No outgoing audio on SIP calls

Ok, first of all forgive me, I really can’t get my head around this NAT stuff.

Basically, I’ve got outgoing/incoming calls connecting but no audio coming inbound when I initiate the call.
However I have two way audio when the call is incoming from the external carrier.

Asterisk 11.8.1
Freepbx 2.11.0.31
Carrier: Gamma

Settings first, then a log at the bottom:

Asterisk box 192.168.0.9/255.255.255.0
Router 192.168.0.1/255.255.255.0 LAN and 1 single IP address now on quoted as 194.xxx.xxx.xxx
Cisco 7960: 192.168.0.76/255.255.255.0
RTP Ports 32200 - 32500 forwarded from router to asterisk box

In “Asterisk SIP settings” :

RTP Start 32200 and RTP End 32500 are set in portrange
NAT=YES
Static IP set = 194.xxx.xx.xxx
Local Networks set = 192.168.0.0 / 255.255.255.0
Reinvite behaviour = no

I assume (perhaps wrongly) that I don’t need to forward port 5060 from my router to the asterisk box as I don’t require any incoming connections. Also that I don’t need to

Telecom provider is GAMMA, have them set up as a peer in TRUNKS:

type=peer
fromuser= 1234567890 for the purpose of this post
fromdomain= 194.xxx.xxx.xxx for purpose of this post
host=88.215.61.201
dtmfmode=rfc2833
canreinvite=no
insecure=very

Extension 6001 for the 7960 is set up as
Device Options:
canreinvite = no
trustrpid = yes
sendrpid = no
NAT=NO (RFC3581)
Transport UDP only

In “Advanced settings”:

Device Settings
SIP canrenivite (directmedia) = NO
SIP trustrpid = yes
SIP sendrpid = no
SIP nat = yes

I have nothing in the “INCOMING” section under TRUNKS/Gamma
I have nothing in the Register section as Gamma only use IP authentication

Now for the log!

There’s a SIP/2.0 401 Unauthorized near the beginning no idea why though as the phone is registered and works (albeit without audio in certain conditions)

<— SIP read from UDP:192.168.0.76:50783 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK0650fbbf
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected]
Call-ID: [email protected]
Max-Forwards: 70
Date: Thu, 15 May 2014 17:58:30 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “6001” sip:[email protected];party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 276
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 20252 0 IN IP4 192.168.0.76
s=SIP Call
t=0 0
m=audio 19822 RTP/AVP 0 8 18 101
c=IN IP4 192.168.0.76
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 0-15
a=sendrecv
<------------->
— (18 headers 13 lines) —
Sending to 192.168.0.76:50783 (NAT)
Sending to 192.168.0.76:50783 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘6001’ for ‘6001’ from 192.168.0.76:50783

<— Reliably Transmitting (no NAT) to 192.168.0.76:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK0650fbbf;received=192.168.0.76
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected];tag=as62e3443d
Call-ID: [email protected]
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.8.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="55a7bd38"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 26752 ms (Method: INVITE)

<— SIP read from UDP:192.168.0.76:50783 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK0650fbbf
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected]
Call-ID: [email protected]
Max-Forwards: 70
Date: Thu, 15 May 2014 17:58:30 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “6001” sip:[email protected];party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 276
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 20252 0 IN IP4 192.168.0.76
s=SIP Call
t=0 0
m=audio 19822 RTP/AVP 0 8 18 101
c=IN IP4 192.168.0.76
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 0-15
a=sendrecv
<------------->
— (18 headers 13 lines) —
Ignoring this INVITE request

<— SIP read from UDP:192.168.0.76:50880 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK0650fbbf
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected];tag=as62e3443d
Call-ID: [email protected]
Date: Thu, 15 May 2014 17:58:30 GMT
CSeq: 101 ACK
Content-Length: 0

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

<— SIP read from UDP:192.168.0.76:50880 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK0650fbbf
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected];tag=as62e3443d
Call-ID: [email protected]
Date: Thu, 15 May 2014 17:58:30 GMT
CSeq: 101 ACK
Content-Length: 0

