Kurt Knudsen
2008-Nov-07 19:35 UTC
[asterisk-users] Outgoing SIP calls dropped after 30 seconds.
Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode)
with a public IP address. We have our phone system setup as 172.16.2.x
that connect through the SonicWall to Asterisk. Incoming calls work
flawlessly and we no longer get one-way audio. We are only using SIP
(3 trunks now, instead of 2) and having all 3 in use is not an issue.
Problem: Make a call on a Polycom 320 IP phone to any number and (4/5
times) it will drop the call after 30 seconds. I noticed that the
little timer that pops up on the LCD on the phone is missing when a
call will be dropped. This timer appears when the phone is answered,
so I have about 30 seconds to talk to them before the call is just
dropped.
Known Causes: It's a NAT issue, I know that much, I just don't know
how to fix it. SIP debugging shows that it attempts to retransmit
packets to my phone and since it can't, it drops it after 30 seconds.
Log snippet:
-- Executing [s at macro-dialout-trunk:19]
Dial("SIP/203-b7a2b558",
"SIP/bw_outbound/+18005551212|300|") in new stack
Audio is at <public IP> port 11968
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 216.82.224.202:5060:
INVITE sip:+18881231234 at 216.82.224.202 SIP/2.0
Via: SIP/2.0/UDP <public IP>:5060;branch=z9hG4bK6ea30a1a;rport
From: "8881231234" <sip:+18881231234 at public
IP>;tag=as3ed791f3
To: <sip:+18005551212 at 216.82.224.202>
Contact: <sip:+18881231234 at public IP>
Call-ID: 28aa1a24047e1bdc3328f645766ddbbb at public IP
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 07 Nov 2008 19:06:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 21520 21520 IN IP4 151.196.61.115
s=session
c=IN IP4 <public IP>
t=0 0
m=audio 11968 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
-- Called bw_outbound/+18885551212
FreePBX*CLI>
<--- SIP read from 216.82.224.202:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060
From: "8881231234" <sip:+14105601717 at public
IP>;tag=as3ed791f3
To: <sip:+18005551212 at 216.82.224.202>
Call-ID: 28aa1a24047e1bdc3328f645766ddbbb at public IP
CSeq: 102 INVITE
Server: Bandwidth.com TRM (bw7.gold.13)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
FreePBX*CLI>
<--- SIP read from 216.82.224.202:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060
Record-Route: <sip:216.82.224.202;lr;ftag=as3ed791f3>
From: "8881231234" <sip:+18881231234 at public
IP>;tag=as3ed791f3
To: <sip:+18005551212 at 216.82.224.202>;tag=VPST50603522629853
Call-ID: 28aa1a24047e1bdc3328f645766ddbbb at public IP
CSeq: 102 INVITE
Contact: <sip:+18005551212 at 209.247.16.221:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 184
v=0
o=- 1226084867 1226084868 IN IP4 209.244.42.253
s=-
c=IN IP4 209.244.42.253
t=0 0
m=audio 64706 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (10 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 209.244.42.253:64706
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 209.244.42.253:64706
-- SIP/bw_outbound-08bf43d0 is making progress passing it to
SIP/203-b7a2b558
Audio is at public IP port 16244
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 172.16.2.203:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
172.16.2.203;branch=z9hG4bKed293df65EAFD78F;received=172.16.2.203
From: "Me" <sip:203 at public IP>;tag=28354B-27A53F00
To: <sip:18005551212 at public IP;user=phone>;tag=as600b952c
Call-ID: 992e82f4-2e300935-39c0ba22 at 172.16.2.203
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:18005551212 at public IP>
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 21520 21520 IN IP4 public IP
s=session
c=IN IP4 public IP
t=0 0
m=audio 16244 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (10 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 209.244.42.253:64706
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 209.244.42.253:64706
list_route: hop: <sip:216.82.224.202;lr;ftag=as3ed791f3>
set_destination: Parsing <sip:216.82.224.202;lr;ftag=as3ed791f3> for
address/port to send to
