[SR-Users] Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)

[Alex Villací­s Lasso]

Please consider the following SIP packet exchange, as seen by a tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a call to 201.234.196.170:

IP 198.58.101.75.5060 > 201.234.196.170.5060
INVITE sip:*43-p4WbfP+***@public.gmane.org:5060 SIP/2.0
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport
Max-Forwards: 70
From: "9002" <sip:9002-5nr+QlEBh/***@public.gmane.org>;tag=as0bc522a9
To: <sip:*43-p4WbfP+***@public.gmane.org:5060>
Contact: <sip:9002-5nr+QlEBh/***@public.gmane.org:5060>
Call-ID: 2c14c21f5052a74a78ca4ab736657b00-5nr+QlEBh/***@public.gmane.org:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Fri, 29 Aug 2014 18:23:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 299

v=0
o=root 521741684 521741684 IN IP4 198.58.101.75
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 198.58.101.75
t=0 0
m=audio 16426 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

IP 201.234.196.170.5060 > 198.58.101.75.5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060
From: "9002" <sip:9002-5nr+QlEBh/***@public.gmane.org>;tag=as0bc522a9
To: <sip:*43-p4WbfP+***@public.gmane.org:5060>
Call-ID: 2c14c21f5052a74a78ca4ab736657b00-5nr+QlEBh/***@public.gmane.org:5060
CSeq: 102 INVITE
Server: kamailio (4.1.5 (x86_64/linux))
Content-Length: 0


IP 201.234.196.170.5060 > 198.58.101.75.5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes>
Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes>
From: "9002" <sip:9002-5nr+QlEBh/***@public.gmane.org>;tag=as0bc522a9
To: <sip:*43-p4WbfP+***@public.gmane.org:5060>;tag=as2798a3b9
Call-ID: 2c14c21f5052a74a78ca4ab736657b00-5nr+QlEBh/***@public.gmane.org:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*43-savSHZN5Fh8qMp+***@public.gmane.org:5080;alias=127.0.0.1~5080~1>
Content-Type: application/sdp
Require: timer
Content-Length: 305

v=0
o=root 159029581 159029581 IN IP4 201.234.196.170
s=Asterisk PBX 11.12.0
c=IN IP4 201.234.196.170
t=0 0
m=audio 18446 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes


According to a strict interpretation of the SIP RFC, which address should the machine at 198.58.101.75 use to send the subsequent ACK? Which field(s) are to be used to extract said address? I am trying to understand an issue of a missing ACK between
201.234.196.17x and a different public IP, with the only difference that the other IP is not running Asterisk. For the exchange shown above, 201.234.196.170 receives an ACK, but I want to know whether the packets correctly indicate the address for the ACK,
or whether the Asterisk at 198.58.101.75 is compensating for a malformed packet.

[Andres]

Post by Alex Villací­s Lasso
Please consider the following SIP packet exchange, as seen by a
tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a
IP 198.58.101.75.5060 > 201.234.196.170.5060
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Fri, 29 Aug 2014 18:23:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 299
v=0
o=root 521741684 521741684 IN IP4 198.58.101.75
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 198.58.101.75
t=0 0
m=audio 16426 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
IP 201.234.196.170.5060 > 198.58.101.75.5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060
CSeq: 102 INVITE
Server: kamailio (4.1.5 (x86_64/linux))
Content-Length: 0
IP 201.234.196.170.5060 > 198.58.101.75.5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060
<sip:127.0.0.1;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes>
<sip:192.168.2.18;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes>
CSeq: 102 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Type: application/sdp
Require: timer
Content-Length: 305
v=0
o=root 159029581 159029581 IN IP4 201.234.196.170
s=Asterisk PBX 11.12.0
c=IN IP4 201.234.196.170
t=0 0
m=audio 18446 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes
According to a strict interpretation of the SIP RFC, which address
should the machine at 198.58.101.75 use to send the subsequent ACK?
Which field(s) are to be used to extract said address? I am trying to
understand an issue of a missing ACK between 201.234.196.17x and a
different public IP, with the only difference that the other IP is not
running Asterisk. For the exchange shown above, 201.234.196.170
receives an ACK, but I want to know whether the packets correctly
indicate the address for the ACK, or whether the Asterisk at
198.58.101.75 is compensating for a malformed packet.