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

<— SIP read from UDP:192.168.0.76:50783 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK2b4dac51
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected]
Call-ID: [email protected]
Max-Forwards: 70
Date: Thu, 15 May 2014 17:58:30 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Authorization: Digest username=“6001”,realm=“asterisk”,uri="sip:[email protected]",response=“424daa01a734e3b7ffc42a7311425309”,nonce=“55a7bd38”,algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “6001” sip:[email protected];party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 276
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 20252 0 IN IP4 192.168.0.76
s=SIP Call
t=0 0
m=audio 19822 RTP/AVP 0 8 18 101
c=IN IP4 192.168.0.76
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 0-15
a=sendrecv
<------------->
— (19 headers 13 lines) —
Sending to 192.168.0.76:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘6001’ for ‘6001’ from 192.168.0.76:50783
== 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 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/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 192.168.0.76:19822
Looking for 908001114567 in from-internal (domain 192.168.0.9)
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘CHAN_START’,{ts ‘2014-05-15 18:58:32’},‘6001’,‘6001’,’’,’’,’’,‘908001114567’,‘from-internal’,‘SIP/6001-0000000e’,’’,’’,3,’’,‘1400176712.14’,‘1400176712.14’,’’,’’,’’)]
list_route: hop: sip:[email protected]:5060;transport=udp

<— Transmitting (no NAT) to 192.168.0.76:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK2b4dac51;received=192.168.0.76
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.8.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0

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

<— SIP read from UDP:192.168.0.76:50783 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK2b4dac51
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected]
Call-ID: [email protected]
Max-Forwards: 70
Date: Thu, 15 May 2014 17:58:30 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Authorization: Digest username=“6001”,realm=“asterisk”,uri="sip:[email protected]",response=“424daa01a734e3b7ffc42a7311425309”,nonce=“55a7bd38”,algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “6001” sip:[email protected];party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 276
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 20252 0 IN IP4 192.168.0.76
s=SIP Call
t=0 0
m=audio 19822 RTP/AVP 0 8 18 101
c=IN IP4 192.168.0.76
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 0-15
a=sendrecv
<------------->
— (19 headers 13 lines) —
Ignoring this INVITE request

