vicidial.org

Version: 2.10-475a Build: 150218-0924 asterisk: 1.8.29.0

Hello, new here and in the process of evaluating VICIdial for production use. Currently have it set up and can dial out (using a sipstation trunk) but not receiving any audio. I've searched the internet to see if I could find something remotely close to my issue but doesn't appear to be any. I already run a full asterisk implementation for phones so am not figuring out why for the VICIdial, I am not receiving any audio. Any help would be greatly appreciated.

[fpbx-1]
disallow=all
allow=ulaw
context=from-pstn
type=friend
insecure=port,invite
qualify=yes
sendrpid=yes
trustrpid=yes
dtmfmode=rfc2833
outofcall_message_context=sms-incoming
username=username
secret=secret
host=freepbx trunk


exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXXX,2,Dial(${fpbx-1}/${EXTEN},,tTor)
exten => _1NXXNXXXXXX,3,Hangup

exten => _NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _NXXNXXXXXX,2,Dial(${fpbx-1}/${EXTEN},,tTor)
exten => _NXXNXXXXXX,3,Hangup

[Mar 4 13:48:20]
<--- SIP read from UDP:192.168.11.107:55983 --->
INVITE sip:XXXXXXXXXX@192.168.11.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.107:55983;branch=z9hG4bK-d8754z-79699c6784028002-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:gs102@192.168.11.107:55983;rinstance=f71424684d9cf180>
To: <sip:XXXXXXXXXX@192.168.11.9:5060>
From: "msh"<sip:gs102@192.168.11.9:5060>;tag=de2fbc7c
Call-ID: ZGJhZDM1ZWYyODk5Yjc2OWYzMjY5NTg0NDM2YjdjOGI.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Content-Length: 409

v=0
o=3cxVCE 254874870 56256165 IN IP4 192.168.11.107
s=3cxVCE Audio Call
c=IN IP4 192.168.11.107
t=0 0
m=audio 40030 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40028 RTP/AVP 34
c=IN IP4 192.168.11.107
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
a=sendrecv
<------------->
[Mar 4 13:48:20] --- (13 headers 18 lines) ---
[Mar 4 13:48:20] Sending to 192.168.11.107:55983 (NAT)
[Mar 4 13:48:20] Using INVITE request as basis request - ZGJhZDM1ZWYyODk5Yjc2OWYzMjY5NTg0NDM2YjdjOGI.
[Mar 4 13:48:20] Found peer 'gs102' for 'gs102' from 192.168.11.107:55983
[Mar 4 13:48:20]
<--- Reliably Transmitting (NAT) to 192.168.11.107:55983 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.11.107:55983;branch=z9hG4bK-d8754z-79699c6784028002-1---d8754z-;received=192.168.11.107;rport=55983
From: "msh"<sip:gs102@192.168.11.9:5060>;tag=de2fbc7c
To: <sip:XXXXXXXXXX@192.168.11.9:5060>;tag=as400ecef7
Call-ID: ZGJhZDM1ZWYyODk5Yjc2OWYzMjY5NTg0NDM2YjdjOGI.
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.29.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1505922b"
Content-Length: 0


<------------>
[Mar 4 13:48:20] Scheduling destruction of SIP dialog 'ZGJhZDM1ZWYyODk5Yjc2OWYzMjY5NTg0NDM2YjdjOGI.' in 6656 ms (Method: INVITE)
[Mar 4 13:48:20]
<--- SIP read from UDP:192.168.11.107:55983 --->
ACK sip:XXXXXXXXXX@192.168.11.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.107:55983;branch=z9hG4bK-d8754z-79699c6784028002-1---d8754z-;rport
Max-Forwards: 70
To: <sip:XXXXXXXXXX@192.168.11.9:5060>;tag=as400ecef7
From: "msh"<sip:gs102@192.168.11.9:5060>;tag=de2fbc7c
Call-ID: ZGJhZDM1ZWYyODk5Yjc2OWYzMjY5NTg0NDM2YjdjOGI.
CSeq: 1 ACK
Content-Length: 0

