: initializing media..
13:05:13.290 pjsua_media.c ..RTP socket reachable at 172.19.39.6:4000
13:05:13.290 pjsua_media.c ..RTCP socket reachable at 172.19.39.6:4001
13:05:13.290 pjsua_media.c ..Media index 0 selected for audio call 0
13:05:13.292 pjsua_core.c ....TX 938 bytes Request msg INVITE/cseq=21570 (tdta0x560678070b48) to UDP 172.19.39.4:5060:
INVITE sip:21831@milena.mithril SIP/2.0
Via: SIP/2.0/UDP 172.19.39.6:5060;rport;branch=z9hG4bKPj898f657b-cb3a-4e3c-a37c-4dae230b4ac7
Max-Forwards: 70
From: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
To: sip:21831@milena.mithril
Contact: sip:21056@172.19.39.6:5060;ob
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
CSeq: 21570 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 336

v=0
o=- 3969608713 3969608713 IN IP4 172.19.39.6
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 8 0 120
c=IN IP4 172.19.39.6
b=TIAS:64000
a=rtcp:4001 IN IP4 172.19.39.6
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-16
a=ssrc:407200375 cname:14134a3d4d742300

--end msg--
13:05:13.292 d-modem.c .......Call 0 state=CALLING
13:05:13.299 pjsua_core.c .RX 559 bytes Response msg 401/INVITE/cseq=21570 (rdata0x5606780563a8) from UDP 172.19.39.4:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.19.39.6:5060;rport=5060;received=172.19.39.6;branch=z9hG4bKPj898f657b-cb3a-4e3c-a37c-4dae230b4ac7
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
From: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
To: sip:21831@milena.mithril;tag=z9hG4bKPj898f657b-cb3a-4e3c-a37c-4dae230b4ac7
CSeq: 21570 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1760619913/a628da38d98ff5dae29c7d14a8138d2e",opaque="2f8e186073d3b979",algorithm=MD5,qop="auth"
Server: AsteriskPBX
Content-Length: 0

--end msg--
13:05:13.299 pjsua_core.c ..TX 389 bytes Request msg ACK/cseq=21570 (tdta0x7f2b08000bc8) to UDP 172.19.39.4:5060:
ACK sip:21831@milena.mithril SIP/2.0
Via: SIP/2.0/UDP 172.19.39.6:5060;rport;branch=z9hG4bKPj898f657b-cb3a-4e3c-a37c-4dae230b4ac7
Max-Forwards: 70
From: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
To: sip:21831@milena.mithril;tag=z9hG4bKPj898f657b-cb3a-4e3c-a37c-4dae230b4ac7
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
CSeq: 21570 ACK
Content-Length: 0

--end msg--
13:05:13.300 pjsua_core.c .......TX 1234 bytes Request msg INVITE/cseq=21571 (tdta0x560678070b48) to UDP 172.19.39.4:5060:
INVITE sip:21831@milena.mithril SIP/2.0
Via: SIP/2.0/UDP 172.19.39.6:5060;rport;branch=z9hG4bKPj34145732-969b-461a-a255-518c45724db0
Max-Forwards: 70
From: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
To: sip:21831@milena.mithril
Contact: sip:21056@172.19.39.6:5060;ob
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
CSeq: 21571 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Authorization: Digest username="21056", realm="asterisk", nonce="1760619913/a628da38d98ff5dae29c7d14a8138d2e", uri="sip:21831@milena.mithril", response="b46d58567edb771661ca60ac5edd4fa6", algorithm=MD5, cnonce="abc4186da0b543c1a2be6ece1e97d755", opaque="2f8e186073d3b979", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length: 336

v=0
o=- 3969608713 3969608713 IN IP4 172.19.39.6
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 8 0 120
c=IN IP4 172.19.39.6
b=TIAS:64000
a=rtcp:4001 IN IP4 172.19.39.6
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-16
a=ssrc:407200375 cname:14134a3d4d742300

--end msg--
13:05:13.301 pjsua_core.c .RX 357 bytes Response msg 100/INVITE/cseq=21571 (rdata0x7f2b08004ae8) from UDP 172.19.39.4:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.19.39.6:5060;rport=5060;received=172.19.39.6;branch=z9hG4bKPj34145732-969b-461a-a255-518c45724db0
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
From: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
To: sip:21831@milena.mithril
CSeq: 21571 INVITE
Server: AsteriskPBX
Content-Length: 0

