System MOH plays on outbound call when calling a cell that has a ringback

Hello! Odd issue here, randomly when calling cell phones that have a ringback tone instead of a normal ring the PBX plays its own hold music until the call is answered. This doesn’t happen on every call to the number, but randomly happens. What is very odd is that it only happens when calling a number which has a ringback and does not have a normal ring. Does anyone know what could be causing this? I seem to be getting the 180/183 messages back during the initial call setup, so I’m not sure why the system would be playing the hold music other than no media is actually being delivered, however that is very difficult to tell with the hold music playing. At first I thought it may be because of early media, or reinvites, so I tried the early media settings (prematuremedia=no progressinband=yes) and still was able to reproduce the problem regularly. Any setting change I could make to prevent this?

I’ve attached a log of a call in which this happens, I do see during the call it actually looks like the call is put on hold until the party answers:
— (12 headers 11 lines) —
– Started music on hold, class ‘default’, on channel 'SIP/999-000134c0’
sip_route_dump: route/path hop: sip:95.27.128.251:5060;lr=on;ftag=as02bfc97f
– SIP/sips-000134c1 is making progress passing it to SIP/999-000134c0

EDIT:
I had to post the call log in a reply below

EDIT 2:
Talked to someone at verizon, they say that this is probably happening because when you call one of their customers that has a ringback there can sometimes be a longer than normal “pause”, the pause is probably what is causing the system to kick in the hold music is what I am guessing, but I have no idea how to tell the system to wait longer.

This is the full log:

<--- SIP read from UDP:10.10.10.31:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.10.10.31:5060;rport;branch=z9hG4bKPjiPtzjDId1GKRmlC5XbUdk43g0Z9w4a2k
Max-Forwards: 70
From: "frank <999>" <sip:[email protected]>;tag=kJXTA-cBhStssPmHlxqNs5qjeBIr1Zml
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060;ob>
Call-ID: 2i-NO2.j0h4-epwr5jTmtgQtc4PpgQsx
CSeq: 7626 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Digium D70 2_2_1_7
Authorization: Digest username="999", realm="asterisk", nonce="76e78cf9", uri="sip:[email protected]", response="cf184672d62788f31990b15c052a6a5b", algorithm=MD5
Content-Type: application/sdp
Content-Length: 376

v=0
o=- 199479458 199479458 IN IP4 192.168.1.126
s=digphn
b=AS:84
t=0 0
a=X-nat:0
m=audio 4106 RTP/AVP 0 8 9 111 18 96
c=IN IP4 192.168.1.126
b=TIAS:64000
a=rtcp:4107 IN IP4 192.168.1.126
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
<------------->
--- (16 headers 18 lines) ---
Sending to 10.10.10.31:5060 (NAT)
Using INVITE request as basis request - 2i-NO2.j0h4-epwr5jTmtgQtc4PpgQsx
Found peer '999' for '999' from 10.10.10.31:5060
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 111
Found RTP audio format 18
Found RTP audio format 96
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format G726-32 for ID 111
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 96
Capabilities: us - (ulaw|alaw|gsm), peer - audio=(ulaw|alaw|g722|g729|g726)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.126:4106
Looking for 567712XXXX in from-internal (domain 95.27.163.45)
Reliably Transmitting (NAT) to 10.0.10.179:5060:
NOTIFY sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 10.0.10.5:5060;branch=z9hG4bK42498deb;rport
Max-Forwards: 70
From: sip:[email protected];tag=as6984bd40
To: "Lima Xtra CL Cube <1028>" <sip:[email protected]>;tag=KsZu5KStYLgrmOqWq1.HzPjVs4ywYy5d
Contact: <sip:[email protected]:5060>
Call-ID: wFf1gkgFOAF3GRd7YV0qoprr3fphqAoB
CSeq: 15084 NOTIFY
User-Agent: FPBX-13.0.192.8(13.14.0)
Subscription-State: active
Event: presence
Content-Type: application/pidf+xml
Content-Length: 661

<?xml version="1.0" encoding="ISO-8859-1"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" 
xmlns:pp="urn:ietf:params:xml:ns:pidf:person"
xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status"
xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"
entity="sip:[email protected]">
<pp:person><status>
<ep:activities><ep:busy/></ep:activities>
</status></pp:person>
<note>On the phone</note>
<tuple id="auto_hint_999">
<contact priority="1">sip:[email protected]</contact>
<status><basic>open</basic></status>
</tuple>
<tuple id="digium-presence">
<status>
<digium_presence type="available" subtype=""></digium_presence>
</status>
</tuple>
</presence>

