Lost ACK packet?

Asterisk Asterisk Support
1

Have had an issue with dropped calls via my sip trunking provider (bandwidth.com). They claim to not be receiving ACK packets on certain calls from our Asterisk PBX. Doing a sip trace on a call that was dropped, I see these packets being sent:

U 2010/03/25 09:21:58.245832 172.27.7.6:5060 -> 216.82.224.202:5060
ACK sip:+19195559999@67.16.111.140:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 64.x.x.x:5060;branch=z9hG4bK79b1e727;rport.
Route: sip:216.82.224.202;lr;ftag=as1c8513df.
Max-Forwards: 70.
From: “Jeff Strope” sip:9191112222@64.x.x.x;tag=as1c8513df.
To: sip:+19195559999@216.82.224.202;tag=SDtcksd99-gK0abb9dc4.
Contact: sip:9191112222@64.x.x.x.
Call-ID: 2eb2fd962dd88b3a70cb11fe3094df32@64.x.x.x.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.6.1.6.
Content-Length: 0.
.

In the above, 172.27.7.6 is the PBX’s private address and 64.x.x.x is its public address.

This problem occurs only intermittently, not on every call. Any ideas? Any other information I should provide on this?

2

If they are not receiving the ACK, they should be resending the OK. How many times do they resend the OK?

3

Looks like there are 7 incoming 200 OK’s following the lost outgoing ACK. Here’s the complete sip trace on the bandwidth.com leg:

U 2010/03/25 09:17:56.376930 172.27.7.6:5060 -> 216.82.224.202:5060
INVITE sip:+19195559999@216.82.224.202 SIP/2.0.
Via: SIP/2.0/UDP 64.X.X.X:5060;branch=z9hG4bK0622d778;rport.
Max-Forwards: 70.
From: “Jeff S” sip:9191112222@64.X.X.X;tag=as5b0b91f5.
To: sip:+19195559999@216.82.224.202.
Contact: sip:9191112222@64.X.X.X.
Call-ID: 2178bba50cf338696d0f854626e668f0@64.X.X.X.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 1.6.1.6.
Date: Thu, 25 Mar 2010 13:17:56 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 310.
.
v=0.
o=root 1635480164 1635480164 IN IP4 64.X.X.X.
s=Asterisk PBX 1.6.1.6.
c=IN IP4 64.X.X.X.
t=0 0.
m=audio 12728 RTP/AVP 0 3 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

U 2010/03/25 09:17:56.392022 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 64.X.X.X:5060;branch=z9hG4bK0622d778;rport=5060.
From: “Jeff S” sip:9191112222@64.X.X.X;tag=as5b0b91f5.
To: sip:+19195559999@216.82.224.202.
Call-ID: 2178bba50cf338696d0f854626e668f0@64.X.X.X.
CSeq: 102 INVITE.
Server: Bandwidth.com TRM (bw7.gold.13).
Content-Length: 0.
.

U 2010/03/25 09:17:58.664468 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 64.X.X.X:5060;received=64.X.X.X;branch=z9hG4bK0622d778;rport=5060.
From: “Jeff S” sip:9191112222@64.X.X.X;tag=as5b0b91f5.
To: sip:+19195559999@216.82.224.202;tag=SD6epbf99-gK0ba460cb.
Call-ID: 2178bba50cf338696d0f854626e668f0@64.X.X.X.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as5b0b91f5.
Contact: sip:+19195559999@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Content-Length: 233.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 32239 6271 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 55522 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/03/25 09:18:05.478582 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.X.X.X:5060;received=64.X.X.X;branch=z9hG4bK0622d778;rport=5060.
From: “Jeff S” sip:9191112222@64.X.X.X;tag=as5b0b91f5.
To: sip:+19195559999@216.82.224.202;tag=SD6epbf99-gK0ba460cb.
Call-ID: 2178bba50cf338696d0f854626e668f0@64.X.X.X.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as5b0b91f5.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19195559999@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 233.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 32239 6271 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 55522 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/03/25 09:18:05.478845 172.27.7.6:5060 -> 216.82.224.202:5060
ACK sip:+19195559999@67.16.122.4:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 64.X.X.X:5060;branch=z9hG4bK08f9ec88;rport.
Route: sip:216.82.224.202;lr;ftag=as5b0b91f5.
Max-Forwards: 70.
From: “Jeff S” sip:9191112222@64.X.X.X;tag=as5b0b91f5.
To: sip:+19195559999@216.82.224.202;tag=SD6epbf99-gK0ba460cb.
Contact: sip:9191112222@64.X.X.X.
Call-ID: 2178bba50cf338696d0f854626e668f0@64.X.X.X.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.6.1.6.
Content-Length: 0.
.

