Call needs to put on hold first before having audio

Hi Guys,

We have just installed our FreePBX distro. All is set,(extensions, Trunk, Inbound/Outbound route, etc.) We can now also dial numbers and extensions thru softphone.
The problem was, it seems that there is problem with the audio as we can’t hear each other on both end. But once we put the call on hold and resume again, audio is working as it should meaning, we are then able to hear each other. This is happening both when calling landlines and extensions.

Please help.

Thanks in advance!

Jace

Putting a call on and off hold is likely to be sending a SIP re-INVITE.

Might be worth doing a SIP debug trace and compare the IP address and ports being used in the initial INVITE message and the one after the call is taken off hold.

Could be some kind of NAT / SIP peer “can reinvite” setting thats incorrect, but I’m only guessing.

Thanks for the reply. Appreciate it.

From the asterisk cli, once the call was made we’re only getting this:

“[2013-11-06 11:27:17] WARNING[19309][C-00000050]: ccss.c:1000 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
[2013-11-06 11:27:17] WARNING[19309][C-00000050]: ccss.c:1000 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.”

Thanks,

Yes, sounds like some sort of far end NAT issue to me.

Hi SkykingOH,

Can you explain it further please…

Not really sure what could be the issue, ports 5060, 10000-20000 were opened. Is there any other port we should open? we have also put IP server out of the firewall but with no luck.

But audio is working once call is put on hold and resume.

Thanks,

Ports are open to server, what about the other end? This is a remote phone, correct?

If you could attach / link to a SIP debug trace of an example call that may help.

Hi Guys,

Here is the SIP debug result.

We have dialed ext. 1112 from ext 1111 => answered => waiting for audio (no audio) => put call on hold and resumed => audio works => hungup.

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2013.11.08 02:02:05 =~=~=~=~=~=~=~=~=~=~=~=
ssip show peerset debug offn
localhostCLI> SIP Debugging enabled
localhost
CLI>
<— SIP read from UDP:192.168.1.200:4568 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.1.39:5060;branch=z9hG4bK1c109a53;rport=5060
Contact: sip:[email protected]:4568;rinstance=69a9f233d184a028
To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978
From: "pbx1"sip:[email protected];tag=as12209a93
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: hop: sip:[email protected]:4568;rinstance=69a9f233d184a028
– SIP/1112-000000ea is ringing

localhost*CLI>
<— SIP read from UDP:192.168.1.200:4568 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.39:5060;branch=z9hG4bK1c109a53;rport=5060
Contact: sip:[email protected]:4568;rinstance=69a9f233d184a028
To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978
From: "pbx1"sip:[email protected];tag=as12209a93
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 235

v=0
o=- 4 2 IN IP4 192.168.1.38
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.200
t=0 0
m=audio 38408 RTP/AVP 0 8 18 101
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
— (11 headers 11 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.200:38408
list_route: hop: sip:[email protected]:4568;rinstance=69a9f233d184a028
set_destination: Parsing sip:[email protected]:4568;rinstance=69a9f233d184a028 for address/port to send to
set_destination: set destination to 192.168.1.200:4568
Transmitting (NAT) to 192.168.1.200:4568:
ACK sip:[email protected]:4568;rinstance=69a9f233d184a028 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.39:5060;branch=z9hG4bK3d30d6f1;rport
Max-Forwards: 70
From: “pbx1” sip:[email protected];tag=as12209a93 To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978 Contact: sip:[email protected]:5060 Call-ID: [email protected]:5060 CSeq: 102 ACK User-Agent: FPBX-2.11.0(11.5.1) Content-Length: 0


localhost*CLI> – Connected line update to SIP/1111-000000e9 prevented.
– SIP/1112-000000ea answered SIP/1111-000000e9
Audio is at 12578
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.1.200:6348 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.200:6348;branch=z9hG4bK-d8754z-7120c37316051347-1—d8754z-;received=192.168.1.200;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 2 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Type: application/sdp Content-Length: 310
v=0 o=root 1534867306 1534867306 IN IP4 10.10.1.39 s=Asterisk PBX 11.5.1 c=IN IP4 10.10.1.39 t=0 0 m=audio 12578 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

localhost*CLI> Retransmitting #1 (NAT) to 192.168.1.200:6348:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.200:6348;branch=z9hG4bK-d8754z-7120c37316051347-1—d8754z-;received=192.168.1.200;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 2 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Type: application/sdp Content-Length: 310
v=0 o=root 1534867306 1534867306 IN IP4 10.10.1.39 s=Asterisk PBX 11.5.1 c=IN IP4 10.10.1.39 t=0 0 m=audio 12578 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv


localhost*CLI>
<— SIP read from UDP:192.168.1.200:6348 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:6348;rport;branch=z9hG4bK-d8754z-464cd8356655dc10-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 2 ACK
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri="sip:[email protected]",response=“aeb732c950aba764ad3c7ac65124090b”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
localhost*CLI>
<— SIP read from UDP:192.168.1.200:6348 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:6348;rport;branch=z9hG4bK-d8754z-464cd8356655dc10-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 2 ACK
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri="sip:[email protected]",response=“aeb732c950aba764ad3c7ac65124090b”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
localhostCLI> Really destroying SIP dialog ‘[email protected]:5060’ Method: NOTIFY
localhost
CLI>
<— SIP read from UDP:192.168.1.200:6348 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:6348;rport;branch=z9hG4bK-d8754z-b61eef299d2c5f65-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]:5060”,response=“396109999274c38df514032264ca53eb”,algorithm=MD5
Content-Length: 229

v=0
o=- 3 3 IN IP4 192.168.1.39
s=CounterPath eyeBeam 1.5
c=IN IP4 0.0.0.0
t=0 0
m=audio 54094 RTP/AVP 0 8 18 101
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendonly
<------------->
— (13 headers 11 lines) —
Sending to 192.168.1.200:6348 (NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 0.0.0.0:54094

<— Transmitting (NAT) to 192.168.1.200:6348 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.200:6348;branch=z9hG4bK-d8754z-b61eef299d2c5f65-1—d8754z-;received=192.168.1.200;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 3 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Length: 0

<------------>
Audio is at 12578
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.1.200:6348 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.200:6348;branch=z9hG4bK-d8754z-b61eef299d2c5f65-1—d8754z-;received=192.168.1.200;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 3 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Type: application/sdp Content-Length: 310

v=0 o=root 1534867306 1534867307 IN IP4 10.10.1.39 s=Asterisk PBX 11.5.1 c=IN IP4 10.10.1.39 t=0 0 m=audio 12578 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly

<------------>
– Started music on hold, class ‘default’, on SIP/1112-000000ea
localhostCLI> > 0x7f2a3409a1a0 – Probation passed - setting RTP source address to 192.168.1.200:38408
localhost
CLI> Retransmitting #1 (NAT) to 192.168.1.200:6348:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.200:6348;branch=z9hG4bK-d8754z-b61eef299d2c5f65-1—d8754z-;received=192.168.1.200;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 3 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Type: application/sdp Content-Length: 310
v=0 o=root 1534867306 1534867307 IN IP4 10.10.1.39 s=Asterisk PBX 11.5.1 c=IN IP4 10.10.1.39 t=0 0 m=audio 12578 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly

localhost*CLI>
<— SIP read from UDP:192.168.1.200:6348 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:6348;rport;branch=z9hG4bK-d8754z-eb24866b4844203d-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 3 ACK
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]:5060”,response=“396109999274c38df514032264ca53eb”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
localhost*CLI>
<— SIP read from UDP:192.168.1.200:6348 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:6348;rport;branch=z9hG4bK-d8754z-a86de0287f26b851-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 4 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]:5060”,response=“396109999274c38df514032264ca53eb”,algorithm=MD5
Content-Length: 235

v=0
o=- 3 4 IN IP4 192.168.1.39
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.200
t=0 0
m=audio 54094 RTP/AVP 0 8 18 101
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
— (13 headers 11 lines) —
Sending to 192.168.1.200:6348 (NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.200:54094

<— Transmitting (NAT) to 192.168.1.200:6348 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.200:6348;branch=z9hG4bK-d8754z-a86de0287f26b851-1—d8754z-;received=192.168.1.200;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 4 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Length: 0

<------------>
Audio is at 12578
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.1.200:6348 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.200:6348;branch=z9hG4bK-d8754z-a86de0287f26b851-1—d8754z-;received=192.168.1.200;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 4 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Type: application/sdp Content-Length: 310

v=0 o=root 1534867306 1534867308 IN IP4 10.10.1.39 s=Asterisk PBX 11.5.1 c=IN IP4 10.10.1.39 t=0 0 m=audio 12578 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

<------------>
localhostCLI> – Stopped music on hold on SIP/1112-000000ea
localhost
CLI> > 0x7f29e809bbb0 – Probation passed - setting RTP source address to 192.168.1.200:54094
localhost*CLI> Retransmitting #1 (NAT) to 192.168.1.200:6348:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.200:6348;branch=z9hG4bK-d8754z-a86de0287f26b851-1—d8754z-;received=192.168.1.200;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 4 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Type: application/sdp Content-Length: 310
v=0 o=root 1534867306 1534867308 IN IP4 10.10.1.39 s=Asterisk PBX 11.5.1 c=IN IP4 10.10.1.39 t=0 0 m=audio 12578 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv


localhost*CLI>
<— SIP read from UDP:192.168.1.200:6348 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:6348;rport;branch=z9hG4bK-d8754z-8a76ec35725cd502-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 4 ACK
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]:5060”,response=“396109999274c38df514032264ca53eb”,algorithm=MD5
Content-Length: 0

<------------->
localhostCLI> — (11 headers 0 lines) —
localhost
CLI>
<— SIP read from UDP:192.168.1.200:6348 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:6348;rport;branch=z9hG4bK-d8754z-8a76ec35725cd502-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 4 ACK
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]:5060”,response=“396109999274c38df514032264ca53eb”,algorithm=MD5
Content-Length: 0


localhost*CLI>
<— SIP read from UDP:192.168.1.200:6348 —>
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:6348;rport;branch=z9hG4bK-d8754z-807fde688d5a1701-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 5 BYE
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]:5060”,response=“bc1c006b4e286c384689cbdd2eb49ef9”,algorithm=MD5
Reason: SIP;description="User Hung Up"
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.200:6348 (NAT)
localhost*CLI> Scheduling destruction of SIP dialog ‘OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.’ in 18112 ms (Method: BYE)

<— Transmitting (NAT) to 192.168.1.200:6348 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.200:6348;branch=z9hG4bK-d8754z-807fde688d5a1701-1—d8754z-;received=192.168.1.200;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 5 BYE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0

<------------>
localhostCLI> – Executing [[email protected]:1] Macro(“SIP/1111-000000e9”, “hangupcall,”) in new stack
localhost
CLI> – Executing [[email protected]:1] GotoIf(“SIP/1111-000000e9”, “1?theend”) in new stack
localhostCLI> – Goto (macro-hangupcall,s,3)
localhost
CLI> – Executing [[email protected]:3] ExecIf(“SIP/1111-000000e9”, “0?Set(CDR(recordingfile)=)”) in new stack
localhostCLI> – Executing [[email protected]:4] Hangup(“SIP/1111-000000e9”, “”) in new stack
localhost
CLI> == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/1111-000000e9’ in macro 'hangupcall’
localhostCLI> == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/1111-000000e9’
localhost
CLI> Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 18304 ms (Method: INVITE)
localhostCLI> localhostCLI> set_destination: Parsing sip:[email protected]:4568;rinstance=69a9f233d184a028 for address/port to send to
localhostCLI> set_destination: set destination to 192.168.1.200:4568
localhost
CLI> localhost*CLI> Reliably Transmitting (NAT) to 192.168.1.200:4568:
BYE sip:[email protected]:4568;rinstance=69a9f233d184a028 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.39:5060;branch=z9hG4bK4172a22c;rport
Max-Forwards: 70
From: “pbx1” sip:[email protected];tag=as12209a93
To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: FPBX-2.11.0(11.5.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


localhostCLI> localhostCLI> == Spawn extension (macro-dial-one, s, 42) exited non-zero on ‘SIP/1111-000000e9’ in macro 'dial-one’
localhostCLI> == Spawn extension (macro-exten-vm, s, 16) exited non-zero on ‘SIP/1111-000000e9’ in macro 'exten-vm’
localhost
CLI> == Spawn extension (from-internal, 1112, 2) exited non-zero on 'SIP/1111-000000e9’
localhostCLI> localhostCLI> localhostCLI> localhostCLI> Retransmitting #1 (NAT) to 192.168.1.200:4568:
BYE sip:[email protected]:4568;rinstance=69a9f233d184a028 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.39:5060;branch=z9hG4bK4172a22c;rport
Max-Forwards: 70
From: “pbx1” sip:[email protected];tag=as12209a93
To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: FPBX-2.11.0(11.5.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<— SIP read from UDP:192.168.1.200:4568 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.39:5060;branch=z9hG4bK4172a22c;rport=5060
Contact: sip:[email protected]:4568;rinstance=69a9f233d184a028
To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978
From: "pbx1"sip:[email protected];tag=as12209a93
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: INVITE
localhost*CLI>
<— SIP read from UDP:192.168.1.200:4568 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.39:5060;branch=z9hG4bK4172a22c;rport=5060
Contact: sip:[email protected]:4568;rinstance=69a9f233d184a028
To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978
From: "pbx1"sip:[email protected];tag=as12209a93
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 0

<------------->

BTW you missed out the initial INVITE which is kind of critical to determining where the audio is going prior to the hold.
The initial INVITE didn’t go direct between the end points did it, hence why its missing from your trace? If so then thats probably your issue.

We don’t really need the 1xx messages (180 Ringing etc) or the ACK’s.
Whats most interesting is the SDP content which is included in the various INVITE’s and 200 OK’s.

Can you confirm the IP addresses just for context. e.g.