Regardless of what the first hop of the ACK is going to be, the Contact
Field in the SIP 200 OK is telling 198.58.101.75 that the ACK should be
directed to 127.0.0.1 which is probably not what you want.

[Daniel-Constantin Mierla]

Post by Andres

Post by Alex Villací­s Lasso
Please consider the following SIP packet exchange, as seen by a
tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a
IP 198.58.101.75.5060 > 201.234.196.170.5060
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Fri, 29 Aug 2014 18:23:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 299
v=0
o=root 521741684 521741684 IN IP4 198.58.101.75
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 198.58.101.75
t=0 0
m=audio 16426 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
IP 201.234.196.170.5060 > 198.58.101.75.5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060
CSeq: 102 INVITE
Server: kamailio (4.1.5 (x86_64/linux))
Content-Length: 0
IP 201.234.196.170.5060 > 198.58.101.75.5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060
<sip:127.0.0.1;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes>
<sip:192.168.2.18;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes>
CSeq: 102 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Type: application/sdp
Require: timer
Content-Length: 305
v=0
o=root 159029581 159029581 IN IP4 201.234.196.170
s=Asterisk PBX 11.12.0
c=IN IP4 201.234.196.170
t=0 0
m=audio 18446 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes
According to a strict interpretation of the SIP RFC, which address
should the machine at 198.58.101.75 use to send the subsequent ACK?
Which field(s) are to be used to extract said address? I am trying to
understand an issue of a missing ACK between 201.234.196.17x and a
different public IP, with the only difference that the other IP is
not running Asterisk. For the exchange shown above, 201.234.196.170
receives an ACK, but I want to know whether the packets correctly
indicate the address for the ACK, or whether the Asterisk at
198.58.101.75 is compensating for a malformed packet.

Regardless of what the first hop of the ACK is going to be, the
Contact Field in the SIP 200 OK is telling 198.58.101.75 that the ACK
should be directed to 127.0.0.1 which is probably not what you want.

In this case, the contact in 200ok is the last hop of the ACK, because
the 200ok includes Record-Route headers. Therefore the caller has to
send the ACK to last Record-Route address in 200ok.

That is also private/non-routable address in internet and I expect is
not what it is desired, considering the other endpoints work with public
IP.

I guess the sip server is running on a natted system (e.g.,
amazon-cloud-like). You may want to add 'advertise' address to listen
core parameter in order to use public ip in signaling packets -- see
core cookbook for more on listen parameter.

Cheers,
Daniel

[Alex Villací­s Lasso]

Post by Alex Villací­s Lasso
IP 198.58.101.75.5060 > 201.234.196.170.5060
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Fri, 29 Aug 2014 18:23:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 299
v=0
o=root 521741684 521741684 IN IP4 198.58.101.75
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 198.58.101.75
t=0 0
m=audio 16426 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
IP 201.234.196.170.5060 > 198.58.101.75.5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060
CSeq: 102 INVITE
Server: kamailio (4.1.5 (x86_64/linux))
Content-Length: 0
IP 201.234.196.170.5060 > 198.58.101.75.5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes>
Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes>
CSeq: 102 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Type: application/sdp
Require: timer
Content-Length: 305
v=0
o=root 159029581 159029581 IN IP4 201.234.196.170
s=Asterisk PBX 11.12.0
c=IN IP4 201.234.196.170
t=0 0
m=audio 18446 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes
According to a strict interpretation of the SIP RFC, which address should the machine at 198.58.101.75 use to send the subsequent ACK? Which field(s) are to be used to extract said address? I am trying to understand an issue of a missing ACK between
201.234.196.17x and a different public IP, with the only difference that the other IP is not running Asterisk. For the exchange shown above, 201.234.196.170 receives an ACK, but I want to know whether the packets correctly indicate the address for the
ACK, or whether the Asterisk at 198.58.101.75 is compensating for a malformed packet.