set_destination: set destination to 216.82.224.202, port 5060
Transmitting (no NAT) to 216.82.224.202:5060:
ACK sip:+18005551212 at 209.247.16.221:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK3c4f465e;rport
Route: <sip:216.82.224.202;lr;ftag=as3ed791f3>
From: "8881231234" <sip:+18881231234 at public
IP>;tag=as3ed791f3
To: <sip:+18005551212 at 216.82.224.202>;tag=VPST50603522629853
Contact: <sip:+18881231234 at public IP>
Call-ID: 28aa1a24047e1bdc3328f645766ddbbb at public IP
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/bw_outbound-08bf43d0 answered SIP/203-b7a2b558
Audio is at public IP port 16244
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 172.16.2.203:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.16.2.203;branch=z9hG4bKed293df65EAFD78F;received=172.16.2.203
From: "Me" <sip:203 at public IP>;tag=28354B-27A53F00
To: <sip:18005551212 at public IP;user=phone>;tag=as600b952c
Call-ID: 992e82f4-2e300935-39c0ba22 at 172.16.2.203
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:18005551212 at public IP>
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 21520 21521 IN IP4 public IP
s=session
c=IN IP4 public IP
t=0 0
m=audio 16244 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- Packet2Packet bridging SIP/203-b7a2b558 and SIP/bw_outbound-08bf43d0
Retransmitting #1 (NAT) to 172.16.2.203:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.16.2.203;branch=z9hG4bKed293df65EAFD78F;received=172.16.2.203
From: "Me" <sip:203 at public IP>;tag=28354B-27A53F00
To: <sip:18005551212 at public IP;user=phone>;tag=as600b952c
Call-ID: 992e82f4-2e300935-39c0ba22 at 172.16.2.203
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:18005551212 at public IP>
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 21520 21521 IN IP4 public IP
s=session
c=IN IP4 public IP
t=0 0
m=audio 16244 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '48e1c5682586d0b92915c00a1104adc7 at public
IP' Method: OPTIONS
Retransmitting #2 (NAT) to 172.16.2.203:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.16.2.203;branch=z9hG4bKed293df65EAFD78F;received=172.16.2.203
From: "Me" <sip:203 at public IP>;tag=28354B-27A53F00
To: <sip:18005551212 at public IP;user=phone>;tag=as600b952c
Call-ID: 992e82f4-2e300935-39c0ba22 at 172.16.2.203
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:18005551212 at public IP>
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 21520 21521 IN IP4 public IP
s=session
c=IN IP4 public IP
t=0 0
m=audio 16244 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<snip>
<Then this and call is dropped>
---
Scheduling destruction of SIP dialog
'28aa1a24047e1bdc3328f645766ddbbb at public IP' in 32000 ms (Method:
INVITE)
set_destination: Parsing <sip:216.82.224.202;lr;ftag=as3ed791f3> for
address/port to send to
set_destination: set destination to 216.82.224.202, port 5060
Reliably Transmitting (no NAT) to 216.82.224.202:5060:
BYE sip:+18005551212 at 209.247.16.221:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK5e8d995d;rport
Route: <sip:216.82.224.202;lr;ftag=as3ed791f3>
From: "8881231234" <sip:+18881231234 at public
IP>;tag=as3ed791f3
To: <sip:+18005551212 at 216.82.224.202>;tag=VPST50603522629853
Call-ID: 28aa1a24047e1bdc3328f645766ddbbb at public IP
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
--------------END---------------
Doesn't matter if I have my SIP phone setting as nat=yes or nat=no.
Settings:
Outbound to Bandwidth.com: nat=no
Inbound from Bandwidth.com: nat=no
<Tried nat=yes on 1 and both. No effect>
SIP Phone:
nat=yes <doesn't matter if it's nat=no>
qualify=yes
SonicWall stuff I can't really post, but if you ask about it, I can
answer. Basically, we have an interface that's the phone system that
is assigned a static IP. SIP transformation is on and consistent NAT
is off under VOIP settings. The phone interface has VOIP/SIP allowed
and just blocks intrusion attempts.
Question: Why does it sometimes work and sometimes not? This makes no
sense and it happens on all phones. Any suggestions?
