Outside Calls No Audio

Asterisk
1

I’ve spent pretty much the whole day on this and can’t figure it out.

I’m running *Now on a VirtualBox and internal extension to extension calls all seem to work fine. I signed up for a trial for Sipstation and that all seems to work fine, too. However, when I try to make a call to/from an external line over the Sipstation DID the call rings and connects, but there is no audio on either end.

I’ve read over and over that this is 9/10 a NAT problem, however, I’ve forwarded all of the correct ports set up my external and local ip parameters correctly via FreePBX. I’ve set VirtualBox to be allowed in Windows Firewall, and even gone as far as DMZ’d the *Now server IP.

The only thing I can think of is to also open the ports for the machine running the VirtualBox?

I’ve scoured the net and tried every possible solution to this and nothing is helping.

2

If you are using FreePBX make sure , you have nat set to nat=force_rport,comedia on each sip peer (extensions, trunks)

3

That was set on all extensions. I tried setting a new server and now the DID for Sipstation won’t even ring. There’s no indication there’s any connection to my PBX server.

4

enable sip debug if the invite is reaching Asterisk you will see it on the cli

5

I got the call to go through. In SIPStation module My Contact IP does not match my public IP (it’s showing my internal WAN IP) and it’s giving a warning, but that seemed to make the difference. All I did was turn off DDNS in sip settings and change it back to Public.

Back to not having sound in external calls. I did the sip debug, but I have no idea how to read it.

I didn’t see any “Notices” or “Warnings” that looked out of the ordinary.

I also noticed that calling internal extensions from external extensions doesn’t produce audio, either.

6

Here’s where it seems to go wonky, but I don’t know what this all means:

<— SIP read from UDP:MYSOFTPHONE:60072 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP PBXINTERNAL:5060;branch=z9hG4bK3cc0c076;rport=5060
Contact: sip:101@PBXEXTERNAL:60072
To: sip:101@PBXEXTERNAL:60072;rinstance=415f21e6e76bf181;transport=UDP;tag=fa2b8a34
From: “+MYCELLPHONE” sip:MYCELLPHONE@PBXINTERNAL;tag=as31ef9084
Call-ID: 35be3ca07912c50d4cd8b91824fd96e1@PBXINTERNAL:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 243

v=0
o=Z 0 3 IN IP4 PBXEXTERNAL
s=Z
c=IN IP4 PBXEXTERNAL
t=0 0
m=audio 8000 RTP/AVP 0 3 110 8 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
— (13 headers 12 lines) —
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(gsm|ulaw|alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port PBXEXTERNAL:8000
list_route: hop: sip:101@PBXEXTERNAL:60072
set_destination: Parsing sip:101@PBXEXTERNAL:60072 for address/port to send to
set_destination: set destination to PBXEXTERNAL:60072
Transmitting (NAT) to MYSOFTPHONE:60072:
ACK sip:101@PBXEXTERNAL:60072 SIP/2.0
Via: SIP/2.0/UDP PBXINTERNAL:5060;branch=z9hG4bK31419046;rport
Max-Forwards: 70
From: “+MYCELLPHONE” sip:MYCELLPHONE@PBXINTERNAL;tag=as31ef9084
To: sip:101@PBXEXTERNAL:60072;rinstance=415f21e6e76bf181;transport=UDP;tag=fa2b8a34
Contact: sip:MYCELLPHONE@PBXINTERNAL:5060
Call-ID: 35be3ca07912c50d4cd8b91824fd96e1@PBXINTERNAL:5060
CSeq: 102 ACK
User-Agent: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
Content-Length: 0

<— SIP read from UDP:MYSOFTPHONE:60072 —>
PUBLISH sip:101@PBXINTERNAL;transport=UDP SIP/2.0
Via: SIP/2.0/UDP PBXEXTERNAL:60072;branch=z9hG4bK-524287-1—08a10b3cb0fdc763
Max-Forwards: 70
Contact: sip:101@PBXEXTERNAL:60072;transport=UDP
To: sip:101@PBXINTERNAL;transport=UDP
From: sip:101@PBXINTERNAL;transport=UDP;tag=a103346e
Call-ID: SceKuAaduXybWUPJxcgV-w…
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 264

<?xml version="1.0" encoding="UTF-8"?>

open On the phone

<------------->
— (16 headers 3 lines) —
– SIP/101-0000026f answered SIP/fpbx-1-gDp0wWX7PmSc-0000026e
Sending to MYSOFTPHONE:60072 (NAT)