--end msg--
13:05:14.157 pjsua_core.c .RX 549 bytes Response msg 180/INVITE/cseq=21571 (rdata0x7f2b08004ae8) from UDP 172.19.39.4:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.19.39.6:5060;rport=5060;received=172.19.39.6;branch=z9hG4bKPj34145732-969b-461a-a255-518c45724db0
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
From: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
To: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
CSeq: 21571 INVITE
Server: AsteriskPBX
Contact: sip:172.19.39.4:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Content-Length: 0

--end msg--
13:05:14.157 d-modem.c .....Call 0 state=EARLY
13:05:14.290 pjsua_aud.c Closing sound device after idle for 1 second(s)
13:05:14.290 pjsua_aud.c .Closing null sound device..
13:05:17.996 pjsua_core.c .RX 938 bytes Response msg 200/INVITE/cseq=21571 (rdata0x7f2b08004ae8) from UDP 172.19.39.4:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.19.39.6:5060;rport=5060;received=172.19.39.6;branch=z9hG4bKPj34145732-969b-461a-a255-518c45724db0
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
From: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
To: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
CSeq: 21571 INVITE
Server: AsteriskPBX
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: sip:172.19.39.4:5060
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length: 259

v=0
o=- 3969608713 3969608715 IN IP4 172.19.39.4
s=Asterisk
c=IN IP4 172.19.39.4
t=0 0
m=audio 16768 RTP/AVP 0 8 120
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

--end msg--
13:05:17.996 d-modem.c .....Call 0 state=CONNECTING
13:05:17.996 inv0x560678059598 ....SDP negotiation done: Success
13:05:17.996 pjsua_media.c .....Call 0: updating media..
13:05:17.996 pjsua_media.c .......Media stream call00:0 is destroyed
13:05:17.996 pjsua_aud.c ......Audio channel update..
13:05:17.997 strm0x7f2b08009408 .......Encoder stream started
13:05:17.997 strm0x7f2b08009408 .......Decoder stream started
13:05:17.997 pjsua_media.c ......Audio updated, stream #0: PCMU (sendrecv)
13:05:17.997 pjsua_aud.c .....Conf connect: 1 --> 2
13:05:17.997 pjsua_aud.c ......Set sound device: capture=-99, playback=-99
13:05:17.997 pjsua_aud.c .......Setting null sound device..
13:05:17.997 pjsua_aud.c ........Opening null sound device..
13:05:17.997 conference.c ......Port 1 (sip:21831@milena.mithril) transmitting to port 2 (dmodem)
13:05:17.997 pjsua_aud.c .....Conf connect: 2 --> 1
13:05:17.997 conference.c ......Port 2 (dmodem) transmitting to port 1 (sip:21831@milena.mithril)
13:05:17.997 pjsua_core.c .....TX 376 bytes Request msg ACK/cseq=21571 (tdta0x7f2b08014008) to UDP 172.19.39.4:5060:
ACK sip:172.19.39.4:5060 SIP/2.0
Via: SIP/2.0/UDP 172.19.39.6:5060;rport;branch=z9hG4bKPjf95b00cb-2bbc-4080-b3e6-894422197cd4
Max-Forwards: 70
From: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
To: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
CSeq: 21571 ACK
Content-Length: 0

--end msg--
13:05:17.997 d-modem.c .....Call 0 state=CONFIRMED
13:05:17.997 pjsua_call.c .Call 0 sending UPDATE for updating media session to use only one codec
13:05:17.997 pjsua_core.c ....TX 867 bytes Request msg UPDATE/cseq=21572 (tdta0x7f2b08019238) to UDP 172.19.39.4:5060:
UPDATE sip:172.19.39.4:5060 SIP/2.0
Via: SIP/2.0/UDP 172.19.39.6:5060;rport;branch=z9hG4bKPj539750c5-0a41-4013-b109-31d55ae10201
Max-Forwards: 70
From: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
To: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
Contact: sip:21056@172.19.39.6:5060;ob
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
CSeq: 21572 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
Content-Type: application/sdp
Content-Length: 312

v=0
o=- 3969608713 3969608714 IN IP4 172.19.39.6
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 0 120
c=IN IP4 172.19.39.6
b=TIAS:64000
a=rtcp:4001 IN IP4 172.19.39.6
a=ssrc:407200375 cname:14134a3d4d742300
a=rtpmap:0 PCMU/8000
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-16
a=sendrecv