---


---
  == Extension Changed auto_hint_999[from-internal] new state InUse for Notify User 1508 
sip_route_dump: route/path hop: <sip:[email protected]:5060;ob>
Reliably Transmitting (NAT) to 10.0.10.158:5060:
NOTIFY sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 10.0.10.5:5060;branch=z9hG4bK35b964b0;rport
Max-Forwards: 70
From: sip:[email protected];tag=as18597ab7
To: "Lima Large Conference Room <1022>" <sip:[email protected]>;tag=RzJQz6Nwe6SjPufcXz7FZVJ2dARJGGSV
Contact: <sip:[email protected]:5060>
Call-ID: X4WGzQ2MU0uHOErEVRfjy2jKMf.iOe0r
CSeq: 7460 NOTIFY
User-Agent: FPBX-13.0.192.8(13.14.0)
Subscription-State: active
Event: presence
Content-Type: application/pidf+xml
Content-Length: 661

<?xml version="1.0" encoding="ISO-8859-1"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" 
xmlns:pp="urn:ietf:params:xml:ns:pidf:person"
xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status"
xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"
entity="sip:[email protected]">
<pp:person><status>
<ep:activities><ep:busy/></ep:activities>
</status></pp:person>
<note>On the phone</note>
<tuple id="auto_hint_999">
<contact priority="1">sip:[email protected]</contact>
<status><basic>open</basic></status>
</tuple>
<tuple id="digium-presence">
<status>
<digium_presence type="available" subtype=""></digium_presence>
</status>
</tuple>
</presence>

---
  == Extension Changed auto_hint_999[from-internal] new state InUse for Notify User 1022 

<--- Transmitting (NAT) to 10.10.10.31:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.31:5060;branch=z9hG4bKPjiPtzjDId1GKRmlC5XbUdk43g0Z9w4a2k;received=10.10.10.31;rport=5060
From: "frank <999>" <sip:[email protected]>;tag=kJXTA-cBhStssPmHlxqNs5qjeBIr1Zml
To: <sip:[email protected]>
Call-ID: 2i-NO2.j0h4-epwr5jTmtgQtc4PpgQsx
CSeq: 7626 INVITE
Server: FPBX-13.0.192.8(13.14.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
    -- Executing [[email protected]:1] Macro("SIP/999-000134c0", "user-callerid,LIMIT") in new stack
    -- Executing [[email protected]:1] Set("SIP/999-000134c0", "TOUCH_MONITOR=1504108168.623924") in new stack
    -- Executing [[email protected]:2] Set("SIP/999-000134c0", "AMPUSER=999") in new stack
    -- Executing [[email protected]:3] GotoIf("SIP/999-000134c0", "0?report") in new stack
    -- Executing [[email protected]:4] ExecIf("SIP/999-000134c0", "1?Set(REALCALLERIDNUM=999)") in new stack