U 2010/03/25 09:18:05.988742 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.X.X.X:5060;received=64.X.X.X;branch=z9hG4bK0622d778;rport=5060.
From: “Jeff S” sip:9191112222@64.X.X.X;tag=as5b0b91f5.
To: sip:+19195559999@216.82.224.202;tag=SD6epbf99-gK0ba460cb.
Call-ID: 2178bba50cf338696d0f854626e668f0@64.X.X.X.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as5b0b91f5.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19195559999@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 233.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 32239 6271 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 55522 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/03/25 09:18:06.967591 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.X.X.X:5060;received=64.X.X.X;branch=z9hG4bK0622d778;rport=5060.
From: “Jeff S” sip:9191112222@64.X.X.X;tag=as5b0b91f5.
To: sip:+19195559999@216.82.224.202;tag=SD6epbf99-gK0ba460cb.
Call-ID: 2178bba50cf338696d0f854626e668f0@64.X.X.X.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as5b0b91f5.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19195559999@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 233.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 32239 6271 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 55522 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/03/25 09:18:08.969286 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.X.X.X:5060;received=64.X.X.X;branch=z9hG4bK0622d778;rport=5060.
From: “Jeff S” sip:9191112222@64.X.X.X;tag=as5b0b91f5.
To: sip:+19195559999@216.82.224.202;tag=SD6epbf99-gK0ba460cb.
Call-ID: 2178bba50cf338696d0f854626e668f0@64.X.X.X.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as5b0b91f5.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19195559999@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 233.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 32239 6271 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 55522 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/03/25 09:18:12.967111 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.X.X.X:5060;received=64.X.X.X;branch=z9hG4bK0622d778;rport=5060.
From: “Jeff S” sip:9191112222@64.X.X.X;tag=as5b0b91f5.
To: sip:+19195559999@216.82.224.202;tag=SD6epbf99-gK0ba460cb.
Call-ID: 2178bba50cf338696d0f854626e668f0@64.X.X.X.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as5b0b91f5.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19195559999@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 233.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 32239 6271 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 55522 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/03/25 09:18:16.966199 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.X.X.X:5060;received=64.X.X.X;branch=z9hG4bK0622d778;rport=5060.
From: “Jeff S” sip:9191112222@64.X.X.X;tag=as5b0b91f5.
To: sip:+19195559999@216.82.224.202;tag=SD6epbf99-gK0ba460cb.
Call-ID: 2178bba50cf338696d0f854626e668f0@64.X.X.X.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as5b0b91f5.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19195559999@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 233.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 32239 6271 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 55522 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/03/25 09:18:20.966558 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.X.X.X:5060;received=64.X.X.X;branch=z9hG4bK0622d778;rport=5060.
From: “Jeff S” sip:9191112222@64.X.X.X;tag=as5b0b91f5.
To: sip:+19195559999@216.82.224.202;tag=SD6epbf99-gK0ba460cb.
Call-ID: 2178bba50cf338696d0f854626e668f0@64.X.X.X.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as5b0b91f5.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19195559999@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 233.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 32239 6271 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 55522 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/03/25 09:18:24.967385 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.X.X.X:5060;received=64.X.X.X;branch=z9hG4bK0622d778;rport=5060.
From: “Jeff S” sip:9191112222@64.X.X.X;tag=as5b0b91f5.
To: sip:+19195559999@216.82.224.202;tag=SD6epbf99-gK0ba460cb.
Call-ID: 2178bba50cf338696d0f854626e668f0@64.X.X.X.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as5b0b91f5.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19195559999@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 233.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 32239 6271 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 55522 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/03/25 09:18:28.963767 216.82.224.202:5060 -> 172.27.7.6:5060
BYE sip:9191112222@64.X.X.X SIP/2.0.
Record-Route: sip:216.82.224.202;lr;ftag=SD6epbf99-gK0ba460cb.
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKc552.af5564c3.0.
Via: SIP/2.0/UDP 67.16.122.4:5060;branch=z9hG4bKspsgf6300glhdig3p781.1.
From: sip:+19195559999@216.82.224.202;tag=SD6epbf99-gK0ba460cb.
To: “Jeff S” sip:9191112222@64.X.X.X;tag=as5b0b91f5.
Call-ID: 2178bba50cf338696d0f854626e668f0@64.X.X.X.
CSeq: 4526 BYE.
Max-Forwards: 68.
Content-Length: 0.
.