--end msg--
13:05:17.998 pjsua_core.c .RX 947 bytes Request msg INVITE/cseq=29045 (rdata0x7f2b08004ae8) from UDP 172.19.39.4:5060:
INVITE sip:21056@172.19.39.6:5060;ob SIP/2.0
Via: SIP/2.0/UDP 172.19.39.4:5060;rport;branch=z9hG4bKPj9617428c-4219-4e01-800a-352d357b8d5b
From: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
To: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
Contact: sip:172.19.39.4:5060
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
CSeq: 29045 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800;refresher=uas
Min-SE: 90
Max-Forwards: 70
User-Agent: AsteriskPBX
Content-Type: application/sdp
Content-Length: 236

v=0
o=- 3969608713 3969608716 IN IP4 172.19.39.4
s=Asterisk
c=IN IP4 172.19.4.248
t=0 0
m=audio 16426 RTP/AVP 0 120
a=rtpmap:0 PCMU/8000
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

--end msg--
13:05:17.998 pjsua_core.c .......TX 386 bytes Response msg 491/INVITE/cseq=29045 (tdta0x7f2b0801dd78) to UDP 172.19.39.4:5060:
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 172.19.39.4:5060;rport=5060;received=172.19.39.4;branch=z9hG4bKPj9617428c-4219-4e01-800a-352d357b8d5b
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
From: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
To: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
CSeq: 29045 INVITE
Content-Length: 0

--end msg--
13:05:18.001 strm0x7f2b08009408 !Resetting jitter buffer in stream playback start
13:05:18.010 pjsua_core.c .RX 431 bytes Response msg 491/UPDATE/cseq=21572 (rdata0x7f2b08004ae8) from UDP 172.19.39.4:5060:
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 172.19.39.6:5060;rport=5060;received=172.19.39.6;branch=z9hG4bKPj539750c5-0a41-4013-b109-31d55ae10201
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
From: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
To: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
CSeq: 21572 UPDATE
Reason: Q.850;cause=16
Server: AsteriskPBX
Content-Length: 0

--end msg--
13:05:18.010 pjsua_core.c .RX 414 bytes Request msg ACK/cseq=29045 (rdata0x7f2b08004ae8) from UDP 172.19.39.4:5060:
ACK sip:21056@172.19.39.6:5060;ob SIP/2.0
Via: SIP/2.0/UDP 172.19.39.4:5060;rport;branch=z9hG4bKPj9617428c-4219-4e01-800a-352d357b8d5b
From: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
To: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
CSeq: 29045 ACK
Max-Forwards: 70
User-Agent: AsteriskPBX
Content-Length: 0

--end msg--
13:05:18.287 udp0x56067805ab70 Remote RTP address switched to 172.19.4.248:16426
13:05:18.287 udp0x56067805ab70 Remote RTCP address switched to predicted address 172.19.4.248:16427
<919.577661> DCR: initial DC Evaluation done, DC level 0, enabled
13:05:19.931 pjsua_core.c .RX 947 bytes Request msg INVITE/cseq=29046 (rdata0x7f2b08004ae8) from UDP 172.19.39.4:5060:
INVITE sip:21056@172.19.39.6:5060;ob SIP/2.0
Via: SIP/2.0/UDP 172.19.39.4:5060;rport;branch=z9hG4bKPje93aedf8-3537-438d-8cdc-0e24d8711857
From: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
To: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
Contact: sip:172.19.39.4:5060
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
CSeq: 29046 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800;refresher=uas
Min-SE: 90
Max-Forwards: 70
User-Agent: AsteriskPBX
Content-Type: application/sdp
Content-Length: 236

v=0
o=- 3969608713 3969608716 IN IP4 172.19.39.4
s=Asterisk
c=IN IP4 172.19.4.248
t=0 0
m=audio 16426 RTP/AVP 0 120
a=rtpmap:0 PCMU/8000
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