--- (10 headers 0 lines) ---
    -- Executing [[email protected]:5] Set("SIP/999-000134c0", "AMPUSER=999") in new stack
    -- Executing [[email protected]:6] GotoIf("SIP/999-000134c0", "0?limit") in new stack
    -- Executing [[email protected]:7] Set("SIP/999-000134c0", "AMPUSERCIDNAME=frank") in new stack
    -- Executing [[email protected]:8] GotoIf("SIP/999-000134c0", "0?report") in new stack
    -- Executing [[email protected]:9] Set("SIP/999-000134c0", "AMPUSERCID=999") in new stack
    -- Executing [[email protected]:10] Set("SIP/999-000134c0", "__DIAL_OPTIONS=Ttr") in new stack
    -- Executing [[email protected]:11] Set("SIP/999-000134c0", "CALLERID(all)="frank" <999>") in new stack
    -- Executing [[email protected]:12] GotoIf("SIP/999-000134c0", "0?limit") in new stack
    -- Executing [[email protected]:13] ExecIf("SIP/999-000134c0", "1?Set(GROUP(concurrency_limit)=999)") in new stack
    -- Executing [[email protected]:14] ExecIf("SIP/999-000134c0", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [[email protected]:15] GotoIf("SIP/999-000134c0", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,29)
    -- Executing [[email protected]:29] Set("SIP/999-000134c0", "CALLERID(number)=999") in new stack
    -- Executing [[email protected]:30] Set("SIP/999-000134c0", "CALLERID(name)=frank") in new stack
    -- Executing [[email protected]:31] GotoIf("SIP/999-000134c0", "0?cnum") in new stack
    -- Executing [[email protected]:32] Set("SIP/999-000134c0", "CDR(cnam)=frank") in new stack
    -- Executing [[email protected]:33] Set("SIP/999-000134c0", "CDR(cnum)=999") in new stack
    -- Executing [[email protected]:34] Set("SIP/999-000134c0", "CHANNEL(language)=en") in new stack
    -- Executing [[email protected]:35] GosubIf("SIP/999-000134c0", "0?app-check-classofservce,s,1()") in new stack
    -- Executing [[email protected]:2] Set("SIP/999-000134c0", "ROUTEUSER=999") in new stack
    -- Executing [[email protected]:3] GotoIf("SIP/999-000134c0", "1?notblind") in new stack
    -- Goto (from-internal,567712XXXX,6)
    -- Executing [[email protected]:6] GotoIf("SIP/999-000134c0", "1?restrictedroute-98c6f2c2287f4c73cea3d40ae7ec3ff2,567712XXXX,2:outbound-allroutes,567712XXXX,2") in new stack
    -- Goto (restrictedroute-98c6f2c2287f4c73cea3d40ae7ec3ff2,567712XXXX,2)
    -- Executing [[email protected]:2] Gosub("SIP/999-000134c0", "sub-record-check,s,1(out,567712XXXX,dontcare)") in new stack
    -- Executing [[email protected]:1] GotoIf("SIP/999-000134c0", "0?initialized") in new stack
    -- Executing [[email protected]:2] Set("SIP/999-000134c0", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [[email protected]:3] Set("SIP/999-000134c0", "NOW=1504108168") in new stack
    -- Executing [[email protected]:4] Set("SIP/999-000134c0", "__DAY=30") in new stack
    -- Executing [[email protected]:5] Set("SIP/999-000134c0", "__MONTH=08") in new stack
    -- Executing [[email protected]:6] Set("SIP/999-000134c0", "__YEAR=2017") in new stack
    -- Executing [[email protected]:7] Set("SIP/999-000134c0", "__TIMESTR=20170830-114928") in new stack
    -- Executing [[email protected]:8] Set("SIP/999-000134c0", "__FROMEXTEN=999") in new stack
    -- Executing [[email protected]:9] Set("SIP/999-000134c0", "__MON_FMT=wav") in new stack
    -- Executing [[email protected]:10] NoOp("SIP/999-000134c0", "Recordings initialized") in new stack
    -- Executing [[email protected]:11] ExecIf("SIP/999-000134c0", "0?Set(ARG3=dontcare)") in new stack
    -- Executing [[email protected]:12] Set("SIP/999-000134c0", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [[email protected]:13] ExecIf("SIP/999-000134c0", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [[email protected]:14] GotoIf("SIP/999-000134c0", "3?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [[email protected]:17] GotoIf("SIP/999-000134c0", "1?sub-record-check,out,1") in new stack
    -- Goto (sub-record-check,out,1)
    -- Executing [[email protected]:1] NoOp("SIP/999-000134c0", "Outbound Recording Check from 999 to 567712XXXX") in new stack
    -- Executing [[email protected]:2] Set("SIP/999-000134c0", "RECMODE=dontcare") in new stack
    -- Executing [[email protected]:3] ExecIf("SIP/999-000134c0", "1?Goto(routewins)") in new stack
    -- Goto (sub-record-check,out,7)
    -- Executing [[email protected]:7] Gosub("SIP/999-000134c0", "recordcheck,1(dontcare,out,567712XXXX)") in new stack
    -- Executing [[email protected]:1] NoOp("SIP/999-000134c0", "Starting recording check against dontcare") in new stack
    -- Executing [[email protected]:2] Goto("SIP/999-000134c0", "dontcare") in new stack
    -- Goto (sub-record-check,recordcheck,3)
    -- Executing [[email protected]:3] Return("SIP/999-000134c0", "") in new stack
    -- Executing [[email protected]:8] Return("SIP/999-000134c0", "") in new stack
    -- Executing [[email protected]:3] ExecIf("SIP/999-000134c0", "0 ?Set(CDR(accountcode)=)") in new stack
    -- Executing [[email protected]:4] Set("SIP/999-000134c0", "ROUTE_CIDSAVE="frank" <999>") in new stack
