Eternal SIP RTP

Asterisk Asterisk Support
1

I had an old Asterisk (Asterisk 1.6.2.6 built by bachbuilder). I have since replaced it with this version (Asterisk 13.2.0 built by mockbuild). Reason was the hardware blow up, or I would have kept running it. The firewall was configured to have ports 5060 and 10000-12000 open and pointing to the PBX. The new PBX has the same IP as the old and I can see that I have connection with the phones, but there is no voice. I have done debugs and see that all the connections are trying to get there but no connection for voice. Any new thoughts on this and if needed I can post debugs.

2

I would suggest providing the debugs you mention, but it’s most likely configuration related.

3

What would be the easiest way to do that, one of them is over 2M of info. Sorry haven’t really posted things to forums much lately and don’t want to dump that size file in text onto a reply unless I have to.

4

An initial good thing to provide would be the configuration in use, the SIP signaling (sip set debug on) for the system with a call attempt, and details about layout.

5

Sorry for the huge data dump, here is that debug:

`2016-08-16 14:27:06] VERBOSE[3170] chan_sip.c: Reliably Transmitting (NAT) to 96.39.54.46:1079:
OPTIONS sip:169961@96.39.54.46:1079;line=44b620492578ce9 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.5:5060;branch=z9hG4bK12b8bcbc;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.1.10.5;tag=as5d4f9d03
To: sip:169961@96.39.54.46:1079;line=44b620492578ce9
Contact: sip:Unknown@10.1.10.5:5060
Call-ID: 7e837efb67fcda0a5186739c11bbfc82@10.1.10.5:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.2.0)
Date: Tue, 16 Aug 2016 18:27:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

[2016-08-16 14:27:06] VERBOSE[3170] chan_sip.c:
<— SIP read from UDP:96.39.54.46:1079 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.5:5060;branch=z9hG4bK12b8bcbc;rport=5060;received=173.166.10.30
From: “Unknown” sip:Unknown@10.1.10.5;tag=as5d4f9d03
To: sip:169961@96.39.54.46:1079;line=44b620492578ce9;tag=1666277469
Call-ID: 7e837efb67fcda0a5186739c11bbfc82@10.1.10.5:5060
CSeq: 102 OPTIONS
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Content-Length: 0

<------------->
[2016-08-16 14:27:06] VERBOSE[3170] chan_sip.c: — (10 headers 0 lines) —
[2016-08-16 14:27:06] VERBOSE[3170] chan_sip.c: Really destroying SIP dialog ‘7e837efb67fcda0a5186739c11bbfc82@10.1.10.5:5060’ Method: OPTIONS
[2016-08-16 14:27:17] VERBOSE[3170] chan_sip.c:
<— SIP read from UDP:96.39.54.46:1079 —>
jaK
<------------->
[2016-08-16 14:27:24] VERBOSE[3170] chan_sip.c:
<— SIP read from UDP:96.39.54.46:1079 —>
INVITE sip:*98@173.166.10.30 SIP/2.0
Via: SIP/2.0/UDP 96.39.54.46:1079;rport;branch=z9hG4bK385513576
From: sip:169961@173.166.10.30;tag=140578744
To: sip:*98@173.166.10.30
Call-ID: 1067649114
CSeq: 20 INVITE
Contact: sip:6969@96.39.54.46
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length: 232

v=0
o=169961 1818 4076 IN IP4 96.39.54.46
s=Talk
c=IN IP4 96.39.54.46
t=0 0
m=audio 10820 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
[2016-08-16 14:27:24] VERBOSE[3170] chan_sip.c: — (13 headers 11 lines) —
[2016-08-16 14:27:24] VERBOSE[3170] chan_sip.c: Sending to 96.39.54.46:1079 (no NAT)
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Sending to 96.39.54.46:1079 (no NAT)
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Using INVITE request as basis request - 1067649114
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Found peer ‘169961’ for ‘169961’ from 96.39.54.46:1079
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c:
<— Reliably Transmitting (NAT) to 96.39.54.46:1079 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 96.39.54.46:1079;branch=z9hG4bK385513576;received=96.39.54.46;rport=1079
From: sip:169961@173.166.10.30;tag=140578744
To: sip:*98@173.166.10.30;tag=as458436dd
Call-ID: 1067649114
CSeq: 20 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="475ea3a1"
Content-Length: 0

