same problem here
asreisk it public, client is behind adsl nat
SIP ALG is switched off
here is asterisk side:
[code]Retransmitting #9 (NAT) to 109.184.159.49:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.184.159.49:5060;branch=z9hG4bK544348511;received=109.184.159.49;rport=5060
From: sip:2003@asterisk;tag=310558963
To: sip:84959398000@asterisk;tag=as12a0f774
Call-ID: 1413745589
CSeq: 21 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:84959398000@asterisk:5060
Content-Type: application/sdp
Content-Length: 288
v=0
o=root 108756213 108756214 IN IP4 asterisk ip
s=Asterisk PBX 1.8.7.0
c=IN IP4 asterisk ip
t=0 0
m=audio 15006 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
m=video 0 RTP/AVP 103 99
[Jul 18 22:03:04] WARNING[13188]: chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission 1413745589 for seqno 21 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Jul 18 22:03:04] WARNING[13188]: chan_sip.c:3651 retrans_pkt: Hanging up call 1413745589 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Spawn extension (users, 84959398000, 2) exited non-zero on 'SIP/2003-00000d0a’
Scheduling destruction of SIP dialog ‘1413745589’ in 32000 ms (Method: INVITE)
set_destination: Parsing sip:dsultan@109.184.159.49:5060 for address/port to send to
set_destination: set destination to 109.184.159.49:5060
Reliably Transmitting (NAT) to 109.184.159.49:5060:
BYE sip:dsultan@109.184.159.49:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK5ff440aa;rport
Max-Forwards: 70
From: sip:84959398000@asetrisk;tag=as12a0f774
To: sip:2003@asterisk;tag=310558963
Call-ID: 1413745589
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.7.0
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0 [/code]
and here is what going on on the other leg
[code]SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK544348511;received=109.184.159.49;rport=5060
From: sip:2003@asterisk;tag=310558963
To: sip:84959398000@asterisk;tag=as12a0f774
Call-ID: 1413745589
CSeq: 21 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:84959398000@192.168.1.3:5060
Content-Type: application/sdp
Content-Length: 288
v=0
o=root 108756213 108756214 IN IP4 asterisk ip
s=Asterisk PBX 1.8.7.0
c=IN IP4 asterisk ip
t=0 0
m=audio 15006 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
m=video 0 RTP/AVP 103 99
message: MESSAGE REC. CALLID:1413745589
message: Message received from: asterisk ip:5060
message: This is a request
message: 2xx restransmission receveid.
message: DNS resolution with 192.168.1.3:5060
message: getaddrinfo returned the following addresses:
message: 192.168.1.3 port 5060
message: Message sent: (to dest=192.168.1.3:5060)
ACK sip:84959398000@192.168.1.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bK1886009751
From: sip:2003@asterisk;tag=310558963
To: sip:84959398000@asterisk;tag=as12a0f774
Call-ID: 1413745589
CSeq: 21 ACK
Contact: sip:dsultan@109.184.159.49
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/3.6.0)
Content-Length: 0[/code]
somehow local side send ACK to himself (192.168.1.3) when it gets invite form Asterisk
i have tried many soft phones and hard one - spa-3000 gateway
same problem everywhere, 30 seconds and outgoing call is dropped
incoming are fine