  • xt1112 = 10.10.1.39
  • xt1111 = 192.168.1.39?
  • FreePBX = 192.168.1.200?

This link may be helpful to understand the SDP content: http://en.wikipedia.org/wiki/Session_Description_Protocol#Session_description

Initial 200 OK
o=- 4 2 IN IP4 192.168.1.38
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.200
m=audio 38408 RTP/AVP 0 8 18 101

  • Can 10.10.1.39 definitely route UDP packets to 192.168.1.200 on port 38408?

INVITE to place call on hold:
o=- 3 3 IN IP4 192.168.1.39
s=CounterPath eyeBeam 1.5
c=IN IP4 0.0.0.0

*Note the 0.0.0.0 means stop sending audio

INVITE to take call off hold:
o=- 3 4 IN IP4 192.168.1.39
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.200
m=audio 54094 RTP/AVP 0 8 18 101

*I believe in this case the audio is going to be routed via the FreePBX on port 54094

200 OK:
o=root 1534867306 1534867308 IN IP4 10.10.1.39
s=Asterisk PBX 11.5.1
c=IN IP4 10.10.1.39
m=audio 12578 RTP/AVP 0 8 18 101

  • I assume that 192.168.1.39 is able to route to 10.10.1.39 on port 12578 hence why you have audio.

Hope this helps

Alex

Hi alexb_uk,

Thank you for that reply. Really appreciate it.

My Bad, as i have omitted some of the lines cause i’ve found it very long. Sorry for that.

If possible, would it be fine for you to look over to this trace for the second time?

And is there any other way to post the trace aside from directly posting it here in this thread?

Thanks,

J

IP’s:
2xx.218.xxx.39 = external freepbx server
10.10.1.39 = internal freepbx server
192.168.1.39 = xt1111 internal ip
192.168.1.38? = xt1112 internal ip (not sure)
1xx.53.xxx.82 = external ip of both extensions

We have also tried the rtp debug unfortunately, we’re not getting any reply unless we put the call on hold which gives us a “sent & got rtp pocket” from both extensions.

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2013.11.08 02:02:05 =~=~=~=~=~=~=~=~=~=~=~=
ssip show peerset debug offn
localhostCLI> SIP Debugging enabled
localhost
CLI>
localhost*CLI>
<— SIP read from UDP:1xx.53.xxx.82:6348 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;rport;branch=z9hG4bK-d8754z-e931df28b8119e5c-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 264

v=0
o=- 3 2 IN IP4 192.168.1.39
s=CounterPath eyeBeam 1.5
c=IN IP4 1xx.53.xxx.82
t=0 0
m=audio 54094 RTP/AVP 107 0 8 18 101
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
— (12 headers 12 lines) —
Sending to 1xx.53.xxx.82:6348 (NAT)
localhostCLI> Sending to 1xx.53.xxx.82:6348 (NAT)
localhost
CLI> Using INVITE request as basis request - OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
localhostCLI> Found peer ‘1111’ for ‘1111’ from 1xx.53.xxx.82:6348
localhost
CLI>
<— Reliably Transmitting (NAT) to 1xx.53.xxx.82:6348 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;branch=z9hG4bK-d8754z-e931df28b8119e5c-1—d8754z-;received=1xx.53.xxx.82;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18
To: “1112” sip:[email protected]:5060;tag=as4c55e2ef
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3455579c"
Content-Length: 0

<------------>
localhostCLI> Scheduling destruction of SIP dialog ‘OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.’ in 18112 ms (Method: INVITE)
localhost
CLI>
<— SIP read from UDP:1xx.53.xxx.82:6348 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;rport;branch=z9hG4bK-d8754z-e931df28b8119e5c-1—d8754z-
To: “1112” sip:[email protected]:5060;tag=as4c55e2ef
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 1 ACK
Content-Length: 0

<------------->
localhostCLI> — (7 headers 0 lines) —
localhost
CLI>
<— SIP read from UDP:1xx.53.xxx.82:6348 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;rport;branch=z9hG4bK-d8754z-7120c37316051347-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]”,response=“aeb732c950aba764ad3c7ac65124090b”,algorithm=MD5
Content-Length: 264