<— Transmitting (NAT) to MYSOFTPHONE:60072 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP PBXEXTERNAL:60072;branch=z9hG4bK-524287-1—08a10b3cb0fdc763;received=MYSOFTPHONE;rport=60072
From: sip:101@PBXINTERNAL;transport=UDP;tag=a103346e
To: sip:101@PBXINTERNAL;transport=UDP;tag=as2efc2830
Call-ID: SceKuAaduXybWUPJxcgV-w…
CSeq: 1 PUBLISH
Server: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘SceKuAaduXybWUPJxcgV-w…’ Method: PUBLISH
Audio is at 12422
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— SIP read from UDP:MYSOFTPHONE:60072 —>
SUBSCRIBE sip:101@PBXINTERNAL;transport=UDP SIP/2.0
Via: SIP/2.0/UDP PBXEXTERNAL:60072;branch=z9hG4bK-524287-1—1ba76100708b6abf
Max-Forwards: 70
Contact: sip:101@PBXEXTERNAL:60072;transport=UDP
To: sip:101@PBXINTERNAL;transport=UDP
From: sip:101@PBXINTERNAL;transport=UDP;tag=8229676b
Call-ID: 32acDTR-D4rpB84eudNbvA…
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (16 headers 0 lines) —

<— Reliably Transmitting (NAT) to 162.253.134.135:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 162.253.134.135:5060;branch=z9hG4bKF8B9SUpBSZ6aB;received=162.253.134.135;rport=5060
From: “+MYCELLPHONE” sip:MYCELLPHONE@162.253.134.135:5060;tag=m4Se4j58334Zm
To: sip:MYTRUNKDID@PBXINTERNAL:5060;tag=as1bd2d489
Call-ID: 96c480ea-7050-1234-9eb2-00163c395e02
CSeq: 89292050 INVITE
Server: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:MYTRUNKDID@PBXINTERNAL:5060
Remote-Party-ID: “Christopher Smith” sip:101@162.253.134.135;party=called;privacy=off;screen=no
Content-Type: application/sdp
Require: timer
Content-Length: 236

v=0
o=root 1599634423 1599634423 IN IP4 PBXINTERNAL
s=Asterisk PBX 11.16.0
c=IN IP4 PBXINTERNAL
t=0 0
m=audio 12422 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Sending to MYSOFTPHONE:60072 (NAT)
Creating new subscription
Sending to MYSOFTPHONE:60072 (NAT)
list_route: hop: sip:101@PBXEXTERNAL:60072;transport=UDP
Found peer ‘101’ for ‘101’ from MYSOFTPHONE:60072

<— Transmitting (NAT) to MYSOFTPHONE:60072 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP PBXEXTERNAL:60072;branch=z9hG4bK-524287-1—1ba76100708b6abf;received=MYSOFTPHONE;rport=60072
From: sip:101@PBXINTERNAL;transport=UDP;tag=8229676b
To: sip:101@PBXINTERNAL;transport=UDP;tag=as7a549d91
Call-ID: 32acDTR-D4rpB84eudNbvA…
CSeq: 1 SUBSCRIBE
Server: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“32d8c932”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘32acDTR-D4rpB84eudNbvA…’ in 6400 ms (Method: SUBSCRIBE)

<— SIP read from UDP:MYSOFTPHONE:60072 —>
SUBSCRIBE sip:101@PBXINTERNAL;transport=UDP SIP/2.0
Via: SIP/2.0/UDP PBXEXTERNAL:60072;branch=z9hG4bK-524287-1—9620f29fabf1973a
Max-Forwards: 70
Contact: sip:101@PBXEXTERNAL:60072;transport=UDP
To: sip:101@PBXINTERNAL;transport=UDP
From: sip:101@PBXINTERNAL;transport=UDP;tag=8229676b
Call-ID: 32acDTR-D4rpB84eudNbvA…
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username=“101”,realm=“asterisk”,nonce=“32d8c932”,uri=“sip:101@PBXINTERNAL;transport=UDP”,response=“034b507f7b6c69a59c3f66b44e527e22”,algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (17 headers 0 lines) —
Creating new subscription
Sending to MYSOFTPHONE:60072 (NAT)
Found peer ‘101’ for ‘101’ from MYSOFTPHONE:60072

<— Transmitting (NAT) to MYSOFTPHONE:60072 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP PBXEXTERNAL:60072;branch=z9hG4bK-524287-1—9620f29fabf1973a;received=MYSOFTPHONE;rport=60072
From: sip:101@PBXINTERNAL;transport=UDP;tag=8229676b
To: sip:101@PBXINTERNAL;transport=UDP;tag=as7a549d91
Call-ID: 32acDTR-D4rpB84eudNbvA…
CSeq: 2 SUBSCRIBE
Server: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘32acDTR-D4rpB84eudNbvA…’ Method: SUBSCRIBE

<— SIP read from UDP:MYSOFTPHONE:60072 —>
PUBLISH sip:101@PBXINTERNAL;transport=UDP SIP/2.0
Via: SIP/2.0/UDP PBXEXTERNAL:60072;branch=z9hG4bK-524287-1—a6b46643c54cb5e5
Max-Forwards: 70
Contact: sip:101@PBXEXTERNAL:60072;transport=UDP
To: sip:101@PBXINTERNAL;transport=UDP
From: sip:101@PBXINTERNAL;transport=UDP;tag=fa4b4d76
Call-ID: Zh6QdsOc0mjWQ7d7V4dnrw…
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 264

<?xml version="1.0" encoding="UTF-8"?>

open On the phone

<------------->
— (16 headers 3 lines) —
Sending to MYSOFTPHONE:60072 (NAT)

