Outgoing call drop-outs

Asterisk Asterisk Support
1

Hey all.

Ive been trying to figure out why my outgoing calls just drop out on answering.

Ives tried two different modems.

Ive also tried installing ASterisk 13 and 11 but the same thing still happends.

Upgraded the firmware on a brand new adsl2+ modem…still got problems.

Might note that im not sure if port forwarding needs enabling, however at the moment it is disabled.

I’m using Cisco 7940 Telephones with SIP.

Not sure what to do, just wondering if anyone could help.

Thanks.

heres my log with sip debugging enabled just after a failed call.

Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 102 INVITE
Server: SIMTEX_11_0614
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="09f4dcb1"
Content-Length: 0

<------------->
[2015-01-18 20:43:57] VERBOSE[1859] chan_sip.c: — (11 headers 0 lines) —
[2015-01-18 20:43:57] VERBOSE[1859][C-00000015] chan_sip.c: Transmitting (no NAT) to 203.30.19.164:5060:
ACK sip:1800025123@sip.simtex.com.au:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK651892ff
Max-Forwards: 70
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060;tag=as5a37e112
Contact: sip:64864248029@192.168.1.5:5060
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 102 ACK
User-Agent: FPBX-AsteriskNOW-12.0.32(11.14.1)
Content-Length: 0

[2015-01-18 20:43:57] VERBOSE[1859][C-00000015] chan_sip.c: Audio is at 19744
[2015-01-18 20:43:57] VERBOSE[1859][C-00000015] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[2015-01-18 20:43:57] VERBOSE[1859][C-00000015] chan_sip.c: Adding codec 100004 (alaw) to SDP
[2015-01-18 20:43:57] VERBOSE[1859][C-00000015] chan_sip.c: Adding codec 100002 (gsm) to SDP
[2015-01-18 20:43:57] VERBOSE[1859][C-00000015] chan_sip.c: Adding codec 100011 (g726) to SDP
[2015-01-18 20:43:57] VERBOSE[1859][C-00000015] chan_sip.c: Reliably Transmitting (no NAT) to 203.30.19.164:5060:
INVITE sip:1800025123@sip.simtex.com.au:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK4319088c
Max-Forwards: 70
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060
Contact: sip:64864248029@192.168.1.5:5060
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 103 INVITE
User-Agent: FPBX-AsteriskNOW-12.0.32(11.14.1)
Authorization: Digest username=“71914448”, realm=“asterisk”, algorithm=MD5, uri=“sip:1800025123@sip.simtex.com.au:5060”, nonce=“09f4dcb1”, response="2ea3443c0a29fb898cd12fa55345cacb"
Date: Sun, 18 Jan 2015 12:43:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 1744966065 1744966066 IN IP4 192.168.1.5
s=Asterisk PBX 11.14.1
c=IN IP4 192.168.1.5
t=0 0
m=audio 19744 RTP/AVP 0 8 3 111
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=ptime:20
a=sendrecv

[2015-01-18 20:43:57] VERBOSE[1859] chan_sip.c: Retransmitting #1 (no NAT) to 203.30.19.164:5060:
INVITE sip:1800025123@sip.simtex.com.au:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK4319088c
Max-Forwards: 70
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060
Contact: sip:64864248029@192.168.1.5:5060
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 103 INVITE
User-Agent: FPBX-AsteriskNOW-12.0.32(11.14.1)
Authorization: Digest username=“71914448”, realm=“asterisk”, algorithm=MD5, uri=“sip:1800025123@sip.simtex.com.au:5060”, nonce=“09f4dcb1”, response="2ea3443c0a29fb898cd12fa55345cacb"
Date: Sun, 18 Jan 2015 12:43:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 1744966065 1744966066 IN IP4 192.168.1.5
s=Asterisk PBX 11.14.1
c=IN IP4 192.168.1.5
t=0 0
m=audio 19744 RTP/AVP 0 8 3 111
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=ptime:20
a=sendrecv