<— Transmitting (no NAT) to 192.168.0.76:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK2b4dac51;received=192.168.0.76
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.8.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
– Executing [[email protected]:1] Macro(“SIP/6001-0000000e”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [[email protected]:1] Set(“SIP/6001-0000000e”, “TOUCH_MONITOR=1400176712.14”) in new stack
– Executing [[email protected]o-user-callerid:2] Set(“SIP/6001-0000000e”, “AMPUSER=6001”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/6001-0000000e”, “0?report”) in new stack
– Executing [[email protected]:4] ExecIf(“SIP/6001-0000000e”, “1?Set(REALCALLERIDNUM=6001)”) in new stack
– Executing [[email protected]:5] Set(“SIP/6001-0000000e”, “AMPUSER=6001”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/6001-0000000e”, “0?limit”) in new stack
– Executing [[email protected]:7] Set(“SIP/6001-0000000e”, “AMPUSERCIDNAME=John Smith”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/6001-0000000e”, “0?report”) in new stack
– Executing [[email protected]:9] Set(“SIP/6001-0000000e”, “AMPUSERCID=6001”) in new stack
– Executing [[email protected]:10] Set(“SIP/6001-0000000e”, “__DIAL_OPTIONS=trwW”) in new stack
– Executing [[email protected]:11] Set(“SIP/6001-0000000e”, “CALLERID(all)=“John Smith” <6001>”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/6001-0000000e”, “0?limit”) in new stack
– Executing [[email protected]:13] ExecIf(“SIP/6001-0000000e”, “1?Set(GROUP(concurrency_limit)=6001)”) in new stack
– Executing [[email protected]:14] GosubIf(“SIP/6001-0000000e”, “7?sub-ccss,s,1(from-internal,908001114567)”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/6001-0000000e”, “0?Return()”) in new stack
– Executing [[email protected]:2] Set(“SIP/6001-0000000e”, “CCSS_SETUP=TRUE”) in new stack
– Executing [[email protected]:3] GosubIf(“SIP/6001-0000000e”, “0?monitor_config,1(from-internal,908001114567):monitor_default,1(from-internal,908001114567)”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/6001-0000000e”, “0?is_exten”) in new stack
– Executing [[email protected]:2] StackPop(“SIP/6001-0000000e”, “”) in new stack
– Executing [[email protected]:3] Return(“SIP/6001-0000000e”, “FALSE”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/6001-0000000e”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [[email protected]:16] GotoIf(“SIP/6001-0000000e”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,29)
– Executing [[email protected]:29] Set(“SIP/6001-0000000e”, “CALLERID(number)=6001”) in new stack
– Executing [[email protected]:30] Set(“SIP/6001-0000000e”, “CALLERID(name)=John Smith”) in new stack
– Executing [[email protected]:31] Set(“SIP/6001-0000000e”, “CDR(cnum)=6001”) in new stack
– Executing [[email protected]:32] Set(“SIP/6001-0000000e”, “CDR(cnam)=John Smith”) in new stack
– Executing [[email protected]:33] Set(“SIP/6001-0000000e”, “CHANNEL(language)=en”) in new stack
– Executing [[email protected]:2] Set(“SIP/6001-0000000e”, “MOHCLASS=default”) in new stack
– Executing [[email protected]:3] Set(“SIP/6001-0000000e”, “_NODEST=”) in new stack
– Executing [[email protected]:4] Gosub(“SIP/6001-0000000e”, “sub-record-check,s,1(out,908001114567,)”) in new stack
– Executing [[email protected]:1] Set(“SIP/6001-0000000e”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/6001-0000000e”, “1?check”) in new stack
– Goto (sub-record-check,s,7)
– Executing [[email protected]:7] Set(“SIP/6001-0000000e”, “__MON_FMT=wav”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/6001-0000000e”, “1?next”) in new stack
– Goto (sub-record-check,s,11)
– Executing [[email protected]:11] ExecIf(“SIP/6001-0000000e”, “0?Return()”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/6001-0000000e”, “0?Set(__REC_POLICY_MODE=)”) in new stack
– Executing [[email protected]:13] GotoIf(“SIP/6001-0000000e”, “0?out,1”) in new stack
– Executing [[email protected]:14] Set(“SIP/6001-0000000e”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [[email protected]:15] Set(“SIP/6001-0000000e”, “NOW=1400176712”) in new stack
– Executing [[email protected]:16] Set(“SIP/6001-0000000e”, “__DAY=15”) in new stack
– Executing [[email protected]:17] Set(“SIP/6001-0000000e”, “__MONTH=05”) in new stack
– Executing [[email protected]:18] Set(“SIP/6001-0000000e”, “__YEAR=2014”) in new stack
– Executing [[email protected]:19] Set(“SIP/6001-0000000e”, “__TIMESTR=20140515-185832”) in new stack
– Executing [[email protected]:20] Set(“SIP/6001-0000000e”, “__FROMEXTEN=6001”) in new stack
– Executing [[email protected]:21] Set(“SIP/6001-0000000e”, “__CALLFILENAME=out-908001114567-6001-20140515-185832-1400176712.14”) in new stack
– Executing [[email protected]:22] Goto(“SIP/6001-0000000e”, “out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [[email protected]:1] ExecIf(“SIP/6001-0000000e”, “1?Set(__REC_POLICY_MODE=dontcare)”) in new stack
– Executing [[email protected]:2] GosubIf(“SIP/6001-0000000e”, “0?record,1(exten,908001114567,6001)”) in new stack
– Executing [[email protected]:3] Return(“SIP/6001-0000000e”, “”) in new stack
– Executing [[email protected]:5] Macro(“SIP/6001-0000000e”, “dialout-trunk,3,08001114567,off”) in new stack
– Executing [[email protected]:1] Set(“SIP/6001-0000000e”, “DIAL_TRUNK=3”) in new stack
– Executing [[email protected]:2] GosubIf(“SIP/6001-0000000e”, “0?sub-pincheck,s,1()”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/6001-0000000e”, “0?disabletrunk,1”) in new stack
– Executing [[email protected]:4] Set(“SIP/6001-0000000e”, “DIAL_NUMBER=08001114567”) in new stack
– Executing [[email protected]:5] Set(“SIP/6001-0000000e”, “DIAL_TRUNK_OPTIONS=trwW”) in new stack
– Executing [[email protected]:6] Set(“SIP/6001-0000000e”, “OUTBOUND_GROUP=OUT_3”) in new stack
– Executing [[email protected]:7] GotoIf(“SIP/6001-0000000e”, “0?nomax”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/6001-0000000e”, “0?chanfull”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/6001-0000000e”, “0?skipoutcid”) in new stack
– Executing [[email protected]:10] Set(“SIP/6001-0000000e”, “DIAL_TRUNK_OPTIONS=Tt”) in new stack
– Executing [[email protected]:11] Macro(“SIP/6001-0000000e”, “outbound-callerid,3”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/6001-0000000e”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/6001-0000000e”, “0?Set(REALCALLERIDNUM=6001)”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/6001-0000000e”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [[email protected]:6] Set(“SIP/6001-0000000e”, “USEROUTCID=1234567890”) in new stack
– Executing [[email protected]:7] Set(“SIP/6001-0000000e”, “EMERGENCYCID=”) in new stack
– Executing [[email protected]:8] Set(“SIP/6001-0000000e”, “TRUNKOUTCID=1234567890”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/6001-0000000e”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,14)
– Executing [[email protected]:14] ExecIf(“SIP/6001-0000000e”, “1?Set(CALLERID(all)=1234567890)”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/6001-0000000e”, “1?Set(CALLERID(all)=1234567890)”) in new stack
– Executing [[email protected]:16] ExecIf(“SIP/6001-0000000e”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [[email protected]:17] ExecIf(“SIP/6001-0000000e”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
– Executing [[email protected]:18] Set(“SIP/6001-0000000e”, “CDR(outbound_cnum)=1234567890”) in new stack
– Executing [[email protected]:19] Set(“SIP/6001-0000000e”, “CDR(outbound_cnam)=”) in new stack
– Executing [[email protected]:12] GosubIf(“SIP/6001-0000000e”, “0?sub-flp-3,s,1()”) in new stack
– Executing [[email protected]:13] Set(“SIP/6001-0000000e”, “OUTNUM=08001114567”) in new stack
– Executing [[email protected]:14] Set(“SIP/6001-0000000e”, “custom=SIP/GAMMA_Peer”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/6001-0000000e”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)”) in new stack
– Executing [[email protected]:16] ExecIf(“SIP/6001-0000000e”, “0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))”) in new stack
– Executing [[email protected]:17] Macro(“SIP/6001-0000000e”, “dialout-trunk-predial-hook,”) in new stack
– Executing [[email protected]:1] MacroExit(“SIP/6001-0000000e”, “”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/6001-0000000e”, “0?bypass,1”) in new stack
– Executing [[email protected]:19] ExecIf(“SIP/6001-0000000e”, “1?Set(CONNECTEDLINE(num,i)=08001114567)”) in new stack
– Executing [[email protected]:20] ExecIf(“SIP/6001-0000000e”, “1?Set(CONNECTEDLINE(name,i)=CID:1234567890)”) in new stack
– Executing [[email protected]:21] GotoIf(“SIP/6001-0000000e”, “0?customtrunk”) in new stack
– Executing [[email protected]:22] Dial(“SIP/6001-0000000e”, “SIP/GAMMA_Peer/08001114567,300,Tt”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘CHAN_START’,{ts ‘2014-05-15 18:58:32’},’’,’’,’’,’’,’’,‘s’,‘from-trunk-sip-GAMMA_Peer’,‘SIP/GAMMA_Peer-0000000f’,’’,’’,3,’’,‘1400176712.15’,‘1400176712.14’,’’,’’,’’)]
Audio is at 32490
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 88.215.61.201:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 194.xxx.xxx.xxx:5060;branch=z9hG4bK087045d5;rport
Max-Forwards: 70
From: sip:[email protected];tag=as1c28a0ce
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.8.1)
Date: Thu, 15 May 2014 17:58:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 1295982853 1295982853 IN IP4 194.xxx.xxx.xxx
s=Asterisk PBX 11.8.1
c=IN IP4 194.xxx.xxx.xxx
t=0 0
m=audio 32490 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