Regardless of what the first hop of the ACK is going to be, the Contact Field in the SIP 200 OK is telling 198.58.101.75 that the ACK should be directed to 127.0.0.1 which is probably not what you want.

In this case, the contact in 200ok is the last hop of the ACK, because the 200ok includes Record-Route headers. Therefore the caller has to send the ACK to last Record-Route address in 200ok.
That is also private/non-routable address in internet and I expect is not what it is desired, considering the other endpoints work with public IP.
I guess the sip server is running on a natted system (e.g., amazon-cloud-like). You may want to add 'advertise' address to listen core parameter in order to use public ip in signaling packets -- see core cookbook for more on listen parameter.
Cheers,
Daniel

The test system is running inside a local network and has a local address of 192.168.2.18.

So, it is true that the remote asterisk is covering up for a mistake in my kamailio config?

[Alex Villací­s Lasso]

Post by Alex Villací­s Lasso

Post by Alex Villací­s Lasso
IP 198.58.101.75.5060 > 201.234.196.170.5060
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Fri, 29 Aug 2014 18:23:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 299
v=0
o=root 521741684 521741684 IN IP4 198.58.101.75
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 198.58.101.75
t=0 0
m=audio 16426 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
IP 201.234.196.170.5060 > 198.58.101.75.5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060
CSeq: 102 INVITE
Server: kamailio (4.1.5 (x86_64/linux))
Content-Length: 0
IP 201.234.196.170.5060 > 198.58.101.75.5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes>
Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes>
CSeq: 102 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Type: application/sdp
Require: timer
Content-Length: 305
v=0
o=root 159029581 159029581 IN IP4 201.234.196.170
s=Asterisk PBX 11.12.0
c=IN IP4 201.234.196.170
t=0 0
m=audio 18446 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes
According to a strict interpretation of the SIP RFC, which address should the machine at 198.58.101.75 use to send the subsequent ACK? Which field(s) are to be used to extract said address? I am trying to understand an issue of a missing ACK between
201.234.196.17x and a different public IP, with the only difference that the other IP is not running Asterisk. For the exchange shown above, 201.234.196.170 receives an ACK, but I want to know whether the packets correctly indicate the address for the
ACK, or whether the Asterisk at 198.58.101.75 is compensating for a malformed packet.

Regardless of what the first hop of the ACK is going to be, the Contact Field in the SIP 200 OK is telling 198.58.101.75 that the ACK should be directed to 127.0.0.1 which is probably not what you want.

In this case, the contact in 200ok is the last hop of the ACK, because the 200ok includes Record-Route headers. Therefore the caller has to send the ACK to last Record-Route address in 200ok.
That is also private/non-routable address in internet and I expect is not what it is desired, considering the other endpoints work with public IP.
I guess the sip server is running on a natted system (e.g., amazon-cloud-like). You may want to add 'advertise' address to listen core parameter in order to use public ip in signaling packets -- see core cookbook for more on listen parameter.
Cheers,
Daniel

The test system is running inside a local network and has a local address of 192.168.2.18.
So, it is true that the remote asterisk is covering up for a mistake in my kamailio config?
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Maybe I should explain my setup better.

The test setup I want to run is, in a way, twice natted. The asterisk instance runs in localhost, the innermost net. The asterisk is not supposed to get its SIP signaling from anyone but Kamailio. The /etc/asterisk/sip.conf contains this:

[***@elx3 ~]# cat /etc/asterisk/sip.conf
[general]
context=default
allowoverlap=no
allowguest=no
realm=asterisk
srvlookup=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
relaxdtmf=yes
trustrpid=no
sendrpid=yes
sendrpid=pai
disallow=all
allow=ulaw
allow=alaw
allow=gsm
rtcachefriends=yes
callcounter=yes
alwaysauthreject=yes
faxdetect=yes
t38pt_udptl=yes
vmexten=*97
videosupport=yes
maxcallbitrate=384
nat=force_rport,comedia
directmedia=no
accept_outofcall_message=yes
auth_message_requests=yes

;The following settings restrict Asterisk to localhost for Kamailio integration
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
bindport=5080
outboundproxy=127.0.0.1
outboundproxyport=5060

#include sip_general_custom.conf
#include sip_register.conf
#include sip_custom.conf


The kamailio instance is the first instance of routing. It listens on all interfaces on port 5060 and routes packets from all the other interfaces to localhost and back. One of these interfaces is the local network (192.168.2.18), which routes to our gateway.