[2015-01-18 20:43:57] VERBOSE[1859] chan_sip.c:
<— SIP read from UDP:203.30.19.164:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK651892ff;received=202.165.87.43;rport=6331
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060;tag=as5a37e112
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 102 INVITE
Server: SIMTEX_11_0614
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="09f4dcb1"
Content-Length: 0

<------------->
[2015-01-18 20:43:57] VERBOSE[1859] chan_sip.c: — (11 headers 0 lines) —
[2015-01-18 20:43:57] VERBOSE[1859][C-00000015] chan_sip.c: Transmitting (no NAT) to 203.30.19.164:5060:
ACK sip:1800025123@sip.simtex.com.au:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK4319088c
Max-Forwards: 70
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060
Contact: sip:64864248029@192.168.1.5:5060
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 102 ACK
User-Agent: FPBX-AsteriskNOW-12.0.32(11.14.1)
Content-Length: 0

[2015-01-18 20:43:57] VERBOSE[1859] chan_sip.c:
<— SIP read from UDP:203.30.19.164:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK4319088c;received=202.165.87.43;rport=6331
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 103 INVITE
Server: SIMTEX_11_0614
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:1800025123@203.30.19.164:5060
Content-Length: 0

<------------->
[2015-01-18 20:43:57] VERBOSE[1859] chan_sip.c: — (12 headers 0 lines) —
[2015-01-18 20:43:57] VERBOSE[1859] chan_sip.c:
<— SIP read from UDP:203.30.19.164:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK4319088c;received=202.165.87.43;rport=6331
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 103 INVITE
Server: SIMTEX_11_0614
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:1800025123@203.30.19.164:5060
Content-Length: 0

<------------->
[2015-01-18 20:43:57] VERBOSE[1859] chan_sip.c: — (12 headers 0 lines) —
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c:
<— SIP read from UDP:203.30.19.164:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK4319088c;received=202.165.87.43;rport=6331
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060;tag=as44ea7fb2
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 103 INVITE
Server: SIMTEX_11_0614
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 227

v=0
o=root 915189457 915189457 IN IP4 203.30.19.164
s=Asterisk PBX 11.10.2
c=IN IP4 203.30.19.164
t=0 0
m=audio 13798 RTP/AVP 3 8 0
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c: — (11 headers 11 lines) —
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Found RTP audio format 3
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Found RTP audio format 8
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Found RTP audio format 0
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Found audio description format GSM for ID 3
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Found audio description format PCMA for ID 8
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Found audio description format PCMU for ID 0
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(gsm|ulaw|alaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Peer audio RTP is at port 203.30.19.164:13798
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: list_route: no route
[2015-01-18 20:44:01] WARNING[1859][C-00000015] chan_sip.c: Invalid contact uri (missing sip: or sips:), attempting to use anyway
[2015-01-18 20:44:01] WARNING[1859][C-00000015] chan_sip.c: Invalid URI: parse_uri failed to acquire hostport
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Transmitting (no NAT) to 203.30.19.164:5060:
ACK sip:1800025123@sip.simtex.com.au:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK0eb62f43
Max-Forwards: 70
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060;tag=as44ea7fb2
Contact: sip:64864248029@192.168.1.5:5060
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 103 ACK
User-Agent: FPBX-AsteriskNOW-12.0.32(11.14.1)
Content-Length: 0

[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Reliably Transmitting (no NAT) to 203.30.19.164:5060:
BYE sip:1800025123@sip.simtex.com.au:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK6bfed116
Max-Forwards: 70
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060;tag=as44ea7fb2
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 104 BYE
User-Agent: FPBX-AsteriskNOW-12.0.32(11.14.1)
Authorization: Digest username=“71914448”, realm=“asterisk”, algorithm=MD5, uri=“sip:1800025123@sip.simtex.com.au:5060”, nonce=“09f4dcb1”, response="ce0b1486fdb1a20eb7c9b33714a23839"
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0

