Hi,
I have phones that are connected over the internet to Asterisk. There is not a VPN or anything like that. Remote phones - and only remote phones - cut off calls after 6-30 seconds.
I am running Asterisk 11.5.1 and have portforwarded port 5060 and ports 10000 to 20000 (I understand that it is is quite risky not changing the default RTP Media Ports to something else, and I will do so after everything works).
I am very new to Asterisk and have no idea what is going on- any ideas?
Thanks!
Sam
I can’t think of a timeout that short. You need to provide logs and detailed configuration.
I don’t know what risks are involved with the port range for RTP. Attacks on VoIP are done on port 5060, not the RTP ports.
I attempted to make a call with extension 301, a remote extension. Just noticed “no reply to our critical packet”- I didn’t see that when I checked the log earlier… Could it be a problem with our firewall at the Asterisk server site? This issue occurs with ip phones at 3 different locations.
<<<>>>
You will need to create a SIP trace to be sure of which packet didn’t arrive, but I suspect that the caller didn’t respond with ACK to the 200 OK. Either it didn’t receive the response, or it sent it to the wrong place. The latter may be due to an invalid external address being configured.
So I understand that the ACK is sent by the caller to signify the 200 OK has been received and the call is to be setup? So if the ACK is sent once, then it shouldn’t be an issue since the call does connect properly and you can talk fine for the 6-30 seconds it stays connected.
Also something that I didn’t mention earlier is that the call never drops when the phone is ringing.
This is what I got when I turned on SIP debug:::
<<<>>>
The call that is probably failing is the incoming one. Your trace does not include any packets from that one.
[2014-02-17 17:01:34] WARNING[1968] chan_sip.c: Retransmission timeout reached on transmission 405e0000-57f6d2d5@ for seqno 1784110679 (Critical Response) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 7552ms with no response
[2014-02-17 17:01:34] WARNING[1968] chan_sip.c: Hanging up call 405e0000-57f6d2d5@ - no reply to our critical packet (see wiki.asterisk.org/wiki/display/ … nsmissions).
[2014-02-17 17:01:34] VERBOSE[18973][C-00000588] pbx.c: – Executing [h@macro-dialout-trunk:1] Macro(“SIP/301-00001267”, “hangupcall,”) in new stack
[2014-02-17 17:01:34] VERBOSE[18973][C-00000588] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/301-00001267”, “1?theend”) in new stack
[2014-02-17 17:01:34] VERBOSE[18973][C-00000588] pbx.c: – Goto (macro-hangupcall,s,3)
[2014-02-17 17:01:34] VERBOSE[18973][C-00000588] pbx.c: – Executing [s@macro-hangupcall:3] ExecIf(“SIP/301-00001267”, “0?Set(CDR(recordingfile)=)”) in new stack
[2014-02-17 17:01:34] VERBOSE[18973][C-00000588] pbx.c: – Executing [s@macro-hangupcall:4] Hangup(“SIP/301-00001267”, “”) in new stack
[2014-02-17 17:01:34] VERBOSE[18973][C-00000588] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/301-00001267’ in macro ‘hangupcall’
[2014-02-17 17:01:34] VERBOSE[18973][C-00000588] pbx.c: == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on ‘SIP/301-00001267’
[2014-02-17 17:01:34] VERBOSE[18973][C-00000588] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on ‘SIP/301-00001267’ in macro ‘dialout-trunk’
[2014-02-17 17:01:34] VERBOSE[18973][C-00000588] pbx.c: == Spawn extension (from-internal, , 6) exited non-zero on ‘SIP/301-00001267’
[2014-02-17 17:01:34] VERBOSE[18973][C-00000588] chan_sip.c: Scheduling destruction of SIP dialog ‘405e0000-57f6d2d5@’ in 7552 ms (Method: INVITE)
[2014-02-17 17:01:34] VERBOSE[18973][C-00000588] chan_sip.c: set_destination: Parsing sip:301@10.0.0.6:6050;transport=UDP for address/port to send to
[2014-02-17 17:01:34] VERBOSE[18973][C-00000588] chan_sip.c: set_destination: set destination to 10.0.0.6:6050
[2014-02-17 17:01:34] VERBOSE[18973][C-00000588] chan_sip.c: Reliably Transmitting (NAT) to :6050:
BYE sip:301@10.0.0.6:6050;transport=UDP SIP/2.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK4e85251c;rport
Max-Forwards: 70
From: <sip:@:6050>;tag=as4b614dbd
To: “Extension301” <sip:301@>;tag=5302405e-273-638ddaa9
Call-ID: 405e0000-57f6d2d5@
CSeq: 102 BYE
User-Agent: FPBX-2.11.0(11.5.1)
Proxy-Authorization: Digest username=“301”, realm=“asterisk”, algorithm=MD5, uri=“sip:”, nonce=“496065b8”, response=“22b86fe63319953644524261e220ee3e”
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
[2014-02-17 17:01:34] VERBOSE[1968] chan_sip.c: Retransmitting #1 (NAT) to :6050:
BYE sip:301@10.0.0.6:6050;transport=UDP SIP/2.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK4e85251c;rport
Max-Forwards: 70
From: <sip:@:6050>;tag=as4b614dbd
To: “Extension301” <sip:301@>;tag=5302405e-273-638ddaa9
Call-ID: 405e0000-57f6d2d5@
CSeq: 102 BYE
User-Agent: FPBX-2.11.0(11.5.1)
Proxy-Authorization: Digest username=“301”, realm=“asterisk”, algorithm=MD5, uri=“sip:”, nonce=“496065b8”, response=“22b86fe63319953644524261e220ee3e”
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
[2014-02-17 17:01:34] VERBOSE[1968] chan_sip.c:
<— SIP read from UDP::6050 —>
SIP/2.0 200 OK
Via:SIP/2.0/UDP :5060;rport=6050;received=;branch=z9hG4bK4e85251c
From:<sip:@:6050>;tag=as4b614dbd
To:“Extension301” <sip:301@>;tag=5302405e-273-638ddaa9
CSeq:102 BYE
User-Agent:Mitel-5224-SIP-Phone 08.00.00.04 08000F3235F5
Call-ID:405e0000-57f6d2d5@
Contact:“Extension301” sip:301@10.0.0.6:6050;transport=UDP
Allow:INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY,PRACK,UPDATE
Supported:timer,100rel,replaces
Content-Length:0
<------------->
[2014-02-17 17:01:34] VERBOSE[1968] chan_sip.c: — (11 headers 0 lines) —
[2014-02-17 17:01:34] VERBOSE[1968][C-00000588] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[2014-02-17 17:01:34] VERBOSE[1968] chan_sip.c: Really destroying SIP dialog ‘405e0000-57f6d2d5@’ Method: INVITE
[2014-02-17 17:01:34] VERBOSE[1968] chan_sip.c:
<— SIP read from UDP::6050 —>
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via:SIP/2.0/UDP :5060;rport=6050;received=;branch=z9hG4bK4e85251c
From:<sip:@:6050>;tag=as4b614dbd
To:“Extension301” <sip:301@>;tag=5302405e-273-638ddaa9
CSeq:102 BYE
User-Agent:Mitel-5224-SIP-Phone 08.00.00.04 08000F3235F5
Call-ID:405e0000-57f6d2d5@
Content-Length:0
<------------->
[2014-02-17 17:01:34] VERBOSE[1968] chan_sip.c: — (8 headers 0 lines) —