-- Called SIP/GAMMA_Peer/08001114567

<— SIP read from UDP:88.215.61.201:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.9:5060;rport=5060;received=192.168.0.9;branch=z9hG4bK087045d5
From: sip:[email protected]:5060;tag=as1c28a0ce
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:88.215.61.201:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.9:5060;rport=5060;received=192.168.0.9;branch=z9hG4bK087045d5
From: sip:[email protected]:5060;tag=as1c28a0ce
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:88.215.61.201:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.9:5060;rport=5060;received=192.168.0.9;branch=z9hG4bK087045d5
From: sip:[email protected]:5060;tag=as1c28a0ce
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:88.215.61.201:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.9:5060;rport=5060;received=192.168.0.9;branch=z9hG4bK087045d5
From: sip:[email protected]:5060;tag=as1c28a0ce
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:88.215.61.201:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.9:5060;rport=5060;received=192.168.0.9;branch=z9hG4bK087045d5
To: sip:[email protected];tag=3609165512-376653
From: sip:[email protected]:5060;tag=as1c28a0ce
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 214

v=0
o=MSX23 8050298145208028827 1 IN IP4 88.215.61.202
s=sip call
c=IN IP4 88.215.61.202
t=0 0
m=audio 6392 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (11 headers 10 lines) —
list_route: hop: sip:[email protected]:5060
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 88.215.61.202:6392