--end msg--
13:05:19.932 pjsua_call.c .....Call 0: received updated media offer
13:05:19.932 pjsua_media.c ......Call 0: re-initializing media..
13:05:19.932 pjsua_media.c .......Media index 0 selected for audio call 0
13:05:19.932 inv0x560678059598 .....SDP negotiation done: Success
13:05:19.932 pjsua_media.c ......Call 0: updating media..
13:05:19.933 pjsua_media.c ........Media stream call00:0 is destroyed
13:05:19.933 pjsua_aud.c .......Audio channel update..
13:05:19.934 strm0x7f2b08009408 ........Encoder stream started
13:05:19.934 strm0x7f2b08009408 ........Decoder stream started
13:05:19.934 pjsua_media.c .......Audio updated, stream #0: PCMU (sendrecv)
13:05:19.935 pjsua_core.c ........TX 957 bytes Response msg 200/INVITE/cseq=29046 (tdta0x7f2b08025038) to UDP 172.19.39.4:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.19.39.4:5060;rport=5060;received=172.19.39.4;branch=z9hG4bKPje93aedf8-3537-438d-8cdc-0e24d8711857
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
From: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
To: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
CSeq: 29046 INVITE
Session-Expires: 1800;refresher=uas
Contact: sip:21056@172.19.39.6:5060;ob
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub, trickle-ice
Content-Type: application/sdp
Content-Length: 312

v=0
o=- 3969608713 3969608715 IN IP4 172.19.39.6
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 0 120
c=IN IP4 172.19.39.6
b=TIAS:64000
a=rtcp:4001 IN IP4 172.19.39.6
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-16
a=ssrc:407200375 cname:14134a3d4d742300

--end msg--
13:05:19.938 pjsua_core.c .RX 414 bytes Request msg ACK/cseq=29046 (rdata0x7f2b08004ae8) from UDP 172.19.39.4:5060:
ACK sip:21056@172.19.39.6:5060;ob SIP/2.0
Via: SIP/2.0/UDP 172.19.39.4:5060;rport;branch=z9hG4bKPje9be36a7-2e2f-4234-9522-be7e802d097e
From: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
To: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
CSeq: 29046 ACK
Max-Forwards: 70
User-Agent: AsteriskPBX
Content-Length: 0

--end msg--
13:05:24.547 strm0x7f2b08009408 Bad RTP pt 8 (expecting 0)
13:05:24.564 pjsua_core.c .RX 946 bytes Request msg INVITE/cseq=29047 (rdata0x7f2b08004ae8) from UDP 172.19.39.4:5060:
INVITE sip:21056@172.19.39.6:5060;ob SIP/2.0
Via: SIP/2.0/UDP 172.19.39.4:5060;rport;branch=z9hG4bKPjabb45d63-f8a1-4d25-83cb-af0d56a5eb9e
From: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
To: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
Contact: sip:172.19.39.4:5060
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
CSeq: 29047 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800;refresher=uas
Min-SE: 90
Max-Forwards: 70
User-Agent: AsteriskPBX
Content-Type: application/sdp
Content-Length: 235

v=0
o=- 3969608713 3969608717 IN IP4 172.19.39.4
s=Asterisk
c=IN IP4 172.19.39.4
t=0 0
m=audio 16768 RTP/AVP 0 120
a=rtpmap:0 PCMU/8000
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

--end msg--
13:05:24.564 pjsua_call.c .....Call 0: received updated media offer
13:05:24.564 pjsua_media.c ......Call 0: re-initializing media..
13:05:24.564 pjsua_media.c .......Media index 0 selected for audio call 0
13:05:24.564 inv0x560678059598 .....SDP negotiation done: Success
13:05:24.564 pjsua_media.c ......Call 0: updating media..
13:05:24.565 pjsua_media.c ........Media stream call00:0 is destroyed
13:05:24.565 pjsua_aud.c .......Audio channel update..
13:05:24.565 strm0x7f2b0801cc08 ........Encoder stream started
13:05:24.565 strm0x7f2b0801cc08 ........Decoder stream started
13:05:24.565 pjsua_media.c .......Audio updated, stream #0: PCMU (sendrecv)
13:05:24.565 pjsua_core.c ........TX 957 bytes Response msg 200/INVITE/cseq=29047 (tdta0x7f2b08019f98) to UDP 172.19.39.4:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.19.39.4:5060;rport=5060;received=172.19.39.4;branch=z9hG4bKPjabb45d63-f8a1-4d25-83cb-af0d56a5eb9e
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
From: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
To: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
CSeq: 29047 INVITE
Session-Expires: 1800;refresher=uas
Contact: sip:21056@172.19.39.6:5060;ob
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub, trickle-ice
Content-Type: application/sdp
Content-Length: 312

v=0
o=- 3969608713 3969608716 IN IP4 172.19.39.6
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 0 120
c=IN IP4 172.19.39.6
b=TIAS:64000
a=rtcp:4001 IN IP4 172.19.39.6
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-16
a=ssrc:407200375 cname:14134a3d4d742300