Kurt Knudsen wrote:> Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) > with a public IP address. We have our phone system setup as 172.16.2.x > that connect through the SonicWall to Asterisk. Incoming calls work > flawlessly and we no longer get one-way audio. We are only using SIP > (3 trunks now, instead of 2) and having all 3 in use is not an issue. > > Problem: Make a call on a Polycom 320 IP phone to any number and (4/5 > times) it will drop the call after 30 seconds. I noticed that the > little timer that pops up on the LCD on the phone is missing when a > call will be dropped. This timer appears when the phone is answered, > so I have about 30 seconds to talk to them before the call is just > dropped. > > Known Causes: It's a NAT issue, I know that much, I just don't know > how to fix it. SIP debugging shows that it attempts to retransmit > packets to my phone and since it can't, it drops it after 30 seconds. > > Log snippet: > -- Executing [s at macro-dialout-trunk:19] Dial("SIP/203-b7a2b558", > "SIP/bw_outbound/+18005551212|300|") in new stack > Audio is at <public IP> port 11968 > Adding codec 0x4 (ulaw) to SDP > Adding codec 0x100 (g729) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (no NAT) to 216.82.224.202:5060: > INVITE sip:+18881231234 at 216.82.224.202 SIP/2.0 > Via: SIP/2.0/UDP <public IP>:5060;branch=z9hG4bK6ea30a1a;rport > From: "8881231234" <sip:+18881231234 at public IP>;tag=as3ed791f3 > To: <sip:+18005551212 at 216.82.224.202> > Contact: <sip:+18881231234 at public IP> > Call-ID: 28aa1a24047e1bdc3328f645766ddbbb at public IP > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Fri, 07 Nov 2008 19:06:30 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 291 > > v=0 > o=root 21520 21520 IN IP4 151.196.61.115 > s=session > c=IN IP4 <public IP> > t=0 0 > m=audio 11968 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > -- Called bw_outbound/+18885551212 > FreePBX*CLI> > <--- SIP read from 216.82.224.202:5060 ---> > SIP/2.0 100 Giving a try > Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060 > From: "8881231234" <sip:+14105601717 at public IP>;tag=as3ed791f3 > To: <sip:+18005551212 at 216.82.224.202> > Call-ID: 28aa1a24047e1bdc3328f645766ddbbb at public IP > CSeq: 102 INVITE > Server: Bandwidth.com TRM (bw7.gold.13) > Content-Length: 0 > > <-------------> > --- (8 headers 0 lines) --- > FreePBX*CLI> > <--- SIP read from 216.82.224.202:5060 ---> > SIP/2.0 183 Session Progress > Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060 > Record-Route: <sip:216.82.224.202;lr;ftag=as3ed791f3> > From: "8881231234" <sip:+18881231234 at public IP>;tag=as3ed791f3 > To: <sip:+18005551212 at 216.82.224.202>;tag=VPST50603522629853 > Call-ID: 28aa1a24047e1bdc3328f645766ddbbb at public IP > CSeq: 102 INVITE > Contact: <sip:+18005551212 at 209.247.16.221:5060;transport=udp> > Content-Type: application/sdp > Content-Length: 184 > > v=0 > o=- 1226084867 1226084868 IN IP4 209.244.42.253 > s=- > c=IN IP4 209.244.42.253 > t=0 0 > m=audio 64706 RTP/AVP 0 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > <-------------> > --- (10 headers 9 lines) --- > Found RTP audio format 0 > Found RTP audio format 101 > Peer audio RTP is at port 209.244.42.253:64706 > Found audio description format telephone-event for ID 101 > Got unsupported a:fmtp in SDP offer > Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 > (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 > (telephone-event), combined - 0x1 (telephone-event) > Peer audio RTP is at port 209.244.42.253:64706 > -- SIP/bw_outbound-08bf43d0 is making progress passing it to > SIP/203-b7a2b558 > Audio is at public IP port 16244 > Adding codec 0x4 (ulaw) to SDP > Adding codec 0x100 (g729) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > > <--- Transmitting (NAT) to 172.16.2.203:5060 ---> > SIP/2.0 183 Session Progress > Via: SIP/2.0/UDP > 172.16.2.203;branch=z9hG4bKed293df65EAFD78F;received=172.16.2.203 > From: "Me" <sip:203 at public IP>;tag=28354B-27A53F00 > To: <sip:18005551212 at public IP;user=phone>;tag=as600b952c > Call-ID: 992e82f4-2e300935-39c0ba22 at 172.16.2.203 > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:18005551212 at public IP> > Content-Type: application/sdp > Content-Length: 291 > > v=0 > o=root 21520 21520 IN IP4 public IP > s=session > c=IN IP4 public IP > t=0 0 > m=audio 16244 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > > <-------------> > --- (10 headers 9 lines) --- > Found RTP audio format 0 > Found RTP audio format 101 > Peer audio RTP is at port 209.244.42.253:64706 > Found audio description format telephone-event for ID 101 > Got unsupported a:fmtp in SDP offer > Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 > (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 > (telephone-event), combined - 0x1 (telephone-event) > Peer audio RTP is at port 209.244.42.253:64706 > list_route: hop: <sip:216.82.224.202;lr;ftag=as3ed791f3> > set_destination: Parsing <sip:216.82.224.202;lr;ftag=as3ed791f3> for > address/port to send to > set_destination: set destination to 216.82.224.202, port 5060 > Transmitting (no NAT) to 216.82.224.202:5060: > ACK sip:+18005551212 at 209.247.16.221:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK3c4f465e;rport > Route: <sip:216.82.224.202;lr;ftag=as3ed791f3> > From: "8881231234" <sip:+18881231234 at public IP>;tag=as3ed791f3 > To: <sip:+18005551212 at 216.82.224.202>;tag=VPST50603522629853 > Contact: <sip:+18881231234 at public IP> > Call-ID: 28aa1a24047e1bdc3328f645766ddbbb at public IP > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > > --- > -- SIP/bw_outbound-08bf43d0 answered SIP/203-b7a2b558 > Audio is at public IP port 16244 > Adding codec 0x4 (ulaw) to SDP > Adding codec 0x100 (g729) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > > <--- Reliably Transmitting (NAT) to 172.16.2.203:5060 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 172.16.2.203;branch=z9hG4bKed293df65EAFD78F;received=172.16.2.203 > From: "Me" <sip:203 at public IP>;tag=28354B-27A53F00 > To: <sip:18005551212 at public IP;user=phone>;tag=as600b952c > Call-ID: 992e82f4-2e300935-39c0ba22 at 172.16.2.203 > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:18005551212 at public IP> > Content-Type: application/sdp > Content-Length: 291 > > v=0 > o=root 21520 21521 IN IP4 public IP > s=session > c=IN IP4 public IP > t=0 0 > m=audio 16244 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > <------------> > -- Packet2Packet bridging SIP/203-b7a2b558 and SIP/bw_outbound-08bf43d0 > Retransmitting #1 (NAT) to 172.16.2.203:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 172.16.2.203;branch=z9hG4bKed293df65EAFD78F;received=172.16.2.203 > From: "Me" <sip:203 at public IP>;tag=28354B-27A53F00 > To: <sip:18005551212 at public IP;user=phone>;tag=as600b952c > Call-ID: 992e82f4-2e300935-39c0ba22 at 172.16.2.203 > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:18005551212 at public IP> > Content-Type: application/sdp > Content-Length: 291 > > v=0 > o=root 21520 21521 IN IP4 public IP > s=session > c=IN IP4 public IP > t=0 0 > m=audio 16244 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > <-------------> > --- (10 headers 0 lines) --- > Really destroying SIP dialog '48e1c5682586d0b92915c00a1104adc7 at public > IP' Method: OPTIONS > Retransmitting #2 (NAT) to 172.16.2.203:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 172.16.2.203;branch=z9hG4bKed293df65EAFD78F;received=172.16.2.203 > From: "Me" <sip:203 at public IP>;tag=28354B-27A53F00 > To: <sip:18005551212 at public IP;user=phone>;tag=as600b952c > Call-ID: 992e82f4-2e300935-39c0ba22 at 172.16.2.203 > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:18005551212 at public IP> > Content-Type: application/sdp > Content-Length: 291 > > v=0 > o=root 21520 21521 IN IP4 public IP > s=session > c=IN IP4 public IP > t=0 0 > m=audio 16244 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > --- > <snip> > <Then this and call is dropped> > --- > Scheduling destruction of SIP dialog > '28aa1a24047e1bdc3328f645766ddbbb at public IP' in 32000 ms (Method: > INVITE) > set_destination: Parsing <sip:216.82.224.202;lr;ftag=as3ed791f3> for > address/port to send to > set_destination: set destination to 216.82.224.202, port 5060 > Reliably Transmitting (no NAT) to 216.82.224.202:5060: > BYE sip:+18005551212 at 209.247.16.221:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK5e8d995d;rport > Route: <sip:216.82.224.202;lr;ftag=as3ed791f3> > From: "8881231234" <sip:+18881231234 at public IP>;tag=as3ed791f3 > To: <sip:+18005551212 at 216.82.224.202>;tag=VPST50603522629853 > Call-ID: 28aa1a24047e1bdc3328f645766ddbbb at public IP > CSeq: 103 BYE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > --------------END--------------- > > Doesn't matter if I have my SIP phone setting as nat=yes or nat=no. > > Settings: > > Outbound to Bandwidth.com: nat=no > Inbound from Bandwidth.com: nat=no > <Tried nat=yes on 1 and both. No effect> > > SIP Phone: > nat=yes <doesn't matter if it's nat=no> > qualify=yes > > SonicWall stuff I can't really post, but if you ask about it, I can > answer. Basically, we have an interface that's the phone system that > is assigned a static IP. SIP transformation is on and consistent NAT > is off under VOIP settings. The phone interface has VOIP/SIP allowed > and just blocks intrusion attempts. > > Question: Why does it sometimes work and sometimes not? This makes no > sense and it happens on all phones. Any suggestions? > > >We see this on occasion. It sounds a lot like Asterisk doing its usual routine of deciding that you can't POSSIBLY have a call going through because it can't receive an ACK response properly. Asterisk tries several times to send an ACK and get a response. If the remote system routes ACKs differently than it routes everything else, often times those ACKs get lost, and Asterisk assumes that the call can't be working, so it destroys it. ACK handling is a bit tricky in the real world, and we've run across countless incorrectly-configured SIP servers that don't handle it properly, so calls to them last just about exactly 30 seconds and then drop. There is, unfortunately, no way to turn off Asterisk's 'intelligent' behaviour in this scenario short of possibly patching the code. N.