<------------>
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Scheduling destruction of SIP dialog ‘1067649114’ in 6400 ms (Method: INVITE)
[2016-08-16 14:27:24] VERBOSE[3170] chan_sip.c:
<— SIP read from UDP:96.39.54.46:1079 —>
ACK sip:*98@173.166.10.30 SIP/2.0
Via: SIP/2.0/UDP 96.39.54.46:1079;rport;branch=z9hG4bK385513576
From: sip:169961@173.166.10.30;tag=140578744
To: sip:*98@173.166.10.30;tag=as458436dd
Call-ID: 1067649114
CSeq: 20 ACK
Content-Length: 0

<------------->
[2016-08-16 14:27:24] VERBOSE[3170] chan_sip.c: — (7 headers 0 lines) —
[2016-08-16 14:27:24] VERBOSE[3170] chan_sip.c:
<— SIP read from UDP:96.39.54.46:1079 —>
INVITE sip:*98@173.166.10.30 SIP/2.0
Via: SIP/2.0/UDP 96.39.54.46:1079;rport;branch=z9hG4bK670948151
From: sip:169961@173.166.10.30;tag=140578744
To: sip:*98@173.166.10.30
Call-ID: 1067649114
CSeq: 21 INVITE
Contact: sip:6969@96.39.54.46
Authorization: Digest username=“169961”, realm=“asterisk”, nonce=“475ea3a1”, uri=“sip:*98@173.166.10.30”, response=“d0a0282e1c0b7fe23638e4bfbe6e9a81”, algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length: 232

v=0
o=169961 1818 4076 IN IP4 96.39.54.46
s=Talk
c=IN IP4 96.39.54.46
t=0 0
m=audio 10820 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
[2016-08-16 14:27:24] VERBOSE[3170] chan_sip.c: — (14 headers 11 lines) —
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Sending to 96.39.54.46:1079 (NAT)
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Using INVITE request as basis request - 1067649114
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Found peer ‘169961’ for ‘169961’ from 96.39.54.46:1079
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] netsock2.c: Using SIP RTP TOS bits 184
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] netsock2.c: Using SIP RTP CoS mark 5
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Found RTP audio format 9
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Found RTP audio format 0
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Found RTP audio format 8
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Found RTP audio format 101
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Found audio description format G722 for ID 9
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Found audio description format PCMU for ID 0
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Found audio description format PCMA for ID 8
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Found audio description format telephone-event for ID 101
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Capabilities: us - (ulaw|ilbc|alaw|gsm), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Peer audio RTP is at port 96.39.54.46:10820
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c: Looking for *98 in from-internal (domain 173.166.10.30)
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] sip/route.c: sip_route_dump: route/path hop: sip:6969@96.39.54.46
[2016-08-16 14:27:24] VERBOSE[3170][C-000002a3] chan_sip.c:
<— Transmitting (NAT) to 96.39.54.46:1079 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 96.39.54.46:1079;branch=z9hG4bK670948151;received=96.39.54.46;rport=1079
From: sip:169961@173.166.10.30;tag=140578744
To: sip:*98@173.166.10.30
Call-ID: 1067649114
CSeq: 21 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:*98@10.1.10.5:5060
Content-Length: 0