If kamailio is given a public interface, then our setup works correctly. The exchange in the first mail shows what happens when the packet is routed through our gateway (the second instance of routing, and an actual NAT). Our gateway is a linux system with
a kernel module (nf_nat_sip, nf_conntrack_sip) that rewrites the headers on the fly, resulting in the packet exchange as seen in the first mail. From what I have seen, the kernel modules rewrite To, From, but not Record-Route, where an instance of the
internal IP remains. If I understand correctly, the remote system tries to route to its own interpretation of 192.168.2.18, which fails.

If I add the advertised_address parameter and set it to the public IP, outgoing calls from asterisk to a registered SIP client break and get established with no audio (tested with Jitsi). I get the following exchange from 192.168.2.18:

INVITE sip:avillacis-***@public.gmane.org:5060;transport=udp;registering_acc=pbx_villacis_com SIP/2.0
Record-Route: <sip:201.234.196.170;r2=on;lr=on;ftag=as0551c44f>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as0551c44f>
Via: SIP/2.0/UDP 201.234.196.170;branch=z9hG4bKd5f3.6bce295bab666c7aceeddfebdc70c190.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK696d3bf6;rport=5080
Max-Forwards: 69
From: "Anonymous" <sip:anonymous-7sGlIlNGn3WhjG8JX1ygEWCgeU+***@public.gmane.org:5080>;tag=as0551c44f
To: <sip:avillacis-***@public.gmane.org:5060;transport=udp;registering_acc=pbx_villacis_com>
Contact: <sip:anonymous-savSHZN5Fh8qMp+***@public.gmane.org:5080>
Call-ID: 4f40ffcc123459313daf47397e18b0af-savSHZN5Fh8qMp+***@public.gmane.org:5080
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.12.0
Date: Mon, 01 Sep 2014 16:11:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 277
P-hint: outbound

v=0
o=root 1851320733 1851320733 IN IP4 127.0.0.1
s=Asterisk PBX 11.12.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 18624 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
SIP/2.0 180 Ringing
To: <sip:avillacis-***@public.gmane.org:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=d28a3a6e
Via: SIP/2.0/UDP 201.234.196.170;branch=z9hG4bKd5f3.6bce295bab666c7aceeddfebdc70c190.0;received=192.168.2.18,SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK696d3bf6;rport=5080
Record-Route: <sip:201.234.196.170;r2=on;lr=on;ftag=as0551c44f>,<sip:127.0.0.1;r2=on;lr=on;ftag=as0551c44f>
CSeq: 102 INVITE
Call-ID: 4f40ffcc123459313daf47397e18b0af-savSHZN5Fh8qMp+***@public.gmane.org:5080
From: "Anonymous" <sip:anonymous-7sGlIlNGn3WhjG8JX1ygEWCgeU+***@public.gmane.org:5080>;tag=as0551c44f
Contact: "avillacis" <sip:avillacis-***@public.gmane.org:5060;transport=udp;registering_acc=pbx_villacis_com>
User-Agent: Jitsi2.5.5255Linux
Content-Length: 0