[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Scheduling destruction of SIP dialog ‘1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060’ in 8128 ms (Method: INVITE)
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] app_dial.c: – SIP/71914448-0000002c answered SIP/2000-0000002b
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] chan_sip.c: Audio is at 11476
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] chan_sip.c: Adding codec 100004 (alaw) to SDP
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] chan_sip.c:
<— Reliably Transmitting (NAT) to 192.168.1.112:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK0214dff2;received=192.168.1.112;rport=5060
From: “2000” sip:2000@192.168.1.5;tag=00137fdd34360080127c0a6c-24da93e7
To: sip:01800025123@192.168.1.5;user=phone;tag=as6a10bcf6
Call-ID: 00137fdd-34360013-49d70e23-6166c5e7@192.168.1.112
CSeq: 102 INVITE
Server: FPBX-AsteriskNOW-12.0.32(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:01800025123@192.168.1.5:5060
P-Asserted-Identity: “CID:64864248029” sip:1800025123@192.168.1.5
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 103196951 103196951 IN IP4 192.168.1.5
s=Asterisk PBX 11.14.1
c=IN IP4 192.168.1.5
t=0 0
m=audio 11476 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

<------------>
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] pbx.c: – Executing [h@macro-dialout-trunk:1] Macro(“SIP/2000-0000002b”, “hangupcall,”) in new stack
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/2000-0000002b”, “1?theend”) in new stack
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] pbx.c: – Goto (macro-hangupcall,s,3)
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] pbx.c: – Executing [s@macro-hangupcall:3] ExecIf(“SIP/2000-0000002b”, “0?Set(CDR(recordingfile)=)”) in new stack
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] pbx.c: – Executing [s@macro-hangupcall:4] Hangup(“SIP/2000-0000002b”, “”) in new stack
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/2000-0000002b’ in macro ‘hangupcall’
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] pbx.c: == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on ‘SIP/2000-0000002b’
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] chan_sip.c: Scheduling destruction of SIP dialog ‘1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060’ in 8128 ms (Method: INVITE)
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] chan_sip.c: Reliably Transmitting (no NAT) to 203.30.19.164:5060:
BYE sip:1800025123@sip.simtex.com.au:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK7b6742b5
Max-Forwards: 70
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060;tag=as44ea7fb2
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 105 BYE
User-Agent: FPBX-AsteriskNOW-12.0.32(11.14.1)
Authorization: Digest username=“71914448”, realm=“asterisk”, algorithm=MD5, uri=“sip:1800025123@sip.simtex.com.au:5060”, nonce=“09f4dcb1”, response="ce0b1486fdb1a20eb7c9b33714a23839"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on ‘SIP/2000-0000002b’ in macro ‘dialout-trunk’
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] pbx.c: == Spawn extension (from-internal, 01800025123, 5) exited non-zero on ‘SIP/2000-0000002b’
[2015-01-18 20:44:01] VERBOSE[16286][C-00000015] chan_sip.c: Scheduling destruction of SIP dialog ‘00137fdd-34360013-49d70e23-6166c5e7@192.168.1.112’ in 6400 ms (Method: INVITE)
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.112:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK0214dff2;received=192.168.1.112;rport=5060
From: “2000” sip:2000@192.168.1.5;tag=00137fdd34360080127c0a6c-24da93e7
To: sip:01800025123@192.168.1.5;user=phone;tag=as6a10bcf6
Call-ID: 00137fdd-34360013-49d70e23-6166c5e7@192.168.1.112
CSeq: 102 INVITE
Server: FPBX-AsteriskNOW-12.0.32(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:01800025123@192.168.1.5:5060
P-Asserted-Identity: “CID:64864248029” sip:1800025123@192.168.1.5
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 103196951 103196951 IN IP4 192.168.1.5
s=Asterisk PBX 11.14.1
c=IN IP4 192.168.1.5
t=0 0
m=audio 11476 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

[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c:
<— SIP read from UDP:203.30.19.164:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK4319088c;received=202.165.87.43;rport=6331
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060;tag=as44ea7fb2
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 103 INVITE
Server: SIMTEX_11_0614
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 227

v=0
o=root 915189457 915189457 IN IP4 203.30.19.164
s=Asterisk PBX 11.10.2
c=IN IP4 203.30.19.164
t=0 0
m=audio 13798 RTP/AVP 3 8 0
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c: — (11 headers 11 lines) —
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Transmitting (no NAT) to 203.30.19.164:5060:
ACK sip:1800025123@sip.simtex.com.au:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK04966f2e
Max-Forwards: 70
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060;tag=as44ea7fb2
Contact: sip:64864248029@192.168.1.5:5060
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 103 ACK
User-Agent: FPBX-AsteriskNOW-12.0.32(11.14.1)
Content-Length: 0