v=0
o=- 3 2 IN IP4 192.168.1.39
s=CounterPath eyeBeam 1.5
c=IN IP4 1xx.53.xxx.82
t=0 0
m=audio 54094 RTP/AVP 107 0 8 18 101
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
localhostCLI> — (13 headers 12 lines) —
localhost
CLI> Sending to 1xx.53.xxx.82:6348 (NAT)
localhostCLI> Using INVITE request as basis request - OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
localhost
CLI> Found peer ‘1111’ for ‘1111’ from 1xx.53.xxx.82:6348
localhostCLI> == Using SIP RTP TOS bits 184
localhost
CLI> == Using SIP RTP CoS mark 5
localhostCLI> Found RTP audio format 107
localhost
CLI> Found RTP audio format 0
localhostCLI> Found RTP audio format 8
localhost
CLI> Found RTP audio format 18
localhostCLI> Found RTP audio format 101
localhost
CLI> Found unknown media description format BV32 for ID 107
localhostCLI> Found audio description format G729 for ID 18
localhost
CLI> Found audio description format telephone-event for ID 101
localhostCLI> Capabilities: us - (gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
localhost
CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
localhostCLI> Peer audio RTP is at port 1xx.53.xxx.82:54094
localhost
CLI> Looking for 1112 in from-internal (domain 10.10.1.39)
localhostCLI> localhostCLI> list_route: hop: sip:[email protected]:6348
localhost*CLI>
<— Transmitting (NAT) to 1xx.53.xxx.82:6348 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;branch=z9hG4bK-d8754z-7120c37316051347-1—d8754z-;received=1xx.53.xxx.82;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18
To: “1112” sip:[email protected]:5060
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
localhostCLI> – Executing [[email protected]:1] Set(“SIP/1111-000000e9”, “__RINGTIMER=15”) in new stack
localhost
CLI> – Executing [[email protected]:2] Macro(“SIP/1111-000000e9”, “exten-vm,novm,1112,0,0,0”) in new stack
localhostCLI> – Executing [[email protected]:1] Macro(“SIP/1111-000000e9”, “user-callerid,”) in new stack
localhost
CLI> – Executing [[email protected]:1] Set(“SIP/1111-000000e9”, “TOUCH_MONITOR=1383847405.233”) in new stack
localhostCLI> – Executing [[email protected]:2] Set(“SIP/1111-000000e9”, “AMPUSER=1111”) in new stack
localhost
CLI> – Executing [[email protected]:3] GotoIf(“SIP/1111-000000e9”, “0?report”) in new stack
localhostCLI> – Executing [[email protected]:4] ExecIf(“SIP/1111-000000e9”, “1?Set(REALCALLERIDNUM=1111)”) in new stack
localhost
CLI> – Executing [[email protected]:5] Set(“SIP/1111-000000e9”, “AMPUSER=1111”) in new stack
localhostCLI> – Executing [[email protected]:6] Set(“SIP/1111-000000e9”, “AMPUSERCIDNAME=pbx1”) in new stack
localhost
CLI> – Executing [[email protected]:7] GotoIf(“SIP/1111-000000e9”, “0?report”) in new stack
localhostCLI> – Executing [[email protected]:8] Set(“SIP/1111-000000e9”, “AMPUSERCID=1111”) in new stack
localhost
CLI> – Executing [[email protected]:9] Set(“SIP/1111-000000e9”, “__DIAL_OPTIONS=Ttr”) in new stack
localhostCLI> – Executing [[email protected]:10] Set(“SIP/1111-000000e9”, “CALLERID(all)=“pbx1” <1111>”) in new stack
localhost
CLI> – Executing [[email protected]:11] GotoIf(“SIP/1111-000000e9”, “0?limit”) in new stack
localhostCLI> – Executing [[email protected]:12] ExecIf(“SIP/1111-000000e9”, “0?Set(GROUP(concurrency_limit)=1111)”) in new stack
localhost
CLI> – Executing [[email protected]:13] ExecIf(“SIP/1111-000000e9”, “0?Set(CHANNEL(language)=)”) in new stack
localhostCLI> – Executing [[email protected]:14] GosubIf(“SIP/1111-000000e9”, “7?sub-ccss,s,1(macro-exten-vm,1112)”) in new stack
localhost
CLI> – Executing [[email protected]:1] ExecIf(“SIP/1111-000000e9”, “0?Return()”) in new stack
localhostCLI> – Executing [[email protected]:2] Set(“SIP/1111-000000e9”, “CCSS_SETUP=TRUE”) in new stack
localhost
CLI> – Executing [[email protected]:3] GosubIf(“SIP/1111-000000e9”, “0?monitor_config,1(macro-exten-vm,1112):monitor_default,1(macro-exten-vm,1112)”) in new stack
localhostCLI> – Executing [[email protected]:1] GotoIf(“SIP/1111-000000e9”, “1?is_exten”) in new stack
localhost
CLI> – Goto (sub-ccss,monitor_default,4)
localhostCLI> – Executing [[email protected]:4] Set(“SIP/1111-000000e9”, “CALLCOMPLETION(cc_monitor_policy)=generic”) in new stack
localhost
CLI> – Executing [[email protected]:5] Set(“SIP/1111-000000e9”, “CALLCOMPLETION(cc_max_monitors)=5”) in new stack
localhostCLI> – Executing [[email protected]:6] Return(“SIP/1111-000000e9”, “TRUE”) in new stack
localhost
CLI> – Executing [[email protected]:4] GosubIf(“SIP/1111-000000e9”, “7?agent_config,1():agent_default,1()”) in new stack
localhostCLI> – Executing [[email protected]:1] Set(“SIP/1111-000000e9”, “CALLCOMPLETION(cc_agent_policy)=generic”) in new stack
localhost
CLI> – Executing [[email protected]:2] Set(“SIP/1111-000000e9”, “CALLCOMPLETION(cc_offer_timer)=30”) in new stack
localhostCLI> – Executing [[email protected]:3] Set(“SIP/1111-000000e9”, “CALLCOMPLETION(ccbs_available_timer)=”) in new stack
localhost
CLI> – Executing [[email protected]:4] Set(“SIP/1111-000000e9”, “CALLCOMPLETION(ccnr_available_timer)=”) in new stack
localhostCLI> – Executing [[email protected]:5] Set(“SIP/1111-000000e9”, “CALLCOMPLETION(cc_callback_macro)=ccss-default”) in new stack
localhost
CLI> [2013-11-07 10:03:25] WARNING[23379][C-0000009f]: ccss.c:1000 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
localhostCLI> – Executing [[email protected]:6] ExecIf(“SIP/1111-000000e9”, “1?Set(CALLCOMPLETION(cc_recall_timer)=)”) in new stack
localhost
CLI> – Executing [[email protected]:7] ExecIf(“SIP/1111-000000e9”, “1?Set(CALLCOMPLETION(cc_max_agents)=)”) in new stack
localhostCLI> – Executing [[email protected]:8] ExecIf(“SIP/1111-000000e9”, “0?Set(CALLCOMPLETION(cc_agent_dialstring)=Local/[email protected])”) in new stack
localhost
CLI> – Executing [[email protected]:9] Set(“SIP/1111-000000e9”, “CALLCOMPLETION(cc_callback_macro)=ccss-default”) in new stack
localhostCLI> [2013-11-07 10:03:25] WARNING[23379][C-0000009f]: ccss.c:1000 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
localhost
CLI> – Executing [[email protected]:10] Return(“SIP/1111-000000e9”, “”) in new stack
localhostCLI> – Executing [[email protected]:5] Set(“SIP/1111-000000e9”, “DB(AMPUSER/1111/ccss/last_number)=1112”) in new stack
localhost
CLI> – Executing [[email protected]:6] Return(“SIP/1111-000000e9”, “”) in new stack
localhostCLI> – Executing [[email protected]:15] GotoIf(“SIP/1111-000000e9”, “0?continue”) in new stack
localhost
CLI> – Executing [[email protected]:16] Set(“SIP/1111-000000e9”, “__TTL=64”) in new stack
localhostCLI> – Executing [[email protected]:17] GotoIf(“SIP/1111-000000e9”, “1?continue”) in new stack
localhost
CLI> – Goto (macro-user-callerid,s,28)
– Executing [[email protected]:28] Set(“SIP/1111-000000e9”, “CALLERID(number)=1111”) in new stack
– Executing [[email protected]:29] Set(“SIP/1111-000000e9”, “CALLERID(name)=pbx1”) in new stack
– Executing [[email protected]:30] Set(“SIP/1111-000000e9”, “CDR(cnum)=1111”) in new stack
localhostCLI> – Executing [[email protected]:31] Set(“SIP/1111-000000e9”, “CDR(cnam)=pbx1”) in new stack
localhost
CLI> – Executing [[email protected]:32] Set(“SIP/1111-000000e9”, “CHANNEL(language)=en”) in new stack
localhostCLI> – Executing [[email protected]:2] Set(“SIP/1111-000000e9”, “RingGroupMethod=none”) in new stack
localhost
CLI> – Executing [[email protected]:3] Set(“SIP/1111-000000e9”, “__EXTTOCALL=1112”) in new stack
localhostCLI> – Executing [[email protected]:4] Set(“SIP/1111-000000e9”, “__PICKUPMARK=1112”) in new stack
localhost
CLI> – Executing [[email protected]:5] Set(“SIP/1111-000000e9”, “RT=”) in new stack
localhostCLI> – Executing [[email protected]:6] ExecIf(“SIP/1111-000000e9”, “0?Macro(vm,novm,DIRECTDIAL,)”) in new stack
localhost
CLI> – Executing [[email protected]:7] ExecIf(“SIP/1111-000000e9”, “0?MacroExit()”) in new stack
localhostCLI> – Executing [[email protected]:8] Gosub(“SIP/1111-000000e9”, “sub-record-check,s,1(exten,1112,)”) in new stack
localhost
CLI> – Executing [[email protected]:1] Set(“SIP/1111-000000e9”, “REC_POLICY_MODE_SAVE=”) in new stack
localhostCLI> – Executing [[email protected]:2] GotoIf(“SIP/1111-000000e9”, “1?check”) in new stack
localhost
CLI> – Goto (sub-record-check,s,7)
localhostCLI> – Executing [[email protected]:7] Set(“SIP/1111-000000e9”, “__MON_FMT=wav”) in new stack
localhost
CLI> – Executing [[email protected]:8] GotoIf(“SIP/1111-000000e9”, “1?next”) in new stack
localhostCLI> – Goto (sub-record-check,s,11)
localhost
CLI> – Executing [[email protected]:11] ExecIf(“SIP/1111-000000e9”, “0?Return()”) in new stack