U 2010/03/25 09:18:28.963957 172.27.7.6:5060 -> 216.82.224.202:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKc552.af5564c3.0;received=216.82.224.202.
Via: SIP/2.0/UDP 67.16.122.4:5060;branch=z9hG4bKspsgf6300glhdig3p781.1.
Record-Route: sip:216.82.224.202;lr;ftag=SD6epbf99-gK0ba460cb.
From: sip:+19195559999@216.82.224.202;tag=SD6epbf99-gK0ba460cb.
To: “Jeff S” sip:9191112222@64.X.X.X;tag=as5b0b91f5.
Call-ID: 2178bba50cf338696d0f854626e668f0@64.X.X.X.
CSeq: 4526 BYE.
Server: Asterisk PBX 1.6.1.6.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Length: 0.
.

4

Asterisk should be re-sending the ACK. You need to follow the full SIP bug guidelines and run sip set debug on , core set verbose 10, core set debug 10, and also obtain a sip history for the call.

Hopefully, the debug output will tell you why Asterisk is ignoring the repeat OKs. It should ignore their content but still re-transmit the ACK.

If this doesn’t show a good reason, and I can’t see an obvious one, you need to check whether this is fixed in a later 1.6.1.x and, if not, report it on issues.asterisk.org, preferably using a trace from the latest 1.6.1.x.

I half remember a bug a bit like this being fixed recently.

5

David,

Thanks so much for the detailed reply. I’ll run some tests and try to capture more info when these calls drop.

One thing this might be related to (or not) – we see lots of 401 Unauthorized messages from Asterisk to our Aastra 9480i phones on both REGISTER and INVITE messages. For instance, when an extension registers, it always looks like this (REGISTER - 401 - REGISTER - OK):

<— SIP read from UDP://192.168.3.7:5060 —>
REGISTER sip:172.27.7.6:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.7:5060;branch=z9hG4bKe2f92f905db873a73.8964420c182b53b4c
Max-Forwards: 70
From: sip:203@172.27.7.6:5060;tag=dd81240932
To: sip:203@172.27.7.6:5060
Call-ID: ec9649e8ca835333
CSeq: 3399 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username=“203”,realm=“asterisk”,nonce=“519d8ca0”,uri=“sip:172.27.7.6:5060”,response=" 521795056abd83bac6f2927ebc7a6b8e",algorithm=MD5
Contact: “Jeff S” sip:203@192.168.3.7:5060;transport=udp;+sip.instance="<urn:uuid:00000000-0000-1000-8000- 00085D21D8E3>"
Supported: gruu, path
User-Agent: Aastra 9480i/2.5.3.18
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 192.168.3.7 : 5060 (no NAT)