<------------->
[Mar 4 13:48:20] --- (8 headers 0 lines) ---
[Mar 4 13:48:20]
<--- SIP read from UDP:192.168.11.107:55983 --->
INVITE sip:XXXXXXXXXX@192.168.11.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.107:55983;branch=z9hG4bK-d8754z-954f5d6cb9285968-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:gs102@192.168.11.107:55983;rinstance=f71424684d9cf180>
To: <sip:XXXXXXXXXX@192.168.11.9:5060>
From: "msh"<sip:gs102@192.168.11.9:5060>;tag=de2fbc7c
Call-ID: ZGJhZDM1ZWYyODk5Yjc2OWYzMjY5NTg0NDM2YjdjOGI.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="gs102",realm="asterisk",nonce="1505922b",uri="sip:XXXXXXXXXX@192.168.11.9:5060",response="4513da85fe322657162632deae03d5de",algorithm=MD5
Content-Length: 409

v=0
o=3cxVCE 254874870 56256165 IN IP4 192.168.11.107
s=3cxVCE Audio Call
c=IN IP4 192.168.11.107
t=0 0
m=audio 40030 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40028 RTP/AVP 34
c=IN IP4 192.168.11.107
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
a=sendrecv
<------------->
[Mar 4 13:48:20] --- (14 headers 18 lines) ---
[Mar 4 13:48:20] Sending to 192.168.11.107:55983 (NAT)
[Mar 4 13:48:20] Using INVITE request as basis request - ZGJhZDM1ZWYyODk5Yjc2OWYzMjY5NTg0NDM2YjdjOGI.
[Mar 4 13:48:20] Found peer 'gs102' for 'gs102' from 192.168.11.107:55983
[Mar 4 13:48:20] == Using SIP RTP CoS mark 5
[Mar 4 13:48:20] Found RTP audio format 0
[Mar 4 13:48:20] Found RTP audio format 8
[Mar 4 13:48:20] Found RTP audio format 3
[Mar 4 13:48:20] Found RTP audio format 101
[Mar 4 13:48:20] Found audio description format PCMU for ID 0
[Mar 4 13:48:20] Found audio description format PCMA for ID 8
[Mar 4 13:48:20] Found audio description format GSM for ID 3
[Mar 4 13:48:20] Found audio description format telephone-event for ID 101
[Mar 4 13:48:20] Found RTP video format 34
[Mar 4 13:48:20] Found video description format H263 for ID 34
[Mar 4 13:48:20] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0xe (gsm|ulaw|alaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Mar 4 13:48:20] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Mar 4 13:48:20] Peer audio RTP is at port 192.168.11.107:40030
[Mar 4 13:48:20] Looking for XXXXXXXXXX in default (domain 192.168.11.9)
[Mar 4 13:48:20] list_route: hop: <sip:gs102@192.168.11.107:55983;rinstance=f71424684d9cf180>
[Mar 4 13:48:20]
<--- Transmitting (NAT) to 192.168.11.107:55983 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.107:55983;branch=z9hG4bK-d8754z-954f5d6cb9285968-1---d8754z-;received=192.168.11.107;rport=55983
From: "msh"<sip:gs102@192.168.11.9:5060>;tag=de2fbc7c
To: <sip:XXXXXXXXXX@192.168.11.9:5060>
Call-ID: ZGJhZDM1ZWYyODk5Yjc2OWYzMjY5NTg0NDM2YjdjOGI.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.29.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:XXXXXXXXXX@192.168.11.9:5060>
Content-Length: 0

Have you checked that all the RTP ports defined in /etc/asterisk/rtp.conf are actually open? Your output suggests you're using nat=yes in /etc/asterisk/sip.conf, are you in fact using nat? It's more likely the RTP ports, NAT traversal issues usually result in 1 way audio, not silence.