[SR-Users] Freepbx 2.11.0rc1 with Asterisk 11.3.0 and Kamailio 4.0.1

Thu May 30 22:09:13 CEST 2013

Thank you so much for pointing me in the right direction! It was the
missing alias. Now the extensions are working, but there's one more
problem: When I call the PBX (echo-test) or from one extension to another,
I get a hangup because Asterisk doesn't seem to receive an ACK from the
client:

"Retransmission timeout reached on transmission
OGNkYTY1ZmJmY2VmMjQ2YmM4MWU1YWY0YjU3NjlhYjA for seqno 2 (Critical Response)"

I have attached the log for two local calls that hangup after about 30s and
the sequence for an incoming call from one of the trunks that works without
problems. So you can map all the IPs:

--------
198.23.139.21 (5060): Kamailio
198.23.139.21 (5080): Asterisk
--------
192.168.178.33: Home network local IP
188.105.112.187: Home network external IP
--------
94.75.247.45: Trunk
--------

NAT was set to "No - RFC3581" for these captures, but I've tried all other
possibilities (including nathelper / rtpproxy) without success. Do you (or
anybody else) have any idea where to look in order to solve this problem?

On Thu, May 30, 2013 at 10:36 AM, Barry Flanagan <barry at flanagan.ie> wrote:

> On 29 May 2013 19:23, Michael Leuker <michael at leuker.me> wrote:
>
>> Sure, here's the sequence for an inbound call via the "LPhone" trunk that
>> was supposed to go through to extension 1001. The extension was set to
>> "NAT" in the FreePBX settings. Just ask if you need more background.
>>
>>
> The Asterisk part looks fine. It is sending the call to
> 1001 at 198.23.139.21:5060 which I presume is your Kamailio instance.
>
> for some reason Kamailio is not recognising the user. Could it be that you
> do not have the ip 198.23.139.21 set up on Kamailio as an alias, or that
> the user 1001 is registering to a different domain?
>
> Kamailio would be looking in the location table for "username='1001' AND
> domain='198.23.139.21'"
>
> You should check the Kamailio logs for what is happening when Asterisk
> sends it the INVITE for 1001 at 198.23.139.21
>
> Hope this helps.
>
> -Barry
>
>
>> On Wed, May 29, 2013 at 6:14 PM, Barry Flanagan <barry at flanagan.ie>wrote:
>>
>>> On 29 May 2013 10:25, Michael Leuker <michael at leuker.me> wrote:
>>>
>>>> Thank you very much for sharing your insights, Barry! I am facing the
>>>> same problem that Trevor described:
>>>>
>>>> Things are working just fine on their own, but as soon as FreePBX comes
>>>> into play, calling extensions becomes impossible because of the different
>>>> tables used. Removing the password from FreePBX (and setting the Kamailio
>>>> IP in the ACL field) seems to mitigate the issue somewhat, but even though
>>>> the extension shows as registered in FreePBX, it always shows as busy:
>>>>
>>>> chan_sip.c:23237 handle_response_invite: Failed to authenticate on
>>>> INVITE to '"xxxxxxxx" <sip:xxxxxxxx at 198.23.139.21>;tag=as72a4117a'
>>>>     -- SIP/1001-00000006 is circuit-busy
>>>>
>>>>
>>> Can you do "sip set debug on" on Asterisk and make a call and  post the
>>> output?
>>>
>>> -Barry
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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==========================================================================================
Incoming Call from Trunk
==========================================================================================

<--- SIP read from UDP:94.75.247.45:5060 --->
INVITE sip:s at 198.23.139.21:5080 SIP/2.0
Record-Route: <sip:94.75.247.45;lr=on;ftag=6pg6jZ1Q6Ky7c>
Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.0
Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.0
Max-Forwards: 12
From: <sip:4940306988122 at 66.217.168.172>;tag=6pg6jZ1Q6Ky7c
To: <sip:4158000777 at oak2-td2-ha.starnetusa.net>
Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58
CSeq: 44635656 INVITE
Contact: <sip:95.211.119.240;did=778.faf4204>
User-Agent: PAETEC
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY
Supported: precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 253
X-SIPE: 62997

v=0
o=FreeSWITCH 1369924508 1369924509 IN IP4 66.217.168.172
s=FreeSWITCH
c=IN IP4 66.217.168.172
t=0 0
m=audio 18932 RTP/AVP 0 18 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (18 headers 11 lines) ---