<— Transmitting (NAT) to 192.168.3.7:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.3.7:5060;branch=z9hG4bKe2f92f905db873a73.8964420c182b53b4c;received=192.168.3.7
From: sip:203@172.27.7.6:5060;tag=dd81240932
To: sip:203@172.27.7.6:5060;tag=as0e600f6d
Call-ID: ec9649e8ca835333
CSeq: 3399 REGISTER
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="07b965a9"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ec9649e8ca835333’ in 32000 ms (Method: REGISTER)
social*CLI>
<— SIP read from UDP://192.168.3.7:5060 —>
REGISTER sip:172.27.7.6:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.7:5060;branch=z9hG4bK119f9840f133fbb93.416d260ac7ddce7e2
Max-Forwards: 70
From: sip:203@172.27.7.6:5060;tag=dd81240932
To: sip:203@172.27.7.6:5060
Call-ID: ec9649e8ca835333
CSeq: 3400 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username=“203”,realm=“asterisk”,nonce=“07b965a9”,uri=“sip:172.27.7.6:5060”,response=" ecc4f3ebd826717e54347cb9c996ac03",algorithm=MD5
Contact: “Jeff S” sip:203@192.168.3.7:5060;transport=udp;+sip.instance="<urn:uuid:00000000-0000-1000-8000- 00085D21D8E3>"
Supported: gruu, path
User-Agent: Aastra 9480i/2.5.3.18
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 192.168.3.7 : 5060 (NAT)
social*CLI>
<— Transmitting (NAT) to 192.168.3.7:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.7:5060;branch=z9hG4bK119f9840f133fbb93.416d260ac7ddce7e2;received=192.168.3.7
From: sip:203@172.27.7.6:5060;tag=dd81240932
To: sip:203@172.27.7.6:5060;tag=as0e600f6d
Call-ID: ec9649e8ca835333
CSeq: 3400 REGISTER
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 120
Contact: sip:203@192.168.3.7:5060;transport=udp;expires=120
Date: Thu, 25 Mar 2010 13:57:13 GMT
Content-Length: 0

INVITES do the same thing (401 the first time, then allowed). Is this normal and possibly related?

6

It is fairly normal to first try with an unauthorised request. You need the response to know what authentication options exist.

7

Upgrading to 1.6.1.18 did not fix this issue – had a similar situation this morning (sip conversation below). Any insight into why Asterisk would not send another ACK after the repeated 200 OKs from bandwidth.com? Here is my config for bandwidth.com from sip.conf (does canreinvite come into play here?):

[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
context=frombandwidth
nat=no

[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
nat=no

Still trying to capture a full debug of this. The issue is very sporadic and makes capturing a debug difficult. Will keep trying.

U 2010/04/12 08:52:14.150329 172.27.7.6:5060 -> 216.82.224.202:5060
INVITE sip:+19199555555@216.82.224.202:5060 SIP/2.0.
Via: SIP/2.0/UDP 64.x.x.x:5060;branch=z9hG4bK0686a323;rport.
Max-Forwards: 70.
From: “Jeff Strope” sip:9193136203@64.x.x.x;tag=as46bf71e0.
To: sip:+19199555555@216.82.224.202:5060.
Contact: sip:9193136203@64.x.x.x.
Call-ID: 26780e6e3b88d7000fe0f1ea2e34f393@64.x.x.x.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 1.6.1.18.
Date: Mon, 12 Apr 2010 12:52:14 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 309.
.
v=0.
o=root 470844103 470844103 IN IP4 64.x.x.x.
s=Asterisk PBX 1.6.1.18.
c=IN IP4 64.x.x.x.
t=0 0.
m=audio 19542 RTP/AVP 0 3 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

U 2010/04/12 08:52:14.166263 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 64.x.x.x:5060;branch=z9hG4bK0686a323;rport=5060.
From: “Jeff Strope” sip:9193136203@64.x.x.x;tag=as46bf71e0.
To: sip:+19199555555@216.82.224.202:5060.
Call-ID: 26780e6e3b88d7000fe0f1ea2e34f393@64.x.x.x.
CSeq: 102 INVITE.
Server: Bandwidth.com TRM (bw7.gold.13).
Content-Length: 0.
.