SIP/2.0 200 OK
To: <sip:avillacis-***@public.gmane.org:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=d28a3a6e
Via: SIP/2.0/UDP 201.234.196.170;branch=z9hG4bKd5f3.6bce295bab666c7aceeddfebdc70c190.0;received=192.168.2.18,SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK696d3bf6;rport=5080
Record-Route: <sip:201.234.196.170;r2=on;lr=on;ftag=as0551c44f>,<sip:127.0.0.1;r2=on;lr=on;ftag=as0551c44f>
CSeq: 102 INVITE
Call-ID: 4f40ffcc123459313daf47397e18b0af-savSHZN5Fh8qMp+***@public.gmane.org:5080
From: "Anonymous" <sip:anonymous-7sGlIlNGn3WhjG8JX1ygEWCgeU+***@public.gmane.org:5080>;tag=as0551c44f
Contact: "avillacis" <sip:avillacis-***@public.gmane.org:5060;transport=udp;registering_acc=pbx_villacis_com>
User-Agent: Jitsi2.5.5255Linux
Content-Type: application/sdp
Content-Length: 217

v=0
o=avillacis-jitsi.org 0 0 IN IP4 192.168.3.2
s=-
c=IN IP4 192.168.3.2
t=0 0
m=audio 5006 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
ACK sip:avillacis-***@public.gmane.org:5060;transport=udp;registering_acc=pbx_villacis_com SIP/2.0
Via: SIP/2.0/UDP 201.234.196.170;branch=z9hG4bKd5f3.805fac86b0d1e9c1dc577d5ca12f12d3.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2d32b799;rport=5080
Max-Forwards: 69
From: "Anonymous" <sip:anonymous-7sGlIlNGn3WhjG8JX1ygEWCgeU+***@public.gmane.org:5080>;tag=as0551c44f
To: <sip:avillacis-***@public.gmane.org:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=d28a3a6e
Contact: <sip:anonymous-savSHZN5Fh8qMp+***@public.gmane.org:5080>
Call-ID: 4f40ffcc123459313daf47397e18b0af-savSHZN5Fh8qMp+***@public.gmane.org:5080
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.12.0
Content-Length: 0

At the same time, I get this on the kamailio log:

WARNING: rr [loose.c:830]: after_loose(): no socket found for match second RR

If I try the incoming call from the internet, while advertised_address is enabled, I get the following exchange. I also get the exact same log message, and one-way audio.

INVITE sip:*43-p4WbfP+***@public.gmane.org:5060 SIP/2.0
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK587b52bc;rport
Max-Forwards: 70
From: "9003" <sip:9003-5nr+QlEBh/***@public.gmane.org>;tag=as69ee0744
To: <sip:*43-p4WbfP+***@public.gmane.org:5060>
Contact: <sip:9003-5nr+QlEBh/***@public.gmane.org:5060>
Call-ID: 0398a11d3149031240ec2e70077a99fe-5nr+QlEBh/***@public.gmane.org:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Mon, 01 Sep 2014 16:46:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 301

v=0
o=root 1281221163 1281221163 IN IP4 198.58.101.75
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 198.58.101.75
t=0 0
m=audio 12958 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK587b52bc;rport=5060
From: "9003" <sip:9003-5nr+QlEBh/***@public.gmane.org>;tag=as69ee0744
To: <sip:*43-p4WbfP+***@public.gmane.org:5060>
Call-ID: 0398a11d3149031240ec2e70077a99fe-5nr+QlEBh/***@public.gmane.org:5060
CSeq: 102 INVITE
Server: kamailio (4.1.5 (x86_64/linux))
Content-Length: 0


SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK587b52bc;rport=5060
Record-Route: <sip:201.234.196.170;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->
Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->
From: "9003" <sip:9003-5nr+QlEBh/***@public.gmane.org>;tag=as69ee0744
To: <sip:*43-p4WbfP+***@public.gmane.org:5060>;tag=as5f3239b9
Call-ID: 0398a11d3149031240ec2e70077a99fe-5nr+QlEBh/***@public.gmane.org:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*43-savSHZN5Fh8qMp+***@public.gmane.org:5080>
Content-Type: application/sdp
Require: timer
Content-Length: 277

v=0
o=root 1757515753 1757515753 IN IP4 127.0.0.1
s=Asterisk PBX 11.12.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 16396 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

ACK sip:*43-savSHZN5Fh8qMp+***@public.gmane.org:5080 SIP/2.0
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK636e2948;rport
Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->,<sip:201.234.196.170;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->
Max-Forwards: 70
From: "9003" <sip:9003-5nr+QlEBh/***@public.gmane.org>;tag=as69ee0744
To: <sip:*43-p4WbfP+***@public.gmane.org:5060>;tag=as5f3239b9
Contact: <sip:9003-5nr+QlEBh/***@public.gmane.org:5060>
Call-ID: 0398a11d3149031240ec2e70077a99fe-5nr+QlEBh/***@public.gmane.org:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.20.0)
Content-Length: 0

How can I fix the "no socket found for match second RR" error?