    -- Executing [[email protected]:5] Set("SIP/999-000134c0", "MOHCLASS=none") in new stack
    -- Executing [[email protected]:6] Set("SIP/999-000134c0", "_NODEST=") in new stack
    -- Executing [[email protected]:7] Macro("SIP/999-000134c0", "dialout-trunk,2,567712XXXX,,off") in new stack
    -- Executing [[email protected]:1] Set("SIP/999-000134c0", "DIAL_TRUNK=2") in new stack
    -- Executing [[email protected]:2] GosubIf("SIP/999-000134c0", "0?sub-pincheck,s,1()") in new stack
    -- Executing [[email protected]:3] GotoIf("SIP/999-000134c0", "0?disabletrunk,1") in new stack
    -- Executing [[email protected]:4] Set("SIP/999-000134c0", "DIAL_NUMBER=567712XXXX") in new stack
    -- Executing [[email protected]:5] Set("SIP/999-000134c0", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
    -- Executing [[email protected]:6] Set("SIP/999-000134c0", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [[email protected]:7] GotoIf("SIP/999-000134c0", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [[email protected]:9] GotoIf("SIP/999-000134c0", "0?skipoutcid") in new stack
    -- Executing [[email protected]:10] Set("SIP/999-000134c0", "DIAL_TRUNK_OPTIONS=T") in new stack
    -- Executing [[email protected]:11] Macro("SIP/999-000134c0", "outbound-callerid,2") in new stack
    -- Executing [[email protected]:1] ExecIf("SIP/999-000134c0", "0?Set(CALLERPRES(name-pres)=)") in new stack
    -- Executing [[email protected]:2] ExecIf("SIP/999-000134c0", "0?Set(CALLERPRES(num-pres)=)") in new stack
    -- Executing [[email protected]:3] ExecIf("SIP/999-000134c0", "0?Set(REALCALLERIDNUM=999)") in new stack
    -- Executing [[email protected]:4] GotoIf("SIP/999-000134c0", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,7)
    -- Executing [[email protected]:7] Set("SIP/999-000134c0", "USEROUTCID=8883821222") in new stack
    -- Executing [[email protected]:8] Set("SIP/999-000134c0", "EMERGENCYCID=") in new stack
    -- Executing [[email protected]:9] Set("SIP/999-000134c0", "TRUNKOUTCID=") in new stack
    -- Executing [[email protected]:10] GotoIf("SIP/999-000134c0", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,15)
    -- Executing [[email protected]:15] ExecIf("SIP/999-000134c0", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [[email protected]:16] ExecIf("SIP/999-000134c0", "1?Set(CALLERID(all)=8883821222)") in new stack
    -- Executing [[email protected]:17] ExecIf("SIP/999-000134c0", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [[email protected]:18] ExecIf("SIP/999-000134c0", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
    -- Executing [[email protected]:19] ExecIf("SIP/999-000134c0", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
    -- Executing [[email protected]:20] Set("SIP/999-000134c0", "CDR(outbound_cnum)=8883821222") in new stack
    -- Executing [[email protected]:21] Set("SIP/999-000134c0", "CDR(outbound_cnam)=") in new stack
[2017-08-30 11:49:28] WARNING[16940]: func_cdr.c:383 cdr_write_callback: CDR requires a value (CDR(variable)=value)
    -- Executing [[email protected]:12] GosubIf("SIP/999-000134c0", "0?sub-flp-2,s,1()") in new stack
    -- Executing [[email protected]:13] Set("SIP/999-000134c0", "OUTNUM=567712XXXX") in new stack
    -- Executing [[email protected]:14] Set("SIP/999-000134c0", "custom=SIP/sips") in new stack
    -- Executing [[email protected]:15] ExecIf("SIP/999-000134c0", "1?Set(DIAL_TRUNK_OPTIONS=M(setmusic^none)T)") in new stack
    -- Executing [[email protected]:16] ExecIf("SIP/999-000134c0", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^none)TM(confirm))") in new stack
    -- Executing [[email protected]:17] Macro("SIP/999-000134c0", "dialout-trunk-predial-hook,") in new stack
    -- Executing [[email protected]:1] MacroExit("SIP/999-000134c0", "") in new stack
    -- Executing [[email protected]:18] GotoIf("SIP/999-000134c0", "0?bypass,1") in new stack
    -- Executing [[email protected]:19] ExecIf("SIP/999-000134c0", "1?Set(CONNECTEDLINE(num,i)=567712XXXX)") in new stack
    -- Executing [[email protected]:20] ExecIf("SIP/999-000134c0", "1?Set(CONNECTEDLINE(name,i)=CID:8883821222)") in new stack
    -- Executing [[email protected]:21] ExecIf("SIP/999-000134c0", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)8883821222)") in new stack
    -- Executing [[email protected]:22] GotoIf("SIP/999-000134c0", "0?customtrunk") in new stack
    -- Executing [[email protected]:23] Dial("SIP/999-000134c0", "SIP/sips/567712XXXX,300,M(setmusic^none)T") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 16246
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 95.27.128.251:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 95.27.163.45:5060;branch=z9hG4bK3d27122c;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as02bfc97f
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-13.0.192.8(13.14.0)
Date: Wed, 30 Aug 2017 15:49:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "8883821222" <sip:[email protected]>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 299