<— SIP read from UDP:88.215.61.201:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.9:5060;rport=5060;received=192.168.0.9;branch=z9hG4bK087045d5
To: sip:[email protected];tag=3609165512-376653
From: sip:[email protected]:5060;tag=as1c28a0ce
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 214

v=0
o=MSX23 8050298145208028827 1 IN IP4 88.215.61.202
s=sip call
c=IN IP4 88.215.61.202
t=0 0
m=audio 6392 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (11 headers 10 lines) —
list_route: hop: sip:[email protected]:5060
– SIP/GAMMA_Peer-0000000f is making progress passing it to SIP/6001-0000000e
Audio is at 32332
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (no NAT) to 192.168.0.76:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK2b4dac51;received=192.168.0.76
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected];tag=as0b712b1a
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.8.1)
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: 255

v=0
o=root 469897684 469897684 IN IP4 192.168.0.9
s=Asterisk PBX 11.8.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 32332 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
– SIP/GAMMA_Peer-0000000f is making progress passing it to SIP/6001-0000000e
> 0xb744d200 – Probation passed - setting RTP source address to 192.168.0.76:19822

<— SIP read from UDP:88.215.61.201:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.9:5060;rport=5060;received=192.168.0.9;branch=z9hG4bK087045d5
To: sip:[email protected];tag=3609165512-376653
From: sip:[email protected]:5060;tag=as1c28a0ce
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: sip:[email protected]:5060
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: hop: sip:[email protected]:5060

<— SIP read from UDP:88.215.61.201:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.9:5060;rport=5060;received=192.168.0.9;branch=z9hG4bK087045d5
To: sip:[email protected];tag=3609165512-376653
From: sip:[email protected]:5060;tag=as1c28a0ce
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: sip:[email protected]:5060
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: hop: sip:[email protected]:5060
– SIP/GAMMA_Peer-0000000f is ringing

<— Transmitting (no NAT) to 192.168.0.76:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK2b4dac51;received=192.168.0.76
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected];tag=as0b712b1a
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.8.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
– SIP/GAMMA_Peer-0000000f is ringing

<— SIP read from UDP:88.215.61.201:5060 —>
SIP/2.0 200 OK
Session-Expires: 3600;refresher=uas
Require: timer
Via: SIP/2.0/UDP 192.168.0.9:5060;rport=5060;received=192.168.0.9;branch=z9hG4bK087045d5
To: sip:[email protected];tag=3609165512-376653
From: sip:[email protected]:5060;tag=as1c28a0ce
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 214

v=0
o=MSX23 8050298145208028827 1 IN IP4 88.215.61.202
s=sip call
c=IN IP4 88.215.61.202
t=0 0
m=audio 6392 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (13 headers 10 lines) —
list_route: hop: sip:[email protected]:5060
set_destination: Parsing sip:[email protected]:5060 for address/port to send to
set_destination: set destination to 88.215.61.201:5060
Transmitting (NAT) to 88.215.61.201:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 194.xxx.xxx.xxx:5060;branch=z9hG4bK689b9dab;rport
Max-Forwards: 70
From: sip:[email protected];tag=as1c28a0ce
To: sip:[email protected];tag=3609165512-376653
Contact: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.8.1)
Content-Length: 0


<— SIP read from UDP:88.215.61.201:5060 —>
SIP/2.0 200 OK
Session-Expires: 3600;refresher=uas
Require: timer
Via: SIP/2.0/UDP 192.168.0.9:5060;rport=5060;received=192.168.0.9;branch=z9hG4bK087045d5
To: sip:[email protected];tag=3609165512-376653
From: sip:[email protected]:5060;tag=as1c28a0ce
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 214

v=0
o=MSX23 8050298145208028827 1 IN IP4 88.215.61.202
s=sip call
c=IN IP4 88.215.61.202
t=0 0
m=audio 6392 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (13 headers 10 lines) —
set_destination: Parsing sip:[email protected]:5060 for address/port to send to
set_destination: set destination to 88.215.61.201:5060
Transmitting (NAT) to 88.215.61.201:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 194.xxx.xxx.xxx:5060;branch=z9hG4bK7e0e9500;rport
Max-Forwards: 70
From: sip:[email protected];tag=as1c28a0ce
To: sip:[email protected];tag=3609165512-376653
Contact: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.8.1)
Content-Length: 0