[Daniel-Constantin Mierla]

If you get signling routed ok but no audio, then you have problems
bridging rtp stream.

Most probably you need to use rtpproxy (eventually with advertise
address (there is a patch or use second parameter for
rtpproxy_manage())) to bridge.

I never used sip-natting in kernel, so I am not aware how good it is. I
would rather do just simple port forwarding on the firewall and let
kamailio and rtpproxy to work with advertised address.

Then you can use:

listen=privateip advertise publicip

You will get rid of record routing log message about missing socket.
Otherwise, you should ignore it if the sip routing is ok.

Cheers,
Daniel

Post by Alex Villací­s Lasso
[...]
Maybe I should explain my setup better.
The test setup I want to run is, in a way, twice natted. The asterisk
instance runs in localhost, the innermost net. The asterisk is not
supposed to get its SIP signaling from anyone but Kamailio. The
[general]
context=default
allowoverlap=no
allowguest=no
realm=asterisk
srvlookup=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
relaxdtmf=yes
trustrpid=no
sendrpid=yes
sendrpid=pai
disallow=all
allow=ulaw
allow=alaw
allow=gsm
rtcachefriends=yes
callcounter=yes
alwaysauthreject=yes
faxdetect=yes
t38pt_udptl=yes
vmexten=*97
videosupport=yes
maxcallbitrate=384
nat=force_rport,comedia
directmedia=no
accept_outofcall_message=yes
auth_message_requests=yes
;The following settings restrict Asterisk to localhost for Kamailio integration
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
bindport=5080
outboundproxy=127.0.0.1
outboundproxyport=5060
#include sip_general_custom.conf
#include sip_register.conf
#include sip_custom.conf
The kamailio instance is the first instance of routing. It listens on
all interfaces on port 5060 and routes packets from all the other
interfaces to localhost and back. One of these interfaces is the local
network (192.168.2.18), which routes to our gateway.
If kamailio is given a public interface, then our setup works
correctly. The exchange in the first mail shows what happens when the
packet is routed through our gateway (the second instance of routing,
and an actual NAT). Our gateway is a linux system with a kernel module
(nf_nat_sip, nf_conntrack_sip) that rewrites the headers on the fly,
resulting in the packet exchange as seen in the first mail. From what
I have seen, the kernel modules rewrite To, From, but not
Record-Route, where an instance of the internal IP remains. If I
understand correctly, the remote system tries to route to its own
interpretation of 192.168.2.18, which fails.
If I add the advertised_address parameter and set it to the public IP,
outgoing calls from asterisk to a registered SIP client break and get
established with no audio (tested with Jitsi). I get the following
INVITE
SIP/2.0
Record-Route: <sip:201.234.196.170;r2=on;lr=on;ftag=as0551c44f>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as0551c44f>
Via: SIP/2.0/UDP
201.234.196.170;branch=z9hG4bKd5f3.6bce295bab666c7aceeddfebdc70c190.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK696d3bf6;rport=5080
Max-Forwards: 69