Sending to 94.75.247.45:5060 (no NAT)
Sending to 94.75.247.45:5060 (no NAT)
Using INVITE request as basis request - 0c092aee-4405-1231-64a0-0030489f3d58
Found peer 'LPhone' for '4940306988122' from 94.75.247.45:5060
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found RTP audio format 13
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|silk8|silk12|silk16|silk24), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 66.217.168.172:18932
Peer doesn't provide video
Looking for s in from-pstn-toheader (domain 198.23.139.21)
list_route: route/path hop: <sip:94.75.247.45;lr=on;ftag=6pg6jZ1Q6Ky7c>

<--- Transmitting (no NAT) to 94.75.247.45:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.0;received=94.75.247.45
Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.0
Record-Route: <sip:94.75.247.45;lr=on;ftag=6pg6jZ1Q6Ky7c>
From: <sip:4940306988122 at 66.217.168.172>;tag=6pg6jZ1Q6Ky7c
To: <sip:4158000777 at oak2-td2-ha.starnetusa.net>
Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58
CSeq: 44635656 INVITE
Server: FPBX-2.11.0(1.8)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:s at 198.23.139.21:5080>
Content-Length: 0

<------------>

Reliably Transmitting (no NAT) to 198.23.139.21:5060:
INVITE sip:1001 at 198.23.139.21:5060 SIP/2.0
Via: SIP/2.0/UDP 198.23.139.21:5080;branch=z9hG4bK4f7baafe
Max-Forwards: 70
From: "4940306988122" <sip:4940306988122 at 198.23.139.21:5080>;tag=as373889aa
To: <sip:1001 at 198.23.139.21:5060>
Contact: <sip:4940306988122 at 198.23.139.21:5080>
Call-ID: 10a421e0545886cc09b6af4a00c599d9 at 198.23.139.21:5080
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(1.8)
Date: Thu, 30 May 2013 19:50:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 688

v=0
o=root 1520806685 1520806685 IN IP4 198.23.139.21
s=Asterisk PBX SVN-trunk-r389770
c=IN IP4 198.23.139.21
t=0 0
m=audio 11966 RTP/AVP 0 108 107 9 8 96 97 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:108 SILK/24000
a=fmtp:108 usedtx=0
a=fmtp:108 useinbandfec=1
a=rtpmap:107 SILK/16000
a=fmtp:107 maxaveragebitrate=30000
a=fmtp:107 usedtx=0
a=fmtp:107 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 SILK/8000
a=fmtp:96 maxaveragebitrate=15000
a=fmtp:96 usedtx=0
a=fmtp:96 useinbandfec=1
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called SIP/1001

<--- Transmitting (no NAT) to 94.75.247.45:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.0;received=94.75.247.45
Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.0
Record-Route: <sip:94.75.247.45;lr=on;ftag=6pg6jZ1Q6Ky7c>
From: <sip:4940306988122 at 66.217.168.172>;tag=6pg6jZ1Q6Ky7c
To: <sip:4158000777 at oak2-td2-ha.starnetusa.net>;tag=as199823a4
Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58
CSeq: 44635656 INVITE
Server: FPBX-2.11.0(1.8)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:s at 198.23.139.21:5080>
Content-Length: 0

<------------>

<--- SIP read from UDP:198.23.139.21:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 198.23.139.21:5080;branch=z9hG4bK4f7baafe
From: "4940306988122" <sip:4940306988122 at 198.23.139.21:5080>;tag=as373889aa
To: <sip:1001 at 198.23.139.21:5060>
Call-ID: 10a421e0545886cc09b6af4a00c599d9 at 198.23.139.21:5080
CSeq: 102 INVITE
Server: kamailio (4.0.1 (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:198.23.139.21:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 198.23.139.21:5080;branch=z9hG4bK4f7baafe
Record-Route: <sip:198.23.139.21;transport=tcp;lr;r2=on;ftag=as373889aa>
Record-Route: <sip:198.23.139.21;r2=on;lr=on;ftag=as373889aa>
Contact: <sip:1001 at 188.105.112.187:59986;transport=tcp>
To: <sip:1001 at 198.23.139.21:5060>;tag=d0272262
From: "4940306988122"<sip:4940306988122 at 198.23.139.21:5080>;tag=as373889aa
Call-ID: 10a421e0545886cc09b6af4a00c599d9 at 198.23.139.21:5080
CSeq: 102 INVITE
User-Agent: X-Lite release 4.5.2 stamp 70142
Allow-Events: hold, talk
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
list_route: route/path hop: <sip:198.23.139.21;r2=on;lr=on;ftag=as373889aa>
list_route: route/path hop: <sip:198.23.139.21;transport=tcp;lr;r2=on;ftag=as373889aa>
    -- SIP/1001-00000007 is ringing