<------------>
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [*98@from-internal:1] Macro(“SIP/169961-00000057”, “user-callerid,”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:1] Set(“SIP/169961-00000057”, “TOUCH_MONITOR=1471372044.7330”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:2] Set(“SIP/169961-00000057”, “AMPUSER=169961”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:3] GotoIf(“SIP/169961-00000057”, “0?report”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:4] ExecIf(“SIP/169961-00000057”, “1?Set(REALCALLERIDNUM=169961)”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:5] Set(“SIP/169961-00000057”, “AMPUSER=169961”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:6] GotoIf(“SIP/169961-00000057”, “0?limit”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:7] Set(“SIP/169961-00000057”, “AMPUSERCIDNAME=Test”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:8] GotoIf(“SIP/169961-00000057”, “0?report”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:9] Set(“SIP/169961-00000057”, “AMPUSERCID=169961”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:10] Set(“SIP/169961-00000057”, “__DIAL_OPTIONS=Ttr”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:11] Set(“SIP/169961-00000057”, “CALLERID(all)=“Test” <169961>”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:12] GotoIf(“SIP/169961-00000057”, “0?limit”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:13] ExecIf(“SIP/169961-00000057”, “0?Set(GROUP(concurrency_limit)=169961)”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:14] GosubIf(“SIP/169961-00000057”, “7?sub-ccss,s,1(from-internal,*98)”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@sub-ccss:1] ExecIf(“SIP/169961-00000057”, “0?Return()”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@sub-ccss:2] Set(“SIP/169961-00000057”, “CCSS_SETUP=TRUE”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@sub-ccss:3] GosubIf(“SIP/169961-00000057”, “0?monitor_config,1(from-internal,*98):monitor_default,1(from-internal,*98)”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [monitor_default@sub-ccss:1] GotoIf(“SIP/169961-00000057”, “0?is_exten”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [monitor_default@sub-ccss:2] StackPop(“SIP/169961-00000057”, “”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [monitor_default@sub-ccss:3] Return(“SIP/169961-00000057”, “FALSE”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:15] ExecIf(“SIP/169961-00000057”, “0?Set(CHANNEL(language)=)”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:16] GotoIf(“SIP/169961-00000057”, “0?continue”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:17] ExecIf(“SIP/169961-00000057”, “1?Set(__CALLEE_ACCOUNCODE=)”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:18] Set(“SIP/169961-00000057”, “__TTL=64”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:19] GotoIf(“SIP/169961-00000057”, “1?continue”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Goto (macro-user-callerid,s,30)
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:30] Set(“SIP/169961-00000057”, “CALLERID(number)=169961”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:31] Set(“SIP/169961-00000057”, “CALLERID(name)=Test”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:32] Set(“SIP/169961-00000057”, “CDR(cnum)=169961”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:33] Set(“SIP/169961-00000057”, “CDR(cnam)=Test”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [s@macro-user-callerid:34] Set(“SIP/169961-00000057”, “CHANNEL(language)=en”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [*98@from-internal:2] Answer(“SIP/169961-00000057”, “”) in new stack
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] chan_sip.c: Audio is at 10674
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] chan_sip.c: Adding codec ulaw to SDP
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] chan_sip.c: Adding codec alaw to SDP
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] chan_sip.c: Adding codec ilbc to SDP
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] chan_sip.c: Adding codec gsm to SDP
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] chan_sip.c:
<— Reliably Transmitting (NAT) to 96.39.54.46:1079 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.39.54.46:1079;branch=z9hG4bK670948151;received=96.39.54.46;rport=1079
From: sip:169961@173.166.10.30;tag=140578744
To: sip:*98@173.166.10.30;tag=as1b09e4ce
Call-ID: 1067649114
CSeq: 21 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:*98@10.1.10.5:5060
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 1903141783 1903141783 IN IP4 10.1.10.5
s=Asterisk PBX 13.2.0
c=IN IP4 10.1.10.5
t=0 0
m=audio 10674 RTP/AVP 0 8 97 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:30
a=sendrecv

<------------>
[2016-08-16 14:27:24] VERBOSE[3170] chan_sip.c: Retransmitting #1 (NAT) to 96.39.54.46:1079:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.39.54.46:1079;branch=z9hG4bK670948151;received=96.39.54.46;rport=1079
From: sip:169961@173.166.10.30;tag=140578744
To: sip:*98@173.166.10.30;tag=as1b09e4ce
Call-ID: 1067649114
CSeq: 21 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:*98@10.1.10.5:5060
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 1903141783 1903141783 IN IP4 10.1.10.5
s=Asterisk PBX 13.2.0
c=IN IP4 10.1.10.5
t=0 0
m=audio 10674 RTP/AVP 0 8 97 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:30
a=sendrecv

[2016-08-16 14:27:24] VERBOSE[3170] chan_sip.c: Retransmitting #2 (NAT) to 96.39.54.46:1079:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.39.54.46:1079;branch=z9hG4bK670948151;received=96.39.54.46;rport=1079
From: sip:169961@173.166.10.30;tag=140578744
To: sip:*98@173.166.10.30;tag=as1b09e4ce
Call-ID: 1067649114
CSeq: 21 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:*98@10.1.10.5:5060
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 1903141783 1903141783 IN IP4 10.1.10.5
s=Asterisk PBX 13.2.0
c=IN IP4 10.1.10.5
t=0 0
m=audio 10674 RTP/AVP 0 8 97 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:30
a=sendrecv