v=0
o=root 1799909422 1799909422 IN IP4 95.27.163.45
s=Asterisk PBX 13.14.0
c=IN IP4 95.27.163.45
t=0 0
m=audio 16246 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 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/sips/567712XXXX

<--- SIP read from UDP:95.27.128.251:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.27.163.45:5060;branch=z9hG4bK3d27122c;rport=5060
From: <sip:[email protected]>;tag=as02bfc97f
To: <sip:[email protected]>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:95.27.128.251:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.27.163.45:5060;branch=z9hG4bK3d27122c;rport=5060
From: <sip:[email protected]>;tag=as02bfc97f
To: <sip:[email protected]>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:95.27.128.251:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 95.27.163.45:5060;branch=z9hG4bK3d27122c;rport=5060
From: <sip:[email protected]>;tag=as02bfc97f
To: <sip:[email protected]>;tag=7c69f41d63377b8b61a4183abfc8b16a.430d
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="95.27.163.45", nonce="59a6df7800001455bc2b311ada7fd60e2bc9892d552e5b1a"
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 95.27.128.251:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 95.27.163.45:5060;branch=z9hG4bK3d27122c;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as02bfc97f
To: <sip:[email protected]>;tag=7c69f41d63377b8b61a4183abfc8b16a.430d
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-13.0.192.8(13.14.0)
Content-Length: 0


---
Audio is at 16246
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 95.27.128.251:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 95.27.163.45:5060;branch=z9hG4bK4783e1b5;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as02bfc97f
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: FPBX-13.0.192.8(13.14.0)
Proxy-Authorization: Digest username="419221XXXX", realm="95.27.163.45", algorithm=MD5, uri="sip:[email protected]", nonce="59a6df7800001455bc2b311ada7fd60e2bc9892d552e5b1a", response="c699e72243152195ff7b79850e570ee2"
Date: Wed, 30 Aug 2017 15:49:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "8883821222" <sip:[email protected]>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 299

v=0
o=root 1799909422 1799909423 IN IP4 95.27.163.45
s=Asterisk PBX 13.14.0
c=IN IP4 95.27.163.45
t=0 0
m=audio 16246 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 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:95.27.128.251:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 95.27.163.45:5060;branch=z9hG4bK3d27122c;rport=5060
From: <sip:[email protected]>;tag=as02bfc97f
To: <sip:[email protected]>;tag=7c69f41d63377b8b61a4183abfc8b16a.430d
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="95.27.163.45", nonce="59a6df7800001455bc2b311ada7fd60e2bc9892d552e5b1a"
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 95.27.128.251:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 95.27.163.45:5060;branch=z9hG4bK4783e1b5;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as02bfc97f
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-13.0.192.8(13.14.0)
Content-Length: 0


---

<--- SIP read from UDP:95.27.128.251:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.27.163.45:5060;branch=z9hG4bK4783e1b5;rport=5060
From: <sip:[email protected]>;tag=as02bfc97f
To: <sip:[email protected]>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:95.27.128.251:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.27.163.45:5060;branch=z9hG4bK4783e1b5;rport=5060
From: <sip:[email protected]>;tag=as02bfc97f
To: <sip:[email protected]>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

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


<--- SIP read from UDP:95.27.128.251:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 95.27.163.45:5060;received=95.27.163.45;branch=z9hG4bK4783e1b5;rport=5060
From: <sip:[email protected]>;tag=as02bfc97f
To: <sip:[email protected]>;tag=gK00de6875
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Record-Route: <sip:95.27.128.251:5060;lr=on;ftag=as02bfc97f>
Contact: <sip:[email protected]:5060>
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:95.27.128.251:5060;lr=on;ftag=as02bfc97f>
    -- SIP/sips-000134c1 is ringing