--end msg--
13:05:24.567 pjsua_core.c .RX 414 bytes Request msg ACK/cseq=29047 (rdata0x7f2b08004ae8) from UDP 172.19.39.4:5060:
ACK sip:21056@172.19.39.6:5060;ob SIP/2.0
Via: SIP/2.0/UDP 172.19.39.4:5060;rport;branch=z9hG4bKPj1946249d-568f-4892-b55b-c2ba7ebf2cee
From: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
To: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
CSeq: 29047 ACK
Max-Forwards: 70
User-Agent: AsteriskPBX
Content-Length: 0

--end msg--
13:05:24.568 pjsua_core.c .RX 414 bytes Request msg BYE/cseq=29048 (rdata0x7f2b08004ae8) from UDP 172.19.39.4:5060:
BYE sip:21056@172.19.39.6:5060;ob SIP/2.0
Via: SIP/2.0/UDP 172.19.39.4:5060;rport;branch=z9hG4bKPjbabad547-1789-4f0b-b07e-300babf3fa0d
From: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
To: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
CSeq: 29048 BYE
Max-Forwards: 70
User-Agent: AsteriskPBX
Content-Length: 0

--end msg--
13:05:24.568 pjsua_core.c .......TX 370 bytes Response msg 200/BYE/cseq=29048 (tdta0x7f2b08019f98) to UDP 172.19.39.4:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.19.39.4:5060;rport=5060;received=172.19.39.4;branch=z9hG4bKPjbabad547-1789-4f0b-b07e-300babf3fa0d
Call-ID: 4126e8e1-d3ce-46d7-a59f-4ecfcb3f6486
From: sip:21831@milena.mithril;tag=6b161907-99f4-4380-85d5-474ae2c1c64f
To: sip:21056@milena.mithril;tag=403d6874-3a6b-44c9-9e9b-a231b7395a1f
CSeq: 29048 BYE
Content-Length: 0