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.12.0
Date: Mon, 01 Sep 2014 16:11:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 277
P-hint: outbound
v=0
o=root 1851320733 1851320733 IN IP4 127.0.0.1
s=Asterisk PBX 11.12.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 18624 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
201.234.196.170;branch=z9hG4bKd5f3.6bce295bab666c7aceeddfebdc70c190.0;received=192.168.2.18,SIP/2.0/UDP
127.0.0.1:5080;branch=z9hG4bK696d3bf6;rport=5080
<sip:201.234.196.170;r2=on;lr=on;ftag=as0551c44f>,<sip:127.0.0.1;r2=on;lr=on;ftag=as0551c44f>
CSeq: 102 INVITE
Contact: "avillacis"
User-Agent: Jitsi2.5.5255Linux
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP
201.234.196.170;branch=z9hG4bKd5f3.6bce295bab666c7aceeddfebdc70c190.0;received=192.168.2.18,SIP/2.0/UDP
127.0.0.1:5080;branch=z9hG4bK696d3bf6;rport=5080
<sip:201.234.196.170;r2=on;lr=on;ftag=as0551c44f>,<sip:127.0.0.1;r2=on;lr=on;ftag=as0551c44f>
CSeq: 102 INVITE
Contact: "avillacis"
User-Agent: Jitsi2.5.5255Linux
Content-Type: application/sdp
Content-Length: 217
v=0
o=avillacis-jitsi.org 0 0 IN IP4 192.168.3.2
s=-
c=IN IP4 192.168.3.2
t=0 0
m=audio 5006 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
ACK
SIP/2.0
Via: SIP/2.0/UDP
201.234.196.170;branch=z9hG4bKd5f3.805fac86b0d1e9c1dc577d5ca12f12d3.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2d32b799;rport=5080
Max-Forwards: 69
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.12.0
Content-Length: 0
WARNING: rr [loose.c:830]: after_loose(): no socket found for match second RR
If I try the incoming call from the internet, while advertised_address
is enabled, I get the following exchange. I also get the exact same
log message, and one-way audio.
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK587b52bc;rport
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Mon, 01 Sep 2014 16:46:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 301
v=0
o=root 1281221163 1281221163 IN IP4 198.58.101.75
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 198.58.101.75
t=0 0
m=audio 12958 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK587b52bc;rport=5060
CSeq: 102 INVITE
Server: kamailio (4.1.5 (x86_64/linux))
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK587b52bc;rport=5060
<sip:201.234.196.170;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->
<sip:192.168.2.18;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->
CSeq: 102 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Type: application/sdp
Require: timer
Content-Length: 277
v=0
o=root 1757515753 1757515753 IN IP4 127.0.0.1
s=Asterisk PBX 11.12.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 16396 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK636e2948;rport
<sip:192.168.2.18;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->,<sip:201.234.196.170;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->
Max-Forwards: 70
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.20.0)
Content-Length: 0
How can I fix the "no socket found for match second RR" error?
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[Alex Villací­s Lasso]