[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c: Retransmitting #1 (no NAT) to 203.30.19.164:5060:
BYE sip:1800025123@sip.simtex.com.au:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK6bfed116
Max-Forwards: 70
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060;tag=as44ea7fb2
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 104 BYE
User-Agent: FPBX-AsteriskNOW-12.0.32(11.14.1)
Authorization: Digest username=“71914448”, realm=“asterisk”, algorithm=MD5, uri=“sip:1800025123@sip.simtex.com.au:5060”, nonce=“09f4dcb1”, response="ce0b1486fdb1a20eb7c9b33714a23839"
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0

[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c:
<— SIP read from UDP:203.30.19.164:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK6bfed116;received=202.165.87.43;rport=6331
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060;tag=as44ea7fb2
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 104 BYE
Server: SIMTEX_11_0614
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------->
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c: — (10 headers 0 lines) —
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c: Retransmitting #1 (no NAT) to 203.30.19.164:5060:
BYE sip:1800025123@sip.simtex.com.au:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK7b6742b5
Max-Forwards: 70
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060;tag=as44ea7fb2
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 105 BYE
User-Agent: FPBX-AsteriskNOW-12.0.32(11.14.1)
Authorization: Digest username=“71914448”, realm=“asterisk”, algorithm=MD5, uri=“sip:1800025123@sip.simtex.com.au:5060”, nonce=“09f4dcb1”, response="ce0b1486fdb1a20eb7c9b33714a23839"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c:
<— SIP read from UDP:203.30.19.164:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK7b6742b5;received=202.165.87.43;rport=6331
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060;tag=as44ea7fb2
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 105 BYE
Server: SIMTEX_11_0614
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------->
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c: — (10 headers 0 lines) —
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c: Really destroying SIP dialog ‘1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060’ Method: INVITE
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c:
<— SIP read from UDP:192.168.1.112:5060 —>
ACK sip:01800025123@192.168.1.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK1fca1794
From: “2000” sip:2000@192.168.1.5;tag=00137fdd34360080127c0a6c-24da93e7
To: sip:01800025123@192.168.1.5;user=phone;tag=as6a10bcf6
Call-ID: 00137fdd-34360013-49d70e23-6166c5e7@192.168.1.112
Max-Forwards: 70
CSeq: 102 ACK
User-Agent: Cisco-CP7940G/8.0
Authorization: Digest username=“2000”,realm=“asterisk”,uri="sip:01800025123@192.168.1.5;user=phone",response=“2def72dfb37ade7e88baab90312f4537”,nonce=“6319513a”,algorithm=MD5
Remote-Party-ID: “2000” sip:2000@192.168.1.5;party=calling;id-type=subscriber;privacy=off;screen=yes
Content-Length: 0

