No Audio

FreePBX Distro (FreePBX 2.11.0.11, Asterisk 11.5.1)
open Ports 5060; 10,000 - 20,000

We’ve been trying to call thru extensions (xt100 and xt101), unfortunately we’re not getting any audio in either way.

rtp debug doesn’t show any activity after answering the call.

Please help.

Thanks,

J

P’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.

=~=~=~=~=~=~=~=~=~=~=~= =~=~=~=~=~=~=~=~=~=~=~==~=~=~=~=~=~=~=~=~=~=~==~=

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:
To: "1112"
From: “pbx1” ;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” ;tag=43274e18
To: “1112” ;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” ;tag=as4c55e2ef
From: “pbx1” ;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:
To: "1112"
From: “pbx1” ;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:
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” ;tag=43274e18
To: "1112"
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:
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” ;tag=as12209a93 To: Contact: 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” ;tag=43274e18 To: “1112” ;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: 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” ;tag=as12209a93 To: Contact: 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:
To: ;tag=64794978
From: “pbx1”;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/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” ;tag=43274e18 To: “1112” ;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: 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:
To: ;tag=64794978
From: “pbx1”;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/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:
To: ;tag=64794978
From: “pbx1”;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:
set_destination: Parsing 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” ;tag=as12209a93 To: ;tag=64794978 Contact: 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” ;tag=43274e18 To: “1112” ;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: 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” ;tag=43274e18 To: “1112” ;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: 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:
To: “1112” ;tag=as7788edfc
From: “pbx1” ;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:
To: “1112” ;tag=as7788edfc
From: “pbx1” ;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:
To: “1112” ;tag=as7788edfc
From: “pbx1” ;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” ;tag=43274e18
To: “1112” ;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:
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” ;tag=43274e18
To: “1112” ;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:
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” ;tag=43274e18 To: “1112” ;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: 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:
To: “1112” ;tag=as7788edfc
From: “pbx1” ;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:
To: “1112” ;tag=as7788edfc
From: “pbx1” ;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:
To: “1112” ;tag=as7788edfc
From: “pbx1” ;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” ;tag=43274e18
To: “1112” ;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:
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” ;tag=43274e18 To: “1112” ;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: 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” ;tag=43274e18 To: “1112” ;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: 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:
To: “1112” ;tag=as7788edfc
From: “pbx1” ;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:
To: “1112” ;tag=as7788edfc
From: “pbx1” ;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:
To: “1112” ;tag=as7788edfc
From: “pbx1” ;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” ;tag=43274e18 To: “1112” ;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 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” ;tag=as12209a93
To: ;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” ;tag=as12209a93
To: ;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:
To: ;tag=64794978
From: “pbx1”;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: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:
To: ;tag=64794978
From: “pbx1”;tag=as12209a93
Call-ID:
[email protected]

:5060
CSeq: 103 BYE
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 0

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

Thank you,

What does

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

mean to you? , to us it is hard to figure out. . .

I see you have both a 192 network and a 10 network? Is there NAT between the two? Hopefully not and if so both need to be enumerated in the SIP Settings module under localnet.

Thanks dicko & SkikingOH for the replies. Really appreciate it.

We do setup our extensions outside the server’s network.

Extensions = 192.168.1.x IP’s (1xx.53.xxx.82 - ISP IP)
FreePBX Server = 2xx.218.xxx.39 (10.10.1.39)

SkikingOH,

Sorry really new with FreePBX. Can you please explain it further.How can I enumerate both network in the SIP settings module under local network?

Also,I forgot to mention, it seems that putting the call on hold makes the audio works.

We do have tried setting up IAX2 extensions as well and it seems that the audio issue wasn’t present as we are able to make successful calls.

Thanks again,

j

IAX2 doesn’t use a discrete media stream.

I still don’t understand your topology.

How are the 192.168.1.0/24 and the 10.10.1.0/24 network connected and is there NAT between them (I already asked this in first response).

Oh I see, Sorry for that…

The 10.10.1.0/24 is for the server which is located on a Data center with a Juniper Firewall.
We do setup extensions remotely outside that network (outside the country). Needed ports (5060,10,000-20,000) were opened on the firewall.

Thanks,

j