   > [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('ANSWER',{ts '2014-05-15 18:58:35'},'CID:1234567890','908001114567','','','','908001114567','from-trunk-sip-GAMMA_Peer','SIP/GAMMA_Peer-0000000f','AppDial','(Outgoing Line)',3,'','1400176712.15','1400176712.14','','','')]
-- SIP/GAMMA_Peer-0000000f answered SIP/6001-0000000e

Audio is at 32332
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.0.76:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK2b4dac51;received=192.168.0.76
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected];tag=as0b712b1a
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.8.1)
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: 255

v=0
o=root 469897684 469897684 IN IP4 192.168.0.9
s=Asterisk PBX 11.8.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 32332 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘ANSWER’,{ts ‘2014-05-15 18:58:35’},’’,‘1234567890’,‘6001’,’’,‘908001114567’,‘s’,‘macro-dialout-trunk’,‘SIP/6001-0000000e’,‘Dial’,‘SIP/GAMMA_Peer/08001114567,300,Tt’,3,’’,‘1400176712.14’,‘1400176712.14’,’’,’’,’’)]
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘BRIDGE_START’,{ts ‘2014-05-15 18:58:35’},’’,‘1234567890’,‘6001’,’’,‘908001114567’,‘s’,‘macro-dialout-trunk’,‘SIP/6001-0000000e’,‘Dial’,‘SIP/GAMMA_Peer/08001114567,300,Tt’,3,’’,‘1400176712.14’,‘1400176712.14’,‘SIP/GAMMA_Peer-0000000f’,’’,’’)]

<— SIP read from UDP:192.168.0.76:50783 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK594a7643
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected];tag=as0b712b1a
Call-ID: [email protected]
Max-Forwards: 70
Date: Thu, 15 May 2014 17:58:34 GMT
CSeq: 102 ACK
User-Agent: Cisco-CP7960G/8.0
Authorization: Digest username=“6001”,realm=“asterisk”,uri="sip:[email protected]",response=“424daa01a734e3b7ffc42a7311425309”,nonce=“55a7bd38”,algorithm=MD5
Remote-Party-ID: “6001” sip:[email protected];party=calling;id-type=subscriber;privacy=off;screen=yes
Content-Length: 0

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

<— SIP read from UDP:192.168.0.76:50783 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK594a7643
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected];tag=as0b712b1a
Call-ID: [email protected]
Max-Forwards: 70
Date: Thu, 15 May 2014 17:58:34 GMT
CSeq: 102 ACK
User-Agent: Cisco-CP7960G/8.0
Authorization: Digest username=“6001”,realm=“asterisk”,uri="sip:[email protected]",response=“424daa01a734e3b7ffc42a7311425309”,nonce=“55a7bd38”,algorithm=MD5
Remote-Party-ID: “6001” sip:[email protected];party=calling;id-type=subscriber;privacy=off;screen=yes
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Reliably Transmitting (no NAT) to 192.168.0.77:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK764b1a75
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as51743214
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.8.1)
Date: Thu, 15 May 2014 17:58:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Retransmitting #1 (no NAT) to 192.168.0.77:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK764b1a75
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as51743214
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.8.1)
Date: Thu, 15 May 2014 17:58:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Retransmitting #2 (no NAT) to 192.168.0.77:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK764b1a75
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as51743214
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.8.1)
Date: Thu, 15 May 2014 17:58:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Retransmitting #3 (no NAT) to 192.168.0.77:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK764b1a75
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as51743214
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.8.1)
Date: Thu, 15 May 2014 17:58:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Retransmitting #4 (no NAT) to 192.168.0.77:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK764b1a75
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as51743214
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.8.1)
Date: Thu, 15 May 2014 17:58:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


[2014-05-15 18:58:44] NOTICE[1798]: chan_sip.c:29497 sip_poke_noanswer: Peer ‘6002’ is now UNREACHABLE! Last qualify: 141
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.0.76:50783 —>
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK7e8b8261
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected];tag=as0b712b1a
Call-ID: [email protected]
Max-Forwards: 70
Date: Thu, 15 May 2014 17:58:44 GMT
CSeq: 103 BYE
User-Agent: Cisco-CP7960G/8.0
Content-Length: 0
Authorization: Digest username=“6001”,realm=“asterisk”,uri=“sip:[email protected]:5060”,response=“f682a4aefd01ce187c6f65edc96d2a93”,nonce=“55a7bd38”,algorithm=MD5