--end msg--
13:05:24.568 pjsua_media.c ......Call 0: deinitializing media..
13:05:24.568 pjsua_media.c ........Media stream call00:0 is destroyed
13:05:24.569 d-modem.c NNCTD
13:05:24.569 pjsua_core.c ......Shutting down, flags=0...
13:05:24.569 pjsua_core.c ......PJSUA state changed: RUNNING --> CLOSING
13:05:24.569 pjsua_core.c ......Error joining worker thread: Operation cancelled (PJ_ECANCELLED)
<924.569153> main: lost connection to child socket process
<924.569191> slamr0: modem_hup...2
<924.569200> slamr0: modem set state: 2 --> 9...
<924.569209> slamr0: new state: DP_DISC
<924.569216> slamr0: modem_stop..
<924.569222> main: socket_stop...
13:05:25.569 pjsua_call.c .......Hangup all calls..
13:05:25.569 pjsua_call.c ........Call 0 hanging up: code=0..
13:05:25.569 pjsua_media.c .........Call 0: deinitializing media..
13:05:25.569 pjsua_media.c ...........Media stream call00:0 is destroyed
13:05:25.569 d-modem.c .........Call 0 state=DISCONNCTD
13:05:25.569 pjsua_call.c .........Failed to create end session message: INVITE session already terminated (PJSIP_ESESSIONTERMINATED) [status=171140]
13:05:25.569 pjsua_call.c .........Will retry call 0 hangup in 5000 msec
13:05:25.569 pjsua_media.c .......Call 0: deinitializing media..
13:05:25.569 pjsua_media.c .........Media stream call00:0 is destroyed
13:05:25.569 pjsua_media.c .......Call 1: deinitializing media..
13:05:25.569 pjsua_media.c .......Call 2: deinitializing media..
13:05:25.569 pjsua_media.c .......Call 3: deinitializing media..
13:05:25.569 pjsua_pres.c .......Shutting down presence..
13:05:25.569 pjsua_aud.c .......Closing sound device after idle for 1 second(s)
13:05:25.569 pjsua_aud.c ........Closing null sound device..
13:05:26.576 pjsua_core.c .......Destroying...
13:05:26.576 pjsua_media.c .......Shutting down media..
13:05:26.578 sip_transaction.c .......Stopping transaction layer module
13:05:26.578 sip_transaction.c .......Stopped transaction layer module
13:05:26.578 sip_endpoint.c .......Module "mod-unsolicited-mwi" unregistered
13:05:26.578 sip_endpoint.c .......Module "mod-pjsua-options" unregistered
13:05:26.578 sip_endpoint.c .......Module "mod-pjsua-im" unregistered
13:05:26.578 sip_endpoint.c .......Module "mod-pjsua-pres" unregistered
13:05:26.578 sip_endpoint.c .......Module "mod-pjsua" unregistered
13:05:26.578 sip_endpoint.c .......Module "mod-stateful-util" unregistered
13:05:26.578 sip_endpoint.c .......Module "mod-refer" unregistered
13:05:26.578 sip_endpoint.c .......Module "mod-mwi" unregistered
13:05:26.578 sip_endpoint.c .......Module "mod-presence" unregistered
13:05:26.578 sip_endpoint.c .......Module "mod-evsub" unregistered
13:05:26.578 sip_endpoint.c .......Module "mod-invite" unregistered
13:05:26.578 sip_endpoint.c .......Module "mod-100rel" unregistered
13:05:26.578 sip_endpoint.c .......Module "mod-ua" unregistered
13:05:26.578 sip_transaction.c .......Transaction layer module destroyed
13:05:26.578 sip_endpoint.c .......Module "mod-tsx-layer" unregistered
13:05:26.578 sip_endpoint.c .......Module "mod-msg-print" unregistered
13:05:26.578 sip_endpoint.c .......Module "mod-pjsua-log" unregistered
13:05:26.579 sip_transport.c .......Warning: 2 transmit buffer(s) not freed!
13:05:26.580 timer.c .......Dumping timer heap:
13:05:26.580 timer.c ....... Cur size: 3 entries, max: timer.c ....... Entries:
13:05:26.580 timer.c ....... _id Id Elapsed Source
13:05:26.580 timer.c ....... ----------------------------------
13:05:26.580 timer.c ....... 4 1 3.989 ../src/pjsua-lib/pjsua_call.c:2848
13:05:26.580 timer.c ....... 1 2 1797.986 ../src/pjsip-ua/sip_timer.c:514
13:05:26.580 timer.c ....... 3 1 897.986 ../src/pjsip-ua/sip_timer.c:529
13:05:26.580 sip_endpoint.c .......Endpoint 0x5606780211b8 destroyed
13:05:26.580 udp0x560678054920 .......Pool is not released by application, releasing now
13:05:26.580 glck0x560678054d40 .......Pool is not released by application, releasing now
13:05:26.580 rtd0x560678055370 .......Pool is not released by application, releasing now
13:05:26.580 dlg0x5606780594f0 .......Pool is not released by application, releasing now
13:05:26.580 glck0x560678059d00 .......Pool is not released by application, releasing now
13:05:26.580 inv0x5606780594f0 .......Pool is not released by application, releasing now
13:05:26.580 inv0x5606780594f0 .......Pool is not released by application, releasing now
13:05:26.580 tdta0x560678070b48 .......Pool is not released by application, releasing now
13:05:26.580 auth_cli0x7f2b08003a80 .......Pool is not released by application, releasing now
13:05:26.580 tdta0x7f2b08014008 .......Pool is not released by application, releasing now
13:05:26.580 tsx0x7f2b08003f38 .......Pool is not released by application, releasing now
13:05:26.580 glck0x7f2b0801b960 .......Pool is not released by application, releasing now
13:05:26.580 pjsua_core.c .......PJSUA state changed: CLOSING --> NULL
13:05:26.580 pjsua_core.c .......PJSUA destroyed...
<926.581997> slamr0: modem set hook: 1 --> 0...
<926.582048> main: socket_ioctl: cmd 2, arg 0...
<926.582064> main: socket_ioctl: returning 0
<926.582072> main: socket_ioctl: cmd 8, arg 0...
<926.582105> call: delete...
<926.582120> Dialer was aborted.
<926.582126> CALLPROG_Delete is entered
<926.582140> cadence_delete with CADENCE_DIAL_OBJ is invoked
<926.582148> cadence_delete with CADENCE_OBJ is invoked
<926.582161> CALLPROG_Delete is exited
<926.582170> slamr0: modem set state: 9 --> 1...
<926.582184> slamr0: new state: MODEM_IDLE
<926.582190> slamr0: modem report result: 3 (NO CARRIER)

[VasylSamoilov]

Try looking at forks, for example https://github.com/cryan209/D-Modem - it can even receive calls (very unstable).