localhostCLI> – Executing [[email protected]:12] ExecIf(“SIP/1111-000000e9”, “0?Set(__REC_POLICY_MODE=)”) in new stack
localhost
CLI> – Executing [[email protected]:13] GotoIf(“SIP/1111-000000e9”, “0?exten,1”) in new stack
localhostCLI> – Executing [[email protected]:14] Set(“SIP/1111-000000e9”, “__REC_STATUS=INITIALIZED”) in new stack
localhost
CLI> – Executing [[email protected]:15] Set(“SIP/1111-000000e9”, “NOW=1383847405”) in new stack
localhostCLI> – Executing [[email protected]:16] Set(“SIP/1111-000000e9”, “__DAY=07”) in new stack
localhost
CLI> – Executing [[email protected]:17] Set(“SIP/1111-000000e9”, “__MONTH=11”) in new stack
localhostCLI> – Executing [[email protected]:18] Set(“SIP/1111-000000e9”, “__YEAR=2013”) in new stack
localhost
CLI> – Executing [[email protected]:19] Set(“SIP/1111-000000e9”, “__TIMESTR=20131107-100325”) in new stack
localhostCLI> – Executing [[email protected]:20] Set(“SIP/1111-000000e9”, “__FROMEXTEN=1111”) in new stack
localhost
CLI> – Executing [[email protected]:21] Set(“SIP/1111-000000e9”, “__CALLFILENAME=exten-1112-1111-20131107-100325-1383847405.233”) in new stack
localhostCLI> – Executing [[email protected]:22] Goto(“SIP/1111-000000e9”, “exten,1”) in new stack
localhost
CLI> – Goto (sub-record-check,exten,1)
localhostCLI> – Executing [[email protected]:1] GotoIf(“SIP/1111-000000e9”, “0?callee”) in new stack
localhost
CLI> – Executing [[email protected]:2] Set(“SIP/1111-000000e9”, “__REC_POLICY_MODE=dontcare”) in new stack
localhostCLI> – Executing [[email protected]:3] GotoIf(“SIP/1111-000000e9”, “1?caller”) in new stack
localhost
CLI> – Goto (sub-record-check,exten,10)
localhostCLI> – Executing [[email protected]:10] Set(“SIP/1111-000000e9”, “__REC_POLICY_MODE=dontcare”) in new stack
localhost
CLI> – Executing [[email protected]:11] GosubIf(“SIP/1111-000000e9”, “0?record,1(exten,1112,1111)”) in new stack
localhostCLI> – Executing [[email protected]:12] Return(“SIP/1111-000000e9”, “”) in new stack
localhost
CLI> – Executing [[email protected]:9] GotoIf(“SIP/1111-000000e9”, “1?macrodial”) in new stack
localhostCLI> – Goto (macro-exten-vm,s,15)
localhost
CLI> – Executing [[email protected]:15] GosubIf(“SIP/1111-000000e9”, “0?clrheader,1()”) in new stack
localhostCLI> – Executing [[email protected]:16] Macro(“SIP/1111-000000e9”, “dial-one,Ttr,1112”) in new stack
localhost
CLI> – Executing [[email protected]:1] Set(“SIP/1111-000000e9”, “DEXTEN=1112”) in new stack
localhostCLI> – Executing [[email protected]:2] Set(“SIP/1111-000000e9”, “DIALSTATUS_CW=”) in new stack
localhost
CLI> – Executing [[email protected]:3] GosubIf(“SIP/1111-000000e9”, “0?screen,1()”) in new stack
localhostCLI> – Executing [[email protected]:4] GosubIf(“SIP/1111-000000e9”, “0?cf,1()”) in new stack
localhost
CLI> – Executing [[email protected]:5] GotoIf(“SIP/1111-000000e9”, “1?skip1”) in new stack
localhostCLI> – Goto (macro-dial-one,s,8)
localhost
CLI> – Executing [[email protected]:8] GotoIf(“SIP/1111-000000e9”, “0?nodial”) in new stack
localhostCLI> – Executing [[email protected]:9] GotoIf(“SIP/1111-000000e9”, “0?continue”) in new stack
localhost
CLI> – Executing [[email protected]:10] Set(“SIP/1111-000000e9”, “EXTHASCW=ENABLED”) in new stack
localhostCLI> – Executing [[email protected]:11] GotoIf(“SIP/1111-000000e9”, “0?next1:cwinusebusy”) in new stack
localhost
CLI> – Goto (macro-dial-one,s,23)
localhostCLI> – Executing [[email protected]:23] GotoIf(“SIP/1111-000000e9”, “1?next3:continue”) in new stack
localhost
CLI> – Goto (macro-dial-one,s,24)
localhostCLI> – Executing [[email protected]:24] ExecIf(“SIP/1111-000000e9”, “0?Set(DIALSTATUS_CW=BUSY)”) in new stack
localhost
CLI> – Executing [[email protected]:25] GotoIf(“SIP/1111-000000e9”, “0?nodial”) in new stack
localhostCLI> – Executing [[email protected]:26] GosubIf(“SIP/1111-000000e9”, “1?dstring,1():dlocal,1()”) in new stack
localhost
CLI> – Executing [[email protected]:1] Set(“SIP/1111-000000e9”, “DSTRING=”) in new stack
localhostCLI> – Executing [[email protected]:2] Set(“SIP/1111-000000e9”, “DEVICES=1112”) in new stack
localhost
CLI> – Executing [[email protected]:3] ExecIf(“SIP/1111-000000e9”, “0?Return()”) in new stack
localhostCLI> – Executing [[email protected]:4] ExecIf(“SIP/1111-000000e9”, “0?Set(DEVICES=112)”) in new stack
localhost
CLI> – Executing [[email protected]:5] Set(“SIP/1111-000000e9”, “LOOPCNT=1”) in new stack
localhostCLI> – Executing [[email protected]:6] Set(“SIP/1111-000000e9”, “ITER=1”) in new stack
localhost
CLI> – Executing [[email protected]:7] Set(“SIP/1111-000000e9”, “THISDIAL=SIP/1112”) in new stack
localhostCLI> – Executing [[email protected]:8] GosubIf(“SIP/1111-000000e9”, “1?zap2dahdi,1()”) in new stack
localhost
CLI> – Executing [[email protected]:1] ExecIf(“SIP/1111-000000e9”, “0?Return()”) in new stack
localhostCLI> – Executing [[email protected]:2] Set(“SIP/1111-000000e9”, “NEWDIAL=”) in new stack
localhost
CLI> – Executing [[email protected]:3] Set(“SIP/1111-000000e9”, “LOOPCNT2=1”) in new stack
localhost*CLI> – Executing [[email protected]:4] Set(“SIP/1111-000000e9”, “ITER2=1”) in new stack
– Executing [[email protected]:5] Set(“SIP/1111-000000e9”, “THISPART2=SIP/1112”) in new stack
– Executing [[email protected]:6] ExecIf(“SIP/1111-000000e9”, “0?Set(THISPART2=DAHDI/1112)”) in new stack
– Executing [[email protected]:7] Set(“SIP/1111-000000e9”, “NEWDIAL=SIP/1112&”) in new stack
– Executing [[email protected]:8] Set(“SIP/1111-000000e9”, “ITER2=2”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/1111-000000e9”, “0?begin2”) in new stack
– Executing [[email protected]:10] Set(“SIP/1111-000000e9”, “THISDIAL=SIP/1112”) in new stack
– Executing [[email protected]:11] Return(“SIP/1111-000000e9”, “”) in new stack
– Executing [[email protected]:9] Set(“SIP/1111-000000e9”, “DSTRING=SIP/1112&”) in new stack
– Executing [[email protected]:10] Set(“SIP/1111-000000e9”, “ITER=2”) in new stack
– Executing [[email protected]:11] GotoIf(“SIP/1111-000000e9”, “0?begin”) in new stack
– Executing [[email protected]:12] Set(“SIP/1111-000000e9”, “DSTRING=SIP/1112”) in new stack
– Executing [[email protected]:13] Return(“SIP/1111-000000e9”, “”) in new stack
– Executing [[email protected]:27] GotoIf(“SIP/1111-000000e9”, “0?nodial”) in new stack
– Executing [[email protected]:28] GotoIf(“SIP/1111-000000e9”, “0?skiptrace”) in new stack
– Executing [[email protected]:29] GosubIf(“SIP/1111-000000e9”, “1?ctset,1():ctclear,1()”) in new stack
– Executing [[email protected]:1] Set(“SIP/1111-000000e9”, “DB(CALLTRACE/1112)=1111”) in new stack
– Executing [[email protected]:2] Return(“SIP/1111-000000e9”, “”) in new stack
– Executing [[email protected]:30] Set(“SIP/1111-000000e9”, “D_OPTIONS=Ttr”) in new stack
– Executing [[email protected]:31] ExecIf(“SIP/1111-000000e9”, “0?SIPAddHeader(Alert-Info: )”) in new stack
– Executing [[email protected]:32] ExecIf(“SIP/1111-000000e9”, “0?SIPAddHeader()”) in new stack
– Executing [[email protected]:33] ExecIf(“SIP/1111-000000e9”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [[email protected]:34] GosubIf(“SIP/1111-000000e9”, “0?qwait,1()”) in new stack
– Executing [[email protected]:35] Set(“SIP/1111-000000e9”, “__CWIGNORE=”) in new stack
– Executing [[email protected]:36] Set(“SIP/1111-000000e9”, “__KEEPCID=TRUE”) in new stack
– Executing [[email protected]:37] GotoIf(“SIP/1111-000000e9”, “0?usegoto,1”) in new stack
– Executing [[email protected]:38] GotoIf(“SIP/1111-000000e9”, “0?godial”) in new stack
– Executing [[email protected]:39] Set(“SIP/1111-000000e9”, “CONNECTEDLINE(name,i)=Ramil”) in new stack
– Executing [[email protected]:40] Set(“SIP/1111-000000e9”, “CONNECTEDLINE(num)=1112”) in new stack
– Executing [[email protected]:41] Set(“SIP/1111-000000e9”, “D_OPTIONS=TtrI”) in new stack
– Executing [[email protected]:42] Dial(“SIP/1111-000000e9”, “SIP/1112,TtrI”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 14540
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100010 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 1xx.53.xxx.82:4568:
INVITE sip:[email protected]:4568;rinstance=69a9f233d184a028 SIP/2.0
Via: SIP/2.0/UDP 2xx.218.xxx.39:5060;branch=z9hG4bK1c109a53;rport
Max-Forwards: 70
From: “pbx1” sip:[email protected];tag=as12209a93 To: sip:[email protected]:4568;rinstance=69a9f233d184a028 Contact: sip:[email protected]:5060 Call-ID: [email protected]:5060 CSeq: 102 INVITE User-Agent: FPBX-2.11.0(11.5.1) Date: Thu, 07 Nov 2013 18:03:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 407