[2016-08-16 14:27:24] VERBOSE[14002][C-000002a3] pbx.c: Executing [*98@from-internal:3] Wait(“SIP/169961-00000057”, “1”) in new stack
[2016-08-16 14:27:24] VERBOSE[3170] chan_sip.c: Retransmitting #3 (NAT) to 96.39.54.46:1079:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.39.54.46:1079;branch=z9hG4bK670948151;received=96.39.54.46;rport=1079
From: sip:169961@173.166.10.30;tag=140578744
To: sip:*98@173.166.10.30;tag=as1b09e4ce
Call-ID: 1067649114
CSeq: 21 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:*98@10.1.10.5:5060
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 1903141783 1903141783 IN IP4 10.1.10.5
s=Asterisk PBX 13.2.0
c=IN IP4 10.1.10.5
t=0 0
m=audio 10674 RTP/AVP 0 8 97 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:30
a=sendrecv

[2016-08-16 14:27:25] VERBOSE[3170] chan_sip.c: Retransmitting #4 (NAT) to 96.39.54.46:1079:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.39.54.46:1079;branch=z9hG4bK670948151;received=96.39.54.46;rport=1079
From: sip:169961@173.166.10.30;tag=140578744
To: sip:*98@173.166.10.30;tag=as1b09e4ce
Call-ID: 1067649114
CSeq: 21 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:*98@10.1.10.5:5060
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 1903141783 1903141783 IN IP4 10.1.10.5
s=Asterisk PBX 13.2.0
c=IN IP4 10.1.10.5
t=0 0
m=audio 10674 RTP/AVP 0 8 97 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:30
a=sendrecv

[2016-08-16 14:27:25] VERBOSE[14002][C-000002a3] pbx.c: Executing [*98@from-internal:4] NoOp(“SIP/169961-00000057”, “app-dialvm: Asking for mailbox”) in new stack
[2016-08-16 14:27:25] VERBOSE[14002][C-000002a3] pbx.c: Executing [*98@from-internal:5] Read(“SIP/169961-00000057”, “MAILBOX,vm-login,3,2”) in new stack
[2016-08-16 14:27:25] VERBOSE[14002][C-000002a3] file.c: <SIP/169961-00000057> Playing ‘vm-login.ulaw’ (language ‘en’)
[2016-08-16 14:27:27] VERBOSE[3170] chan_sip.c: Retransmitting #5 (NAT) to 96.39.54.46:1079:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.39.54.46:1079;branch=z9hG4bK670948151;received=96.39.54.46;rport=1079
From: sip:169961@173.166.10.30;tag=140578744
To: sip:*98@173.166.10.30;tag=as1b09e4ce
Call-ID: 1067649114
CSeq: 21 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:*98@10.1.10.5:5060
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 1903141783 1903141783 IN IP4 10.1.10.5
s=Asterisk PBX 13.2.0
c=IN IP4 10.1.10.5
t=0 0
m=audio 10674 RTP/AVP 0 8 97 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:30
a=sendrecv

[2016-08-16 14:27:30] VERBOSE[14002][C-000002a3] app_read.c: User entered nothing, 2 chances left
[2016-08-16 14:27:30] VERBOSE[14002][C-000002a3] file.c: <SIP/169961-00000057> Playing ‘vm-login.ulaw’ (language ‘en’)
[2016-08-16 14:27:30] VERBOSE[3170] chan_sip.c: Retransmitting #6 (NAT) to 96.39.54.46:1079:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.39.54.46:1079;branch=z9hG4bK670948151;received=96.39.54.46;rport=1079
From: sip:169961@173.166.10.30;tag=140578744
To: sip:*98@173.166.10.30;tag=as1b09e4ce
Call-ID: 1067649114
CSeq: 21 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:*98@10.1.10.5:5060
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 1903141783 1903141783 IN IP4 10.1.10.5
s=Asterisk PBX 13.2.0
c=IN IP4 10.1.10.5
t=0 0
m=audio 10674 RTP/AVP 0 8 97 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:30
a=sendrecv