<--- Transmitting (no NAT) to 94.75.247.45:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.0;received=94.75.247.45
Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.0
Record-Route: <sip:94.75.247.45;lr=on;ftag=6pg6jZ1Q6Ky7c>
From: <sip:4940306988122 at 66.217.168.172>;tag=6pg6jZ1Q6Ky7c
To: <sip:4158000777 at oak2-td2-ha.starnetusa.net>;tag=as199823a4
Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58
CSeq: 44635656 INVITE
Server: FPBX-2.11.0(1.8)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:s at 198.23.139.21:5080>
Remote-Party-ID: "Michael Leuker" <sip:1001 at 66.217.168.172>;party=called;privacy=off;screen=no
Content-Length: 0

<------------>

<--- SIP read from UDP:198.23.139.21:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 198.23.139.21:5080;branch=z9hG4bK4f7baafe
Record-Route: <sip:198.23.139.21;transport=tcp;r2=on;lr=on;ftag=as373889aa>
Record-Route: <sip:198.23.139.21;r2=on;lr=on;ftag=as373889aa>
Contact: <sip:1001 at 188.105.112.187:49862;transport=tcp>
From: "4940306988122" <sip:4940306988122 at 198.23.139.21:5080>;tag=as373889aa
Call-ID: 10a421e0545886cc09b6af4a00c599d9 at 198.23.139.21:5080
CSeq: 102 INVITE
To: <sip:1001 at 198.23.139.21:5060>;tag=A88EC20F1B7AAFA8A5F615621E3AD38D
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
list_route: route/path hop: <sip:198.23.139.21;r2=on;lr=on;ftag=as373889aa>
list_route: route/path hop: <sip:198.23.139.21;transport=tcp;r2=on;lr=on;ftag=as373889aa>
    -- SIP/1001-00000007 is ringing

<--- SIP read from UDP:198.23.139.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.23.139.21:5080;branch=z9hG4bK4f7baafe
Record-Route: <sip:198.23.139.21;transport=tcp;lr;r2=on;ftag=as373889aa>
Record-Route: <sip:198.23.139.21;r2=on;lr=on;ftag=as373889aa>
Contact: <sip:1001 at 188.105.112.187:59986;transport=tcp>
To: <sip:1001 at 198.23.139.21:5060>;tag=d0272262
From: "4940306988122"<sip:4940306988122 at 198.23.139.21:5080>;tag=as373889aa
Call-ID: 10a421e0545886cc09b6af4a00c599d9 at 198.23.139.21:5080
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces, eventlist
User-Agent: X-Lite release 4.5.2 stamp 70142
Content-Length: 246

v=0
o=- 13014417040298814 3 IN IP4 192.168.178.33
s=X-Lite 4 release 4.5.2 stamp 70142
c=IN IP4 188.105.112.187
t=0 0
m=audio 59026 RTP/AVP 0 9 8 97 101
a=rtpmap:97 ILBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (14 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 101
Found audio description format ILBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|ilbc|g722|silk8|silk16|silk24), peer - audio=(ulaw|alaw|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|ilbc|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 188.105.112.187:59026
list_route: route/path hop: <sip:198.23.139.21;r2=on;lr=on;ftag=as373889aa>
list_route: route/path hop: <sip:198.23.139.21;transport=tcp;lr;r2=on;ftag=as373889aa>
set_destination: Parsing <sip:198.23.139.21;r2=on;lr=on;ftag=as373889aa> for address/port to send to
set_destination: set destination to 198.23.139.21:5060
Transmitting (no NAT) to 198.23.139.21:5060:
ACK sip:1001 at 188.105.112.187:59986;transport=tcp SIP/2.0
Via: SIP/2.0/UDP 198.23.139.21:5080;branch=z9hG4bK248efa9e
Route: <sip:198.23.139.21;r2=on;lr=on;ftag=as373889aa>,<sip:198.23.139.21;transport=tcp;lr;r2=on;ftag=as373889aa>
Max-Forwards: 70
From: "4940306988122" <sip:4940306988122 at 198.23.139.21:5080>;tag=as373889aa
To: <sip:1001 at 198.23.139.21:5060>;tag=d0272262
Contact: <sip:4940306988122 at 198.23.139.21:5080>
Call-ID: 10a421e0545886cc09b6af4a00c599d9 at 198.23.139.21:5080
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(1.8)
Content-Length: 0