[ROBERT-MCDOWELL]

Try looking at forks, for example https://github.com/cryan209/D-Modem - it can even receive calls (very unstable).

thanks mate!

[merclangrat]

Try looking at forks, for example https://github.com/cryan209/D-Modem - it can even receive calls (very unstable).

Thank you! I will try and share how it worked (or not worked).

As I see, there are some forks, I understand why - the original project hasn't been updated for 4 years. Maybe, it's time to make my own.... :)

[ilovehalflife98]

Try looking at forks, for example https://github.com/cryan209/D-Modem - it can even receive calls (very unstable).

Tried using it and it can answer and do V.34 negotiation perfectly, but I hear no sound. I'm using 3 VMware VMs, all running Debian 10. One running asterisk using chan_sip, other is D-Modem 1, other is D-Modem 2
Asterisk server uses chan_sip
Here is my sip.conf:
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
qualify=yes
allowguest=no
disallow=all
allow=ulaw
jbenable=no
jbforce=no
silencesuppression=no

[2001]
type=friend
secret=***** (i am not going to show it)
host=dynamic
context=d-modem-users
dtmfmode=inband
callerid="D-Modem 1" <2001>
disallow=all
allow=ulaw
progressinband=yes
silencesuppression=no

[2002] (same config but callerid="D-Modem 2" <2002>)
Please answer, thanks.

[VasylSamoilov]

Tried using it and it can answer and do V.34 negotiation perfectly, but I hear no sound

You should probably move questions like that to a corresponding repository. First of all, it works, congratulations :)

[ilovehalflife98]

Tried using it and it can answer and do V.34 negotiation perfectly, but I hear no sound

You should probably move questions like that to a corresponding repository. First of all, it works, congratulations :)

I asked here because there isn't even an option to open an issue onto that fork
Sorry if this annoyed you.

[VasylSamoilov]

it's pretty unstable software. I don't understand what does it mean "not to hear a sound". You will not hear modem if you dial in, as far as I remember. Dialing (originate) modem should send a tone first. I'm not quite sure if it's a part of v.34 standard, it may be the other way around.

[ilovehalflife98]

it's pretty unstable software. I don't understand what does it mean "not to hear a sound". You will not hear modem if you dial in, as far as I remember. Dialing (originate) modem should send a tone first. I'm not quite sure if it's a part of v.34 standard, it may be the other way around.
Again not to annoy you, but recently another fork has yet came out
https://github.com/trainman419/D-Modem
But it looks very similar to cryan209’s fork from the README and I feel like this might just be another copy of his fork so I don’t know the point of me sending this. But I can’t really judge it without trying it so I’ll check it out later and tell you how it goes

[ilovehalflife98]

it's pretty unstable software. I don't understand what does it mean "not to hear a sound". You will not hear modem if you dial in, as far as I remember. Dialing (originate) modem should send a tone first. I'm not quite sure if it's a part of v.34 standard, it may be the other way around.
Again not to annoy you, but recently another fork has yet came out
https://github.com/trainman419/D-Modem
But it looks very similar to cryan209’s fork from the README and I feel like this might just be another copy of his fork so I don’t know the point of me sending this. But I can’t really judge it without trying it so I’ll check it out later and tell you how it goes

Well.. it looks like I was wrong!
That fork is way much better than cryan209's fork, I can call as much as I want with the handshake working, etc...
Try it out if you want

[ROBERT-MCDOWELL]

is anyone has a good example to connect D-Modemwith a SIP trunk?

[ilovehalflife98]

is anyone has a good example to connect D-Modemwith a SIP trunk?

You could possibly try to get asterisk to register with it as a trunk. Then configure the dialplan as needed (for example, when 9 then number dialed, strip 9 and send rest of digits to trunk). Or if the SIP trunk supports regular registration, you might as well just make D-Modem register with the trunk. If you’re using Twilio, make sure to use asterisk and make asterisk authenticate from just IP ACL, no registration. In the console make sure asterisk’s IP is in the list.

[ROBERT-MCDOWELL]

@ilovehalflife98 thanks for your comment, I actually use freeSWITCH. I can see how to set the dialplan and trunk,
but how to register the D-Modem I really don't know how to code that.