<------------->
— (11 headers 0 lines) —
Sending to 192.168.0.76:5060 (no NAT)
Scheduling destruction of SIP dialog ‘[email protected]’ in 26752 ms (Method: BYE)

<— Transmitting (no NAT) to 192.168.0.76:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK7e8b8261;received=192.168.0.76
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected];tag=as0b712b1a
Call-ID: [email protected]
CSeq: 103 BYE
Server: FPBX-2.11.0(11.8.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

<— SIP read from UDP:192.168.0.76:50783 —>
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK7e8b8261
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected];tag=as0b712b1a
Call-ID: [email protected]
Max-Forwards: 70
Date: Thu, 15 May 2014 17:58:44 GMT
CSeq: 103 BYE
User-Agent: Cisco-CP7960G/8.0
Content-Length: 0
Authorization: Digest username=“6001”,realm=“asterisk”,uri=“sip:[email protected]:5060”,response=“f682a4aefd01ce187c6f65edc96d2a93”,nonce=“55a7bd38”,algorithm=MD5

<------------->
— (11 headers 0 lines) —
Sending to 192.168.0.76:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.0.76:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK7e8b8261;received=192.168.0.76
From: “6001” sip:[email protected];tag=001a2f44d64c00391d425ee6-43f80a1b
To: sip:[email protected];tag=as0b712b1a
Call-ID: [email protected]
CSeq: 103 BYE
Server: FPBX-2.11.0(11.8.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
– Executing [[email protected]:1] Macro(“SIP/6001-0000000e”, “hangupcall,”) in new stack
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘BRIDGE_END’,{ts ‘2014-05-15 18:58:45’},’’,‘1234567890’,‘6001’,’’,‘908001114567’,‘s’,‘macro-dialout-trunk’,‘SIP/6001-0000000e’,‘Dial’,‘SIP/GAMMA_Peer/08001114567,300,Tt’,3,’’,‘1400176712.14’,‘1400176712.14’,‘SIP/GAMMA_Peer-0000000f’,’’,’’)]
– Executing [[email protected]:1] GotoIf(“SIP/6001-0000000e”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [[email protected]:3] ExecIf(“SIP/6001-0000000e”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [[email protected]:4] Hangup(“SIP/6001-0000000e”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/6001-0000000e’ in macro ‘hangupcall’
== Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on ‘SIP/6001-0000000e’
> [INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid,cnum,cnam,outbound_cnum) VALUES ({ ts ‘2014-05-15 18:58:32’ },‘6001’,‘6001’,‘908001114567’,‘from-internal’,‘SIP/6001-0000000e’,‘SIP/GAMMA_Peer-0000000f’,‘Dial’,‘SIP/GAMMA_Peer/08001114567,300,Tt’,13,10,‘ANSWERED’,3,‘1400176712.14’,‘6001’,‘John Smith’,‘1234567890’)]
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘HANGUP’,{ts ‘2014-05-15 18:58:45’},‘CID:1234567890’,‘908001114567’,’’,’’,’’,’’,‘macro-dialout-trunk’,‘SIP/GAMMA_Peer-0000000f’,‘AppDial’,’(Outgoing Line)’,3,’’,‘1400176712.15’,‘1400176712.14’,’’,’’,’’)]
set_destination: Parsing sip:[email protected]:5060 for address/port to send to
set_destination: set destination to 88.215.61.201:5060
Reliably Transmitting (NAT) to 88.215.61.201:5060:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 194.xxx.xxx.xxx:5060;branch=z9hG4bK74e064e8;rport
Max-Forwards: 70
From: sip:[email protected];tag=as1c28a0ce
To: sip:[email protected];tag=3609165512-376653
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: FPBX-2.11.0(11.8.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on ‘SIP/6001-0000000e’ in macro ‘dialout-trunk’
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘CHAN_END’,{ts ‘2014-05-15 18:58:45’},‘CID:1234567890’,‘908001114567’,’’,’’,’’,’’,‘macro-dialout-trunk’,‘SIP/GAMMA_Peer-0000000f’,‘AppDial’,’(Outgoing Line)’,3,’’,‘1400176712.15’,‘1400176712.14’,’’,’’,’’)]
== Spawn extension (from-internal, 908001114567, 5) exited non-zero on ‘SIP/6001-0000000e’
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘HANGUP’,{ts ‘2014-05-15 18:58:45’},’’,‘1234567890’,‘6001’,’’,‘908001114567’,‘908001114567’,‘from-internal’,‘SIP/6001-0000000e’,’’,’’,3,’’,‘1400176712.14’,‘1400176712.14’,’’,’’,’’)]
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘CHAN_END’,{ts ‘2014-05-15 18:58:45’},’’,‘1234567890’,‘6001’,’’,‘908001114567’,‘908001114567’,‘from-internal’,‘SIP/6001-0000000e’,’’,’’,3,’’,‘1400176712.14’,‘1400176712.14’,’’,’’,’’)]
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘LINKEDID_END’,{ts ‘2014-05-15 18:58:45’},’’,‘1234567890’,‘6001’,’’,‘908001114567’,‘908001114567’,‘from-internal’,‘SIP/6001-0000000e’,’’,’’,3,’’,‘1400176712.14’,‘1400176712.14’,’’,’’,’’)]