[2016-08-16 14:27:30] VERBOSE[3170] chan_sip.c:
<— SIP read from UDP:96.39.54.46:1079 —>
jaK
<------------->
[2016-08-16 14:27:30] WARNING[3170] chan_sip.c: Retransmission timeout reached on transmission 1067649114 for seqno 21 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2016-08-16 14:27:30] WARNING[3170] chan_sip.c: Hanging up call 1067649114 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2016-08-16 14:27:30] VERBOSE[14002][C-000002a3] app_read.c: User disconnected
[2016-08-16 14:27:30] VERBOSE[14002][C-000002a3] pbx.c: Executing [h@from-internal:1] Hangup(“SIP/169961-00000057”, “”) in new stack
[2016-08-16 14:27:30] VERBOSE[14002][C-000002a3] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/169961-00000057’
[2016-08-16 14:27:30] VERBOSE[14002][C-000002a3] chan_sip.c: Scheduling destruction of SIP dialog ‘1067649114’ in 6400 ms (Method: INVITE)
[2016-08-16 14:27:30] VERBOSE[14002][C-000002a3] chan_sip.c: Reliably Transmitting (NAT) to 96.39.54.46:1079:
BYE sip:6969@96.39.54.46 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.5:5060;branch=z9hG4bK5d2fafbb;rport
Max-Forwards: 70
From: sip:*98@173.166.10.30;tag=as1b09e4ce
To: sip:169961@173.166.10.30;tag=140578744
Call-ID: 1067649114
CSeq: 102 BYE
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.2.0)
Proxy-Authorization: Digest username=“169961”, realm=“asterisk”, algorithm=MD5, uri=“sip:173.166.10.30”, nonce=“475ea3a1”, response="919131675938089bd55717023ea088b6"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0

[2016-08-16 14:27:30] VERBOSE[3170] chan_sip.c:
<— SIP read from UDP:96.39.54.46:1079 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.5:5060;branch=z9hG4bK5d2fafbb;rport=5060;received=173.166.10.30
From: sip:*98@173.166.10.30;tag=as1b09e4ce
To: sip:169961@173.166.10.30;tag=140578744
Call-ID: 1067649114
CSeq: 102 BYE
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Content-Length: 0

<------------->
[2016-08-16 14:27:30] VERBOSE[3170] chan_sip.c: — (8 headers 0 lines) —
[2016-08-16 14:27:30] VERBOSE[3170][C-000002a3] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[2016-08-16 14:27:30] VERBOSE[3170] chan_sip.c: Really destroying SIP dialog ‘1067649114’ Method: INVITE
[2016-08-16 14:27:41] NOTICE[3170] chan_sip.c: – Re-registration for 9000@10.9.0.2
[2016-08-16 14:27:41] NOTICE[3170] chan_sip.c: Outbound Registration: Expiry for 10.9.0.2 is 120 sec (Scheduling reregistration in 105 s)
`

6

I am also getting this depending on the softphone I use

<--- SIP read from UDP:96.39.54.46:1079 --->
jaK
<------------->
7

Your Asterisk is responding that media should be sent to “10.1.10.5” which since your softphone is coming from a public IP seems to be wrong. What is the configuration?

8

I have both the public and private in FreePBX but this is what is in sip_general_additional.conf
accept_outofcall_message=yes
auth_message_requests=no
outofcall_message_context=dpma_message_context
faxdetect=no
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-AsteriskNOW-12.0.76.4(13.2.0)
disallow=all
allow=ulaw
allow=ilbc
allow=alaw
allow=gsm
callevents=yes
rtpstart=10000
rtpend=20000
bindport=5060
jbenable=no
allowguest=no
srvlookup=no
defaultexpiry=120
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
g726nonstandard=no
videosupport=no
maxcallbitrate=384
canreinvite=update
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
checkmwi=10
notifyringing=yes
notifyhold=yes
nat=route
ALLOW_SIP_ANON=no
localnet=10.1.10.0/24
localnet=10.9.0.0/23

from the gui
NAT Settings

These settings apply to both chan_sip and chan_pjsip.
External Address
173.166.10.30
Detect External IP
Local Networks
10.1.10.0
/
24

10.9.0.0
/
23

`

9

10.1.10.5 is the asterisknow box.

10

Here is the config for that account:
[169961]
deny=0.0.0.0/0.0.0.0
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
mediaencryption=no
sendrpid=no
type=friend
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/169961
permit=0.0.0.0/0.0.0.0
callerid=Test <169961>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

11

FreePBX does not appear to have told Asterisk its public IP address, so it is not replacing it. I don’t know FreePBX so I can’t help with it I’m afraid.