<--- SIP read from UDP:95.27.128.251:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 95.27.163.45:5060;received=95.27.163.45;branch=z9hG4bK4783e1b5;rport=5060
From: <sip:[email protected]>;tag=as02bfc97f
To: <sip:[email protected]>;tag=gK00de6875
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Record-Route: <sip:95.27.128.251:5060;lr=on;ftag=as02bfc97f>
Contact: <sip:[email protected]:5060>
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- Transmitting (NAT) to 10.10.10.31:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.10.31:5060;branch=z9hG4bKPjiPtzjDId1GKRmlC5XbUdk43g0Z9w4a2k;received=10.10.10.31;rport=5060
From: "frank <999>" <sip:[email protected]>;tag=kJXTA-cBhStssPmHlxqNs5qjeBIr1Zml
To: <sip:[email protected]>;tag=as57583943
Call-ID: 2i-NO2.j0h4-epwr5jTmtgQtc4PpgQsx
CSeq: 7626 INVITE
Server: FPBX-13.0.192.8(13.14.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
sip_route_dump: route/path hop: <sip:95.27.128.251:5060;lr=on;ftag=as02bfc97f>
    -- SIP/sips-000134c1 is ringing



<--- SIP read from UDP:95.27.128.251:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 95.27.163.45:5060;received=95.27.163.45;branch=z9hG4bK4783e1b5;rport=5060
From: <sip:[email protected]>;tag=as02bfc97f
To: <sip:[email protected]>;tag=gK00de6875
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Record-Route: <sip:95.27.128.251:5060;lr=on;ftag=as02bfc97f>
Contact: <sip:[email protected]:5060>
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS
Content-Length: 233
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 500130 781533 IN IP4 67.231.5.110
s=SIP Media Capabilities
c=IN IP4 67.231.5.76
t=0 0
m=audio 8162 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=inactive
a=ptime:20
<------------->
--- (12 headers 11 lines) ---
sip_route_dump: route/path hop: <sip:95.27.128.251:5060;lr=on;ftag=as02bfc97f>
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 67.231.5.76:8162
    -- Call on SIP/sips-000134c1 placed on hold

<--- SIP read from UDP:95.27.128.251:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 95.27.163.45:5060;received=95.27.163.45;branch=z9hG4bK4783e1b5;rport=5060
From: <sip:[email protected]>;tag=as02bfc97f
To: <sip:[email protected]>;tag=gK00de6875
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Record-Route: <sip:95.27.128.251:5060;lr=on;ftag=as02bfc97f>
Contact: <sip:[email protected]:5060>
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS
Content-Length: 233
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 500130 781533 IN IP4 67.231.5.110
s=SIP Media Capabilities
c=IN IP4 67.231.5.76
t=0 0
m=audio 8162 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=inactive
a=ptime:20
<------------->
--- (12 headers 11 lines) ---
    -- Started music on hold, class 'default', on channel 'SIP/999-000134c0'
sip_route_dump: route/path hop: <sip:95.27.128.251:5060;lr=on;ftag=as02bfc97f>
    -- SIP/sips-000134c1 is making progress passing it to SIP/999-000134c0
Audio is at 11272
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 10.10.10.31:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.10.10.31:5060;branch=z9hG4bKPjiPtzjDId1GKRmlC5XbUdk43g0Z9w4a2k;received=10.10.10.31;rport=5060
From: "frank <999>" <sip:[email protected]>;tag=kJXTA-cBhStssPmHlxqNs5qjeBIr1Zml
To: <sip:[email protected]>;tag=as57583943
Call-ID: 2i-NO2.j0h4-epwr5jTmtgQtc4PpgQsx
CSeq: 7626 INVITE
Server: FPBX-13.0.192.8(13.14.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 271

v=0
o=root 859361367 859361367 IN IP4 95.27.163.45
s=Asterisk PBX 13.14.0
c=IN IP4 95.27.163.45
t=0 0
m=audio 11272 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
    -- SIP/sips-000134c1 is making progress passing it to SIP/999-000134c0
       > 0x7faf88844410 -- Probation passed - setting RTP source address to 10.10.10.31:4106
       > 0x7faf88844410 -- Probation passed - setting RTP source address to 10.10.10.31:4106