<— SIP read from UDP:88.215.61.201:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.9:5060;rport=5060;received=192.168.0.9;branch=z9hG4bK74e064e8
To: sip:[email protected];tag=3609165512-376653
From: sip:[email protected]:5060;tag=as1c28a0ce
Call-ID: [email protected]
CSeq: 103 BYE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: sip:[email protected]:5060
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: INVITE

<— SIP read from UDP:88.215.61.201:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.9:5060;rport=5060;received=192.168.0.9;branch=z9hG4bK74e064e8
To: sip:[email protected];tag=3609165512-376653
From: sip:[email protected]:5060;tag=as1c28a0ce
Call-ID: [email protected]
CSeq: 103 BYE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: sip:[email protected]:5060
Content-Length: 0

<------------->
— (9 headers 0 lines) —
localhostCLI> sip set debug off
SIP Debugging Disabled
localhost
CLI>

Thanks in advance for any hints/tips

John

Just noticed that an incoming call worked fine from a standard PSTN line (audio both directions) in another county but I couldn’t hear any audio from a mobile phone calling the same DDI

From your post

– SIP/GAMMA_Peer-0000000f is making progress passing it to SIP/6001-0000000e

0xb744d200 – Probation passed - setting RTP source address to 192.168.0.76:19822

I guess 19822 won’t get through your firewall. Check with the provider but pragmatically set the rtp port range to 10000-20000 to begin with.

Ok, have set RTP back to 10000 - 20000 in asterisk and on the router. Still no audio though.

Spent last night looking through SIP debug logs but nothing was sticking out.

Thanks
John

Shorten the secret on the Cisco phone

Hi SkykingOH, thanks for the reply.

The secret is only 6 characters currently set as 1234ab just to test as I had problems with the default freepbx generated secret.

Once I get everything running properly I am going to set a more secure secret.

Cheers

John

OK, update… Tried different endpoint hardware, softphones etc, still have the no incoming audio on all outgoing calls and no incoming audio on calls originating from a few mobile phones. Other calls originating from PSTN have really good audio in both directions.

Am I right in thinking that this can’t be a NAT issue as some calls work?

Thanks
John

Just found this in the logs,

WARNING[1798][C-0000001c]: chan_sip.c:10111 process_sdp: Declining non-primary audio stream: audio 40046 RTP/AVP 0 3 8 101

Is that perhaps the problem and if so any suggestions ?

Thanks
John

Just found this in the logs,

WARNING[1798][C-0000001c]: chan_sip.c:10111 process_sdp: Declining non-primary audio stream: audio 40046 RTP/AVP 0 3 8 101

Is that perhaps the problem and if so any suggestions ?

Thanks
John

since you are using ip authentication, try routing all tcp/udp traffic from your provider to the pbx. the provider may use a range of ip addresses for initiating calls. I did not see any mention of the type of router you are using so I can’t offer any suggestions on the specific firewall rules.

Hi Bksales,

Yes, Gamma authenticate on IP. I get no audio on calls initiated from my side, would making the changes you suggest to my firewall config make any difference considering it’s my side that is initiating the call?

The reason I ask is that I don’t think my router/firewall (Netgear WNR1000 v2) has the ability to do what you suggest. The only way I could do that would be to put the FreePBX box in a DMZ which I would rather not do for security reasons.

from the carrier to your pbx. check page 5-6 in the user manual for your device. you will have to create a couple of services, sip (udp 5060) and rtp (udp 10,000-20,000), then forward those services to the internal ip address of your phone system. I am surprised that you are able to make or receive any calls if you have not done this and you are using IP authentication. make sure upnp is turned off as well. this is a really basic router. if this setup works, you will probably want to then also add some rules to iptables to block 5060 from everyone but your itsp. it does not look like your router will allow you do this, so you will have to do it in ip tables.