Server Stats:
Asterisk 1.8.20.1
OS: Centos 5.9
Kernel version: 2.6.18-348.3.1.el5
Network diagram
AT&T —> T1 —> Cisco Router ----> Local Network Switch <----- Asterisk
I have had a working Asterisk server for a few years now, incoming phone service is AT&T PRI on a T1. I have a Digium TE121 card, using Dahdi and everything is fine. Fast forward to a week ago trying to do test and turn up to switch our service to AT&T IP Flexible Reach. All the AT&T hardware is in place and tested (2 - T1 cards, Cisco router, etc.).
I can place a call from my cell phone to a DID test number. It hits Asterisk. I hear my IVR then about 30 seconds into the call, it hangs up.
Snip from CLI with SIP debug on:
[code]Retransmitting #2 (NAT) to XX.XXX.XXX.XX:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XXX.XXX.XX:5060;branch=z9hG4bKhgpr5300d8qg9g89g340.1;received=XX.XXX.XXX.XX;rport=5060
From: “WIRELESS CALLER” sip:XXXXXXXXXX@XX.XXX.XXX.XX:5060;tag=SDfu34001-0439375144239087_c1b01.1.1.1363156247884.0_1088360_2150343
To: sip:XXXXXXXXXX@XXX.XX.XX.XX;tag=as11c0d885
Call-ID: SDfu34001-c2dbea6ac0bde01047e0701646579c8e-cggtr70
CSeq: 2 INVITE
Server: Glastender PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:XXXXXXX@XXX.XX.XX.XX:5060
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 190297827 190297827 IN IP4 XXX.XX.XX.XX
s=Asterisk PBX 1.8.20.1
c=IN IP4 XXX.XX.XX.XX
t=0 0
m=audio 19508 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=sendrecv
[Mar 28 09:46:39] WARNING[30837]: chan_sip.c:3974 retrans_pkt: Retransmission timeout reached on transmission SDfu34001-c2dbea6ac0bde01047e0701646579c8e-cggtr70 for seqno 2 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Mar 28 09:46:39] WARNING[30837]: chan_sip.c:4003 retrans_pkt: Hanging up call SDfu34001-c2dbea6ac0bde01047e0701646579c8e-cggtr70 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Spawn extension (day-menu, s, 3) exited non-zero on 'SIP/first_gateway-00000c0e’
Scheduling destruction of SIP dialog ‘SDfu34001-c2dbea6ac0bde01047e0701646579c8e-cggtr70’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:XX.XXX.XXX.XX:5060;transport=udp for address/port to send to
set_destination: set destination to XX.XXX.XXX.XX:5060
Reliably Transmitting (NAT) to XX.XXX.XXX.XX:5060:
BYE sip:XX.XXX.XXX.XX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP XXX.XX.XX.XX:5060;branch=z9hG4bK3a68b925;rport
Max-Forwards: 70
From: sip:XXXXXXXXXX@XXX.XX.XX.XX;tag=as11c0d885
To: “WIRELESS CALLER” sip:XXXXXXXXXX@XX.XXX.XXX.XX:5060;tag=SDfu34001-0439375144239087_c1b01.1.1.1363156247884.0_1088360_2150343
Call-ID: SDfu34001-c2dbea6ac0bde01047e0701646579c8e-cggtr70
CSeq: 102 BYE
User-Agent: Glastender PBX
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
[/code]
Tshark snip:
1002.507728 XXX.XX.XX.XX -> XX.XXX.XXX.XX SIP/SDP Status: 200 OK, with session description
1002.508588 XXX.XX.XX.XX -> XX.XXX.XXX.XX RTP PT=ITU-T G.711 PCMU, SSRC=0x704B955A, Seq=29003, Time=42560
1002.528621 XXX.XX.XX.XX -> XX.XXX.XXX.XX RTP PT=ITU-T G.711 PCMU, SSRC=0x704B955A, Seq=29004, Time=42720
1002.548619 XXX.XX.XX.XX -> XX.XXX.XXX.XX RTP PT=ITU-T G.711 PCMU, SSRC=0x704B955A, Seq=29005, Time=42880
1002.568614 XXX.XX.XX.XX -> XX.XXX.XXX.XX RTP PT=ITU-T G.711 PCMU, SSRC=0x704B955A, Seq=29006, Time=43040
1002.588637 XXX.XX.XX.XX -> XX.XXX.XXX.XX RTP PT=ITU-T G.711 PCMU, SSRC=0x704B955A, Seq=29007, Time=43200
1002.608619 XXX.XX.XX.XX -> XX.XXX.XXX.XX RTP PT=ITU-T G.711 PCMU, SSRC=0x704B955A, Seq=29008, Time=43360
1002.609089 XXX.XX.XX.XX -> XX.XXX.XXX.XX SIP Request: BYE sip:XX.XXX.XXX.XX:5060;transport=udp
1002.679143 XX.XXX.XXX.XX -> XXX.XX.XX.XX SIP Status: 200 OK
1035.418805 XXX.XX.XX.XX -> XX.XXX.XXX.XX SIP Request: OPTIONS sip:XX.XXX.XXX.XX
1035.463751 XX.XXX.XXX.XX -> XXX.XX.XX.XX SIP Status: 405 Method Not Allowed
My sip.conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
;srvlookup=yes
disallow=all
allow=ulaw
allow=gsm
allow=alaw
language=en
trustpid=yes
sendrpid=yes
qualify=yes
callevents=yes
insecure=invite
pedantic=no
useragent=Glastender PBX
videosupport=no
t38pt_udptl=no
t38pt_rtp=no
t38pt_tcp=no
; ******************************************************
; ** Register statements
; ******************************************************
; ** Friends, Users and Peers below this line
; *********** AT&T IP FLEX *****************************
[first_gateway]
context=pbx_trunk_in
type=peer
host=XX.XXX.XXX.XX
qualify=2000
And before you say it’s a problem with NAT, I have tried ALL the various NAT configurations in my sip.conf. I tried nat=yes. I tried adding localnet, directrtpsetup=yes, canreinvite=yes…and the opposite of these. Nothing I do seems to make any difference. I’m at a loss.