U 2010/04/12 08:52:16.082958 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 64.x.x.x:5060;received=64.x.x.x;branch=z9hG4bK0686a323;rport=5060.
From: “Jeff Strope” sip:9193136203@64.x.x.x;tag=as46bf71e0.
To: sip:+19199555555@216.82.224.202:5060;tag=SDg746f99-gK0df86eb3.
Call-ID: 26780e6e3b88d7000fe0f1ea2e34f393@64.x.x.x.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as46bf71e0.
Contact: sip:+19199555555@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Content-Length: 234.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 24676 28349 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 52972 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/04/12 08:52:22.323429 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.x.x.x:5060;received=64.x.x.x;branch=z9hG4bK0686a323;rport=5060.
From: “Jeff Strope” sip:9193136203@64.x.x.x;tag=as46bf71e0.
To: sip:+19199555555@216.82.224.202:5060;tag=SDg746f99-gK0df86eb3.
Call-ID: 26780e6e3b88d7000fe0f1ea2e34f393@64.x.x.x.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as46bf71e0.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19199555555@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 234.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 24676 28349 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 52972 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/04/12 08:52:22.323740 172.27.7.6:5060 -> 216.82.224.202:5060
ACK sip:+19199555555@67.16.122.4:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 64.x.x.x:5060;branch=z9hG4bK6097d48b;rport.
Route: sip:216.82.224.202;lr;ftag=as46bf71e0.
Max-Forwards: 70.
From: “Jeff Strope” sip:9193136203@64.x.x.x;tag=as46bf71e0.
To: sip:+19199555555@216.82.224.202:5060;tag=SDg746f99-gK0df86eb3.
Contact: sip:9193136203@64.x.x.x.
Call-ID: 26780e6e3b88d7000fe0f1ea2e34f393@64.x.x.x.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.6.1.18.
Content-Length: 0.
.

U 2010/04/12 08:52:22.777129 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.x.x.x:5060;received=64.x.x.x;branch=z9hG4bK0686a323;rport=5060.
From: “Jeff Strope” sip:9193136203@64.x.x.x;tag=as46bf71e0.
To: sip:+19199555555@216.82.224.202:5060;tag=SDg746f99-gK0df86eb3.
Call-ID: 26780e6e3b88d7000fe0f1ea2e34f393@64.x.x.x.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as46bf71e0.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19199555555@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 234.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 24676 28349 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 52972 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/04/12 08:52:23.776464 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.x.x.x:5060;received=64.x.x.x;branch=z9hG4bK0686a323;rport=5060.
From: “Jeff Strope” sip:9193136203@64.x.x.x;tag=as46bf71e0.
To: sip:+19199555555@216.82.224.202:5060;tag=SDg746f99-gK0df86eb3.
Call-ID: 26780e6e3b88d7000fe0f1ea2e34f393@64.x.x.x.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as46bf71e0.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19199555555@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 234.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 24676 28349 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 52972 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/04/12 08:52:25.777131 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.x.x.x:5060;received=64.x.x.x;branch=z9hG4bK0686a323;rport=5060.
From: “Jeff Strope” sip:9193136203@64.x.x.x;tag=as46bf71e0.
To: sip:+19199555555@216.82.224.202:5060;tag=SDg746f99-gK0df86eb3.
Call-ID: 26780e6e3b88d7000fe0f1ea2e34f393@64.x.x.x.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as46bf71e0.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19199555555@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 234.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 24676 28349 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 52972 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/04/12 08:52:29.776470 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.x.x.x:5060;received=64.x.x.x;branch=z9hG4bK0686a323;rport=5060.
From: “Jeff Strope” sip:9193136203@64.x.x.x;tag=as46bf71e0.
To: sip:+19199555555@216.82.224.202:5060;tag=SDg746f99-gK0df86eb3.
Call-ID: 26780e6e3b88d7000fe0f1ea2e34f393@64.x.x.x.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as46bf71e0.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19199555555@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 234.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 24676 28349 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 52972 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/04/12 08:52:33.776563 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.x.x.x:5060;received=64.x.x.x;branch=z9hG4bK0686a323;rport=5060.
From: “Jeff Strope” sip:9193136203@64.x.x.x;tag=as46bf71e0.
To: sip:+19199555555@216.82.224.202:5060;tag=SDg746f99-gK0df86eb3.
Call-ID: 26780e6e3b88d7000fe0f1ea2e34f393@64.x.x.x.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as46bf71e0.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19199555555@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 234.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 24676 28349 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 52972 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/04/12 08:52:37.775902 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.x.x.x:5060;received=64.x.x.x;branch=z9hG4bK0686a323;rport=5060.
From: “Jeff Strope” sip:9193136203@64.x.x.x;tag=as46bf71e0.
To: sip:+19199555555@216.82.224.202:5060;tag=SDg746f99-gK0df86eb3.
Call-ID: 26780e6e3b88d7000fe0f1ea2e34f393@64.x.x.x.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as46bf71e0.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19199555555@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 234.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 24676 28349 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 52972 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/04/12 08:52:41.775740 216.82.224.202:5060 -> 172.27.7.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 64.x.x.x:5060;received=64.x.x.x;branch=z9hG4bK0686a323;rport=5060.
From: “Jeff Strope” sip:9193136203@64.x.x.x;tag=as46bf71e0.
To: sip:+19199555555@216.82.224.202:5060;tag=SDg746f99-gK0df86eb3.
Call-ID: 26780e6e3b88d7000fe0f1ea2e34f393@64.x.x.x.
CSeq: 102 INVITE.
Record-Route: sip:216.82.224.202;lr;ftag=as46bf71e0.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: sip:+19199555555@67.16.122.4:5060;transport=udp.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Supported: timer.
Session-Expires: 1800;refresher=uas.
Content-Length: 234.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 24676 28349 IN IP4 67.16.122.4.
s=SIP Media Capabilities.
c=IN IP4 67.16.122.4.
t=0 0.
m=audio 52972 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