<------------->
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c: — (11 headers 0 lines) —
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: set_destination: Parsing sip:2000@192.168.1.112:5060;transport=udp for address/port to send to
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: set_destination: set destination to 192.168.1.112:5060
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.112:5060:
BYE sip:2000@192.168.1.112:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK17204e56;rport
Max-Forwards: 70
From: sip:01800025123@192.168.1.5;user=phone;tag=as6a10bcf6
To: “2000” sip:2000@192.168.1.5;tag=00137fdd34360080127c0a6c-24da93e7
Call-ID: 00137fdd-34360013-49d70e23-6166c5e7@192.168.1.112
CSeq: 102 BYE
User-Agent: FPBX-AsteriskNOW-12.0.32(11.14.1)
Proxy-Authorization: Digest username=“2000”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.5”, nonce=“6319513a”, response="3cb32a60cd733638ad9a2699633bde58"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: Scheduling destruction of SIP dialog ‘00137fdd-34360013-49d70e23-6166c5e7@192.168.1.112’ in 6400 ms (Method: ACK)
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c:
<— SIP read from UDP:203.30.19.164:5060 —>
SIP/2.0 500 Server error
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK6bfed116;received=202.165.87.43;rport=6331
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060;tag=as44ea7fb2
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 104 BYE
Server: SIMTEX_11_0614
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------->
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c: — (10 headers 0 lines) —
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c:
<— SIP read from UDP:203.30.19.164:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK7b6742b5;received=202.165.87.43;rport=6331
From: sip:64864248029@192.168.1.5;tag=as7ebeb63b
To: sip:1800025123@sip.simtex.com.au:5060;tag=as44ea7fb2
Call-ID: 1ddc8a6f0803024028e0e21b0fb165d5@192.168.1.5:5060
CSeq: 105 BYE
Server: SIMTEX_11_0614
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------->
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c: — (10 headers 0 lines) —
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c:
<— SIP read from UDP:192.168.1.112:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK17204e56;rport
From: sip:01800025123@192.168.1.5;user=phone;tag=as6a10bcf6
To: “2000” sip:2000@192.168.1.5;tag=00137fdd34360080127c0a6c-24da93e7
Call-ID: 00137fdd-34360013-49d70e23-6166c5e7@192.168.1.112
CSeq: 102 BYE
Server: Cisco-CP7940G/8.0
Content-Length: 0

<------------->
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c: — (8 headers 0 lines) —
[2015-01-18 20:44:01] VERBOSE[1859][C-00000015] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[2015-01-18 20:44:01] VERBOSE[1859] chan_sip.c: Really destroying SIP dialog ‘00137fdd-34360013-49d70e23-6166c5e7@192.168.1.112’ Method: ACK

2

The peer is not SIP compliant. It is violating this rule:

[code] Header field where proxy ACK BYE CAN INV OPT REG
___________________________________________________________

  Contact                2xx           -   -   -   m   o   o

[/code]

“m” means it must be present, but it wasn’t present.

Also, you seem to have an ureliable connection (there is a retransmission) and your trace is incomplete (the CSEQ: 103 INVITE request was obviously present, but has not been logged).

Some times disabling pedantic SIP mode can work round broken peers. I don’t know if it will in this case.

3

Thankyou,

May I ask how you modify the setting to allow M?

I understand that this is Invite? for the peer details? it is set to “no”

4

I might also add at this point that I have no information added into the “user” details of the trunk.

5

M means that the field is mandatory, i.e. the peer is broken if it doesn’t set it. The contact header should contain xxxxx@yyyyy, where xxxxx is only relevant to the peer and yyyyy is the domain name or addres to which the peer expects further SIP requests to be sent.

As I said, disabling pedantic checking can work with some broken peers, but it may be that you have to get the peer fixed.

Strangely the peer is sending it on 100, where it is only optional.

“user details of trunk” is FreePBX terminology; Asterisk doesn’t have a concept of a SIP trunk. You need to use community.freepbx.org for questions about that. However, I don’t think that is a problem, as the peer is accepting the call, just not doing so in a way allowed by SIP.

6

Hi,

Thanks so much for your reply, unfortunately I dont quite understand the technical terms and what they mean.

I’m new to this kind of system, I’m sorry about this.

From what I understand, you think that this might be a problem with my SIP provider?

Also, I might note that I have a dynamic IP address on my DSL connection.

7

You also have a problem with address translation. You haven’t told Asterisk how to find its public address. However, that doesn’t remove the problem that your SIP provider is providing something which is not SIP.

8

hmmmmmmmmmm hey thanks for your reply.

I spoke to the engineer at my sip provider, and he says that I need to disable SIP ALG in my router?

9

Yes, it could also be a broken router. Most people have had bad experiences with SIP ALG.

You still need to set at least one method for getting Asterisk to know its public address. There are alternatives which are described in the sample configuration file and is also such a frequent issue that there will be a lot on the web.

10

Turnes out it was SIP ALG that needed to be turned off by telnetting the router.

Thanks for you help anyway I appreciate it!