v=0 o=root 2139640530 2139640530 IN IP4 2xx.218.xxx.39 s=Asterisk PBX 11.5.1 c=IN IP4 2xx.218.xxx.39 t=0 0 m=audio 14540 RTP/AVP 0 8 3 110 18 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

-- Called SIP/1112

<— Transmitting (NAT) to 1xx.53.xxx.82:6348 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;branch=z9hG4bK-d8754z-7120c37316051347-1—d8754z-;received=1xx.53.xxx.82;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 2 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Length: 0
<------------>
– Connected line update to SIP/1111-000000e9 prevented.

localhost*CLI> Retransmitting #1 (NAT) to 1xx.53.xxx.82:4568:
INVITE sip:[email protected]:4568;rinstance=69a9f233d184a028 SIP/2.0
Via: SIP/2.0/UDP 2xx.218.xxx.39:5060;branch=z9hG4bK1c109a53;rport
Max-Forwards: 70
From: “pbx1” sip:[email protected];tag=as12209a93 To: sip:[email protected]:4568;rinstance=69a9f233d184a028 Contact: sip:[email protected]:5060 Call-ID: [email protected]:5060 CSeq: 102 INVITE User-Agent: FPBX-2.11.0(11.5.1) Date: Thu, 07 Nov 2013 18:03:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 407
v=0 o=root 2139640530 2139640530 IN IP4 2xx.218.xxx.39 s=Asterisk PBX 11.5.1 c=IN IP4 2xx.218.xxx.39 t=0 0 m=audio 14540 RTP/AVP 0 8 3 110 18 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv


localhost*CLI>
<— SIP read from UDP:1xx.53.xxx.82:4568 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2xx.218.xxx.39:5060;branch=z9hG4bK1c109a53;rport=5060
Contact: sip:[email protected]:4568;rinstance=69a9f233d184a028
To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978
From: "pbx1"sip:[email protected];tag=as12209a93
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: hop: sip:[email protected]:4568;rinstance=69a9f233d184a028
– SIP/1112-000000ea is ringing