U 2010/04/12 08:52:45.773119 216.82.224.202:5060 -> 172.27.7.6:5060
BYE sip:9193136203@64.x.x.x SIP/2.0.
Record-Route: sip:216.82.224.202;lr;ftag=SDg746f99-gK0df86eb3.
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK7e3c.831579c4.0.
Via: SIP/2.0/UDP 67.16.122.4:5060;branch=z9hG4bK6m5jko20eol0eg8f70o0.1.
From: sip:+19199555555@216.82.224.202:5060;tag=SDg746f99-gK0df86eb3.
To: “Jeff Strope” sip:9193136203@64.x.x.x;tag=as46bf71e0.
Call-ID: 26780e6e3b88d7000fe0f1ea2e34f393@64.x.x.x.
CSeq: 20642 BYE.
Max-Forwards: 68.
Content-Length: 0.
.

U 2010/04/12 08:52:45.773289 172.27.7.6:5060 -> 216.82.224.202:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK7e3c.831579c4.0;received=216.82.224.202.
Via: SIP/2.0/UDP 67.16.122.4:5060;branch=z9hG4bK6m5jko20eol0eg8f70o0.1.
Record-Route: sip:216.82.224.202;lr;ftag=SDg746f99-gK0df86eb3.
From: sip:+19199555555@216.82.224.202:5060;tag=SDg746f99-gK0df86eb3.
To: “Jeff Strope” sip:9193136203@64.x.x.x;tag=as46bf71e0.
Call-ID: 26780e6e3b88d7000fe0f1ea2e34f393@64.x.x.x.
CSeq: 20642 BYE.
Server: Asterisk PBX 1.6.1.18.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Length: 0.
.

8

Unless this is fixed in a later version, you need proper SIP debug output to progress it; traces from an external packat logger are not normally accepted. In particular, you need any output explaining why Asterisk thinks it should completely ignore the ACK.

9

I’m still trying to capture a full debug on this – will post and send to digium when I have one.

10

Any progress with this issue?
We are experiencing similar problem.
After some period of active use and making calls our PBX stops sending ACK after our provider has sent 183 ringing and 200 OK with SDP.
This causes provider SBC to resend the message again and again until the SBC gives up and the call fails.
Any ideas how this could be resolved?