<— Transmitting (NAT) to MYSOFTPHONE:60072 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP PBXEXTERNAL:60072;branch=z9hG4bK-524287-1—a6b46643c54cb5e5;received=MYSOFTPHONE;rport=60072
From: sip:101@PBXINTERNAL;transport=UDP;tag=fa4b4d76
To: sip:101@PBXINTERNAL;transport=UDP;tag=as43184277
Call-ID: Zh6QdsOc0mjWQ7d7V4dnrw…
CSeq: 1 PUBLISH
Server: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘Zh6QdsOc0mjWQ7d7V4dnrw…’ Method: PUBLISH

<— SIP read from UDP:162.253.134.135:5060 —>
ACK sip:MYTRUNKDID@PBXINTERNAL:5060 SIP/2.0
Via: SIP/2.0/UDP 162.253.134.135:5060;rport;branch=z9hG4bKHtytXHrjKHKgj
Max-Forwards: 70
From: “+MYCELLPHONE” sip:MYCELLPHONE@162.253.134.135:5060;tag=m4Se4j58334Zm
To: sip:MYTRUNKDID@PBXINTERNAL:5060;tag=as1bd2d489
Call-ID: 96c480ea-7050-1234-9eb2-00163c395e02
CSeq: 89292050 ACK
Contact: sip:mod_sofia@162.253.134.135:5060
Content-Length: 0

<------------->
— (9 headers 0 lines) —

<— SIP read from UDP:162.253.134.135:5060 —>
BYE sip:MYTRUNKDID@PBXINTERNAL:5060 SIP/2.0
Via: SIP/2.0/UDP 162.253.134.135:5060;rport;branch=z9hG4bKKcHc17SSD3ZNS
Max-Forwards: 70
From: “+MYCELLPHONE” sip:MYCELLPHONE@162.253.134.135:5060;tag=m4Se4j58334Zm
To: sip:MYTRUNKDID@PBXINTERNAL:5060;tag=as1bd2d489
Call-ID: 96c480ea-7050-1234-9eb2-00163c395e02
CSeq: 89292051 BYE
Contact: sip:mod_sofia@162.253.134.135:5060
User-Agent: SIPStation 2.11.3
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=16;text=“NORMAL_CLEARING”
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 162.253.134.135:5060 (NAT)
Scheduling destruction of SIP dialog ‘96c480ea-7050-1234-9eb2-00163c395e02’ in 33344 ms (Method: BYE)

<— Transmitting (NAT) to 162.253.134.135:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 162.253.134.135:5060;branch=z9hG4bKKcHc17SSD3ZNS;received=162.253.134.135;rport=5060
From: “+MYCELLPHONE” sip:MYCELLPHONE@162.253.134.135:5060;tag=m4Se4j58334Zm
To: sip:MYTRUNKDID@PBXINTERNAL:5060;tag=as1bd2d489
Call-ID: 96c480ea-7050-1234-9eb2-00163c395e02
CSeq: 89292051 BYE
Server: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
– Executing [h@macro-dial-one:1] Macro(“SIP/fpbx-1-gDp0wWX7PmSc-0000026e”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] ExecIf(“SIP/fpbx-1-gDp0wWX7PmSc-0000026e”, “0?Set(CDR(recordingfile)=.wav)”) in new stack
– Executing [s@macro-hangupcall:2] GotoIf(“SIP/fpbx-1-gDp0wWX7PmSc-0000026e”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [s@macro-hangupcall:4] Hangup(“SIP/fpbx-1-gDp0wWX7PmSc-0000026e”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/fpbx-1-gDp0wWX7PmSc-0000026e’ in macro ‘hangupcall’
== Spawn extension (macro-dial-one, h, 1) exited non-zero on ‘SIP/fpbx-1-gDp0wWX7PmSc-0000026e’
Scheduling destruction of SIP dialog ‘35be3ca07912c50d4cd8b91824fd96e1@PBXINTERNAL:5060’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:101@PBXEXTERNAL:60072 for address/port to send to
set_destination: set destination to PBXEXTERNAL:60072
Reliably Transmitting (NAT) to MYSOFTPHONE:60072:
BYE sip:101@PBXEXTERNAL:60072 SIP/2.0
Via: SIP/2.0/UDP PBXINTERNAL:5060;branch=z9hG4bK25ddc226;rport
Max-Forwards: 70
From: “+MYCELLPHONE” sip:MYCELLPHONE@PBXINTERNAL;tag=as31ef9084
To: sip:101@PBXEXTERNAL:60072;rinstance=415f21e6e76bf181;transport=UDP;tag=fa2b8a34
Call-ID: 35be3ca07912c50d4cd8b91824fd96e1@PBXINTERNAL:5060
CSeq: 103 BYE
User-Agent: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

7

Please re-edit to use preformatted text, so we don’t lose important information like the contents of the Contact header.

Please ensure you have the INVITE request, and just the 200 OK.

Removing the non-INVITE, non-ACK transactions would make it easier to read.

We don’t know what the FreePBX sip configuration screen looks like. We need to know what changes are actually happening in sip.conf (after includes have been resolved).