<— Transmitting (NAT) to 1xx.53.xxx.82:6348 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;branch=z9hG4bK-d8754z-7120c37316051347-1—d8754z-;received=1xx.53.xxx.82;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 2 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Length: 0
<------------>
localhost*CLI>
<— SIP read from UDP:1xx.53.xxx.82:4568 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2xx.218.xxx.39:5060;branch=z9hG4bK1c109a53;rport=5060
Contact: sip:[email protected]:4568;rinstance=69a9f233d184a028
To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978
From: "pbx1"sip:[email protected];tag=as12209a93
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: hop: sip:[email protected]:4568;rinstance=69a9f233d184a028
– SIP/1112-000000ea is ringing
localhost*CLI>
<— SIP read from UDP:1xx.53.xxx.82:4568 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2xx.218.xxx.39:5060;branch=z9hG4bK1c109a53;rport=5060
Contact: sip:[email protected]:4568;rinstance=69a9f233d184a028
To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978
From: "pbx1"sip:[email protected];tag=as12209a93
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 235

v=0
o=- 4 2 IN IP4 192.168.1.38
s=CounterPath eyeBeam 1.5
c=IN IP4 1xx.53.xxx.82
t=0 0
m=audio 38408 RTP/AVP 0 8 18 101
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
— (11 headers 11 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1xx.53.xxx.82:38408
list_route: hop: sip:[email protected]:4568;rinstance=69a9f233d184a028
set_destination: Parsing sip:[email protected]:4568;rinstance=69a9f233d184a028 for address/port to send to
set_destination: set destination to 1xx.53.xxx.82:4568
Transmitting (NAT) to 1xx.53.xxx.82:4568:
ACK sip:[email protected]:4568;rinstance=69a9f233d184a028 SIP/2.0
Via: SIP/2.0/UDP 2xx.218.xxx.39:5060;branch=z9hG4bK3d30d6f1;rport
Max-Forwards: 70
From: “pbx1” sip:[email protected];tag=as12209a93 To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978 Contact: sip:[email protected]:5060 Call-ID: [email protected]:5060 CSeq: 102 ACK User-Agent: FPBX-2.11.0(11.5.1) Content-Length: 0


localhost*CLI> – Connected line update to SIP/1111-000000e9 prevented.
– SIP/1112-000000ea answered SIP/1111-000000e9
Audio is at 12578
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 1xx.53.xxx.82:6348 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;branch=z9hG4bK-d8754z-7120c37316051347-1—d8754z-;received=1xx.53.xxx.82;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 2 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Type: application/sdp Content-Length: 310
v=0 o=root 1534867306 1534867306 IN IP4 2xx.218.xxx.39 s=Asterisk PBX 11.5.1 c=IN IP4 2xx.218.xxx.39 t=0 0 m=audio 12578 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

<------------>
localhostCLI> localhostCLI> Retransmitting #1 (NAT) to 1xx.53.xxx.82:6348:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;branch=z9hG4bK-d8754z-7120c37316051347-1—d8754z-;received=1xx.53.xxx.82;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 2 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Type: application/sdp Content-Length: 310
v=0 o=root 1534867306 1534867306 IN IP4 2xx.218.xxx.39 s=Asterisk PBX 11.5.1 c=IN IP4 2xx.218.xxx.39 t=0 0 m=audio 12578 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

localhost*CLI>
<— SIP read from UDP:1xx.53.xxx.82:6348 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;rport;branch=z9hG4bK-d8754z-464cd8356655dc10-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 2 ACK
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]”,response=“aeb732c950aba764ad3c7ac65124090b”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
localhost*CLI>
<— SIP read from UDP:1xx.53.xxx.82:6348 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;rport;branch=z9hG4bK-d8754z-464cd8356655dc10-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 2 ACK
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]”,response=“aeb732c950aba764ad3c7ac65124090b”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
localhostCLI> Really destroying SIP dialog ‘[email protected]:5060’ Method: NOTIFY
localhost
CLI>
<— SIP read from UDP:1xx.53.xxx.82:6348 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;rport;branch=z9hG4bK-d8754z-b61eef299d2c5f65-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]:5060”,response=“396109999274c38df514032264ca53eb”,algorithm=MD5
Content-Length: 229

v=0
o=- 3 3 IN IP4 192.168.1.39
s=CounterPath eyeBeam 1.5
c=IN IP4 0.0.0.0
t=0 0
m=audio 54094 RTP/AVP 0 8 18 101
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendonly
<------------->
— (13 headers 11 lines) —
Sending to 1xx.53.xxx.82:6348 (NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 0.0.0.0:54094

<— Transmitting (NAT) to 1xx.53.xxx.82:6348 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;branch=z9hG4bK-d8754z-b61eef299d2c5f65-1—d8754z-;received=1xx.53.xxx.82;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18
To: “1112” sip:[email protected]:5060;tag=as7788edfc
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 3 INVITE
Server: FPBX-2.11.0(11.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
Audio is at 12578
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 1xx.53.xxx.82:6348 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;branch=z9hG4bK-d8754z-b61eef299d2c5f65-1—d8754z-;received=1xx.53.xxx.82;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18
To: “1112” sip:[email protected]:5060;tag=as7788edfc
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 3 INVITE
Server: FPBX-2.11.0(11.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 310
v=0 o=root 1534867306 1534867307 IN IP4 2xx.218.xxx.39 s=Asterisk PBX 11.5.1 c=IN IP4 2xx.218.xxx.39 t=0 0 m=audio 12578 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly

<------------>
– Started music on hold, class ‘default’, on SIP/1112-000000ea
localhostCLI> > 0x7f2a3409a1a0 – Probation passed - setting RTP source address to 1xx.53.xxx.82:38408
localhost
CLI> Retransmitting #1 (NAT) to 1xx.53.xxx.82:6348:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;branch=z9hG4bK-d8754z-b61eef299d2c5f65-1—d8754z-;received=1xx.53.xxx.82;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 3 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Type: application/sdp Content-Length: 310
v=0 o=root 1534867306 1534867307 IN IP4 2xx.218.xxx.39 s=Asterisk PBX 11.5.1 c=IN IP4 2xx.218.xxx.39 t=0 0 m=audio 12578 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly


localhost*CLI>
<— SIP read from UDP:1xx.53.xxx.82:6348 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;rport;branch=z9hG4bK-d8754z-eb24866b4844203d-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 3 ACK
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]:5060”,response=“396109999274c38df514032264ca53eb”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
localhost*CLI>
<— SIP read from UDP:1xx.53.xxx.82:6348 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;rport;branch=z9hG4bK-d8754z-eb24866b4844203d-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 3 ACK
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]:5060”,response=“396109999274c38df514032264ca53eb”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
localhost*CLI>
<— SIP read from UDP:1xx.53.xxx.82:6348 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;rport;branch=z9hG4bK-d8754z-a86de0287f26b851-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 4 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]:5060”,response=“396109999274c38df514032264ca53eb”,algorithm=MD5
Content-Length: 235

v=0
o=- 3 4 IN IP4 192.168.1.39
s=CounterPath eyeBeam 1.5
c=IN IP4 1xx.53.xxx.82
t=0 0
m=audio 54094 RTP/AVP 0 8 18 101
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
— (13 headers 11 lines) —
Sending to 1xx.53.xxx.82:6348 (NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1xx.53.xxx.82:54094