If you get signling routed ok but no audio, then you have problems bridging rtp stream.
Most probably you need to use rtpproxy (eventually with advertise address (there is a patch or use second parameter for rtpproxy_manage())) to bridge.

I am already using rtpproxy in all my tests. All of my audio is routed correctly when I do not use advertised_address, without any log messages. However, in the original setup, when the non-Asterisk SIP trunk fails to route the ACK correctly, the Asterisk
in localhost drops the call after 10 to 30 seconds.

When I introduce advertised_address into the setup, with no other changes, I get the audio issues.

I never used sip-natting in kernel, so I am not aware how good it is. I would rather do just simple port forwarding on the firewall and let kamailio and rtpproxy to work with advertised address.

I will try to negotiate this.

listen=privateip advertise publicip

The private ip serves both phones in the same LAN as the Kamailio interface, and the gateway. Will the publicip advertising cause problems with the SIP clients in the LAN?

You will get rid of record routing log message about missing socket. Otherwise, you should ignore it if the sip routing is ok.
Cheers,
Daniel

Post by Alex Villací­s Lasso
[...]
Maybe I should explain my setup better.
[general]
context=default
allowoverlap=no
allowguest=no
realm=asterisk
srvlookup=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
relaxdtmf=yes
trustrpid=no
sendrpid=yes
sendrpid=pai
disallow=all
allow=ulaw
allow=alaw
allow=gsm
rtcachefriends=yes
callcounter=yes
alwaysauthreject=yes
faxdetect=yes
t38pt_udptl=yes
vmexten=*97
videosupport=yes
maxcallbitrate=384
nat=force_rport,comedia
directmedia=no
accept_outofcall_message=yes
auth_message_requests=yes
;The following settings restrict Asterisk to localhost for Kamailio integration
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
bindport=5080
outboundproxy=127.0.0.1
outboundproxyport=5060
#include sip_general_custom.conf
#include sip_register.conf
#include sip_custom.conf
The kamailio instance is the first instance of routing. It listens on all interfaces on port 5060 and routes packets from all the other interfaces to localhost and back. One of these interfaces is the local network (192.168.2.18), which routes to our
gateway.
If kamailio is given a public interface, then our setup works correctly. The exchange in the first mail shows what happens when the packet is routed through our gateway (the second instance of routing, and an actual NAT). Our gateway is a linux system
with a kernel module (nf_nat_sip, nf_conntrack_sip) that rewrites the headers on the fly, resulting in the packet exchange as seen in the first mail. From what I have seen, the kernel modules rewrite To, From, but not Record-Route, where an instance of
the internal IP remains. If I understand correctly, the remote system tries to route to its own interpretation of 192.168.2.18, which fails.
Record-Route: <sip:201.234.196.170;r2=on;lr=on;ftag=as0551c44f>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as0551c44f>
Via: SIP/2.0/UDP 201.234.196.170;branch=z9hG4bKd5f3.6bce295bab666c7aceeddfebdc70c190.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK696d3bf6;rport=5080
Max-Forwards: 69
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.12.0
Date: Mon, 01 Sep 2014 16:11:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 277
P-hint: outbound
v=0
o=root 1851320733 1851320733 IN IP4 127.0.0.1
s=Asterisk PBX 11.12.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 18624 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 201.234.196.170;branch=z9hG4bKd5f3.6bce295bab666c7aceeddfebdc70c190.0;received=192.168.2.18,SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK696d3bf6;rport=5080
Record-Route: <sip:201.234.196.170;r2=on;lr=on;ftag=as0551c44f>,<sip:127.0.0.1;r2=on;lr=on;ftag=as0551c44f>
CSeq: 102 INVITE
User-Agent: Jitsi2.5.5255Linux
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.234.196.170;branch=z9hG4bKd5f3.6bce295bab666c7aceeddfebdc70c190.0;received=192.168.2.18,SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK696d3bf6;rport=5080
Record-Route: <sip:201.234.196.170;r2=on;lr=on;ftag=as0551c44f>,<sip:127.0.0.1;r2=on;lr=on;ftag=as0551c44f>
CSeq: 102 INVITE
User-Agent: Jitsi2.5.5255Linux
Content-Type: application/sdp
Content-Length: 217
v=0
o=avillacis-jitsi.org 0 0 IN IP4 192.168.3.2
s=-
c=IN IP4 192.168.3.2
t=0 0
m=audio 5006 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
Via: SIP/2.0/UDP 201.234.196.170;branch=z9hG4bKd5f3.805fac86b0d1e9c1dc577d5ca12f12d3.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2d32b799;rport=5080
Max-Forwards: 69
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.12.0
Content-Length: 0
WARNING: rr [loose.c:830]: after_loose(): no socket found for match second RR
If I try the incoming call from the internet, while advertised_address is enabled, I get the following exchange. I also get the exact same log message, and one-way audio.
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK587b52bc;rport
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Mon, 01 Sep 2014 16:46:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 301
v=0
o=root 1281221163 1281221163 IN IP4 198.58.101.75
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 198.58.101.75
t=0 0
m=audio 12958 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK587b52bc;rport=5060
CSeq: 102 INVITE
Server: kamailio (4.1.5 (x86_64/linux))
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK587b52bc;rport=5060
Record-Route: <sip:201.234.196.170;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->
Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->
CSeq: 102 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Type: application/sdp
Require: timer
Content-Length: 277
v=0
o=root 1757515753 1757515753 IN IP4 127.0.0.1
s=Asterisk PBX 11.12.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 16396 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK636e2948;rport
Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->,<sip:201.234.196.170;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->
Max-Forwards: 70
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.20.0)
Content-Length: 0
How can I fix the "no socket found for match second RR" error?
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