<— Transmitting (NAT) to 1xx.53.xxx.82:6348 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;branch=z9hG4bK-d8754z-a86de0287f26b851-1—d8754z-;received=1xx.53.xxx.82;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18
To: “1112” sip:[email protected]:5060;tag=as7788edfc
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 4 INVITE
Server: FPBX-2.11.0(11.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
Audio is at 12578
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 1xx.53.xxx.82:6348 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;branch=z9hG4bK-d8754z-a86de0287f26b851-1—d8754z-;received=1xx.53.xxx.82;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 4 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Type: application/sdp Content-Length: 310
v=0 o=root 1534867306 1534867308 IN IP4 2xx.218.xxx.39 s=Asterisk PBX 11.5.1 c=IN IP4 2xx.218.xxx.39 t=0 0 m=audio 12578 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

<------------>
localhostCLI> – Stopped music on hold on SIP/1112-000000ea
localhost
CLI> > 0x7f29e809bbb0 – Probation passed - setting RTP source address to 1xx.53.xxx.82:54094
localhost*CLI> Retransmitting #1 (NAT) to 1xx.53.xxx.82:6348:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;branch=z9hG4bK-d8754z-a86de0287f26b851-1—d8754z-;received=1xx.53.xxx.82;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 4 INVITE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:[email protected]:5060 Content-Type: application/sdp Content-Length: 310

v=0 o=root 1534867306 1534867308 IN IP4 2xx.218.xxx.39 s=Asterisk PBX 11.5.1 c=IN IP4 2xx.218.xxx.39 t=0 0 m=audio 12578 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv


localhost*CLI>
<— SIP read from UDP:1xx.53.xxx.82:6348 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;rport;branch=z9hG4bK-d8754z-8a76ec35725cd502-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 4 ACK
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]:5060”,response=“396109999274c38df514032264ca53eb”,algorithm=MD5
Content-Length: 0

<------------->
localhostCLI> — (11 headers 0 lines) —
localhost
CLI>
<— SIP read from UDP:1xx.53.xxx.82:6348 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;rport;branch=z9hG4bK-d8754z-8a76ec35725cd502-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 4 ACK
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]:5060”,response=“396109999274c38df514032264ca53eb”,algorithm=MD5
Content-Length: 0

<------------->
localhost*CLI> — (11 headers 0 lines) —

localhost*CLI>
<— SIP read from UDP:1xx.53.xxx.82:6348 —>
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;rport;branch=z9hG4bK-d8754z-807fde688d5a1701-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:6348
To: “1112” sip:[email protected]:5060;tag=as7788edfc
From: “pbx1” sip:[email protected]:5060;tag=43274e18
Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.
CSeq: 5 BYE
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“1111”,realm=“asterisk”,nonce=“3455579c”,uri=“sip:[email protected]:5060”,response=“bc1c006b4e286c384689cbdd2eb49ef9”,algorithm=MD5
Reason: SIP;description="User Hung Up"
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 1xx.53.xxx.82:6348 (NAT)
localhost*CLI> Scheduling destruction of SIP dialog ‘OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU.’ in 18112 ms (Method: BYE)

<— Transmitting (NAT) to 1xx.53.xxx.82:6348 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1xx.53.xxx.82:6348;branch=z9hG4bK-d8754z-807fde688d5a1701-1—d8754z-;received=1xx.53.xxx.82;rport=6348
From: “pbx1” sip:[email protected]:5060;tag=43274e18 To: “1112” sip:[email protected]:5060;tag=as7788edfc Call-ID: OGMzZGViNDllYzZmODgyNDJhNThiZmI0MWY1YjNiMjU. CSeq: 5 BYE Server: FPBX-2.11.0(11.5.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0

<------------>
localhostCLI> – Executing [[email protected]:1] Macro(“SIP/1111-000000e9”, “hangupcall,”) in new stack
localhost
CLI> – Executing [[email protected]:1] GotoIf(“SIP/1111-000000e9”, “1?theend”) in new stack
localhostCLI> – Goto (macro-hangupcall,s,3)
localhost
CLI> – Executing [[email protected]:3] ExecIf(“SIP/1111-000000e9”, “0?Set(CDR(recordingfile)=)”) in new stack
localhostCLI> – Executing [[email protected]:4] Hangup(“SIP/1111-000000e9”, “”) in new stack
localhost
CLI> == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/1111-000000e9’ in macro 'hangupcall’
localhostCLI> == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/1111-000000e9’
localhost
CLI> Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 18304 ms (Method: INVITE)
localhostCLI> localhostCLI> set_destination: Parsing sip:[email protected]:4568;rinstance=69a9f233d184a028 for address/port to send to
localhostCLI> set_destination: set destination to 1xx.53.xxx.82:4568
localhost
CLI> localhost*CLI> Reliably Transmitting (NAT) to 1xx.53.xxx.82:4568:
BYE sip:[email protected]:4568;rinstance=69a9f233d184a028 SIP/2.0
Via: SIP/2.0/UDP 2xx.218.xxx.39:5060;branch=z9hG4bK4172a22c;rport
Max-Forwards: 70
From: “pbx1” sip:[email protected];tag=as12209a93
To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: FPBX-2.11.0(11.5.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


localhostCLI> localhostCLI> == Spawn extension (macro-dial-one, s, 42) exited non-zero on ‘SIP/1111-000000e9’ in macro 'dial-one’
localhostCLI> == Spawn extension (macro-exten-vm, s, 16) exited non-zero on ‘SIP/1111-000000e9’ in macro 'exten-vm’
localhost
CLI> == Spawn extension (from-internal, 1112, 2) exited non-zero on 'SIP/1111-000000e9’
localhostCLI> localhostCLI> localhostCLI> localhostCLI> Retransmitting #1 (NAT) to 1xx.53.xxx.82:4568:
BYE sip:[email protected]:4568;rinstance=69a9f233d184a028 SIP/2.0
Via: SIP/2.0/UDP 2xx.218.xxx.39:5060;branch=z9hG4bK4172a22c;rport
Max-Forwards: 70
From: “pbx1” sip:[email protected];tag=as12209a93
To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: FPBX-2.11.0(11.5.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<— SIP read from UDP:1xx.53.xxx.82:4568 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2xx.218.xxx.39:5060;branch=z9hG4bK4172a22c;rport=5060
Contact: sip:[email protected]:4568;rinstance=69a9f233d184a028
To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978
From: "pbx1"sip:[email protected];tag=as12209a93
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]18.xxx.39:5060’ Method: INVITE
localhost*CLI>
<— SIP read from UDP:1xx.53.xxx.82:4568 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2xx.218.xxx.39:5060;branch=z9hG4bK4172a22c;rport=5060
Contact: sip:[email protected]:4568;rinstance=69a9f233d184a028
To: sip:[email protected]:4568;rinstance=69a9f233d184a028;tag=64794978
From: "pbx1"sip:[email protected];tag=as12209a93
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 0

<------------->

Thank you,

j