---
    -- SIP/1001-00000007 answered SIP/LPhone-00000006
Audio is at 11104
Adding codec 100003 (ulaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 94.75.247.45:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.0;received=94.75.247.45
Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.0
Record-Route: <sip:94.75.247.45;lr=on;ftag=6pg6jZ1Q6Ky7c>
From: <sip:4940306988122 at 66.217.168.172>;tag=6pg6jZ1Q6Ky7c
To: <sip:4158000777 at oak2-td2-ha.starnetusa.net>;tag=as199823a4
Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58
CSeq: 44635656 INVITE
Server: FPBX-2.11.0(1.8)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:s at 198.23.139.21:5080>
Remote-Party-ID: "Michael Leuker" <sip:1001 at 66.217.168.172>;party=called;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 320

v=0
o=root 833496020 833496020 IN IP4 198.23.139.21
s=Asterisk PBX SVN-trunk-r389770
c=IN IP4 198.23.139.21
t=0 0
m=audio 11104 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

<------------>
       > 0x7f7ce419e9e0 -- Probation passed - setting RTP source address to 188.105.112.187:59026
Retransmitting #1 (no NAT) to 94.75.247.45:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.0;received=94.75.247.45
Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.0
Record-Route: <sip:94.75.247.45;lr=on;ftag=6pg6jZ1Q6Ky7c>
From: <sip:4940306988122 at 66.217.168.172>;tag=6pg6jZ1Q6Ky7c
To: <sip:4158000777 at oak2-td2-ha.starnetusa.net>;tag=as199823a4
Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58
CSeq: 44635656 INVITE
Server: FPBX-2.11.0(1.8)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:s at 198.23.139.21:5080>
Remote-Party-ID: "Michael Leuker" <sip:1001 at 66.217.168.172>;party=called;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 320

v=0
o=root 833496020 833496020 IN IP4 198.23.139.21
s=Asterisk PBX SVN-trunk-r389770
c=IN IP4 198.23.139.21
t=0 0
m=audio 11104 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

---
       > 0x7f7ca4026610 -- Probation passed - setting RTP source address to 66.217.168.172:18932

<--- SIP read from UDP:94.75.247.45:5060 --->
ACK sip:s at 198.23.139.21:5080 SIP/2.0
Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.2
Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.2
Max-Forwards: 12
From: <sip:4940306988122 at 66.217.168.172>;tag=6pg6jZ1Q6Ky7c
To: <sip:4158000777 at oak2-td2-ha.starnetusa.net>;tag=as199823a4
Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58
CSeq: 44635656 ACK
Contact: <sip:95.211.119.240;did=778.faf4204>
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:94.75.247.45:5060 --->
ACK sip:s at 198.23.139.21:5080 SIP/2.0
Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.2
Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.2
Max-Forwards: 12
From: <sip:4940306988122 at 66.217.168.172>;tag=6pg6jZ1Q6Ky7c
To: <sip:4158000777 at oak2-td2-ha.starnetusa.net>;tag=as199823a4
Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58
CSeq: 44635656 ACK
Contact: <sip:95.211.119.240;did=778.faf4204>
Content-Length: 0
-------------- next part --------------
==========================================================================================
Call to the Asterisk Echo-Test
==========================================================================================
Retransmitting #9 (no NAT) to 198.23.139.21:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.23.139.21;branch=z9hG4bKab14.deb15512.0;i=1;received=198.23.139.21
Via: SIP/2.0/TCP 192.168.178.33:21012;received=188.105.112.187;branch=z9hG4bK-d8754z-5a5e0016ef18b908-1---d8754z-;rport=59986
Record-Route: <sip:198.23.139.21;r2=on;lr=on;ftag=9861a241>
Record-Route: <sip:198.23.139.21;transport=tcp;r2=on;lr=on;ftag=9861a241>
From: "Michael Leuker"<sip:1001 at leuker.me>;tag=9861a241
To: <sip:*43 at leuker.me>;tag=as1cde49be
Call-ID: OGNkYTY1ZmJmY2VmMjQ2YmM4MWU1YWY0YjU3NjlhYjA
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*43 at 198.23.139.21:5080>
Content-Type: application/sdp
Content-Length: 366

v=0
o=root 204152970 204152970 IN IP4 198.23.139.21
s=Asterisk PBX SVN-trunk-r389770
c=IN IP4 198.23.139.21
t=0 0
m=audio 11026 RTP/AVP 9 8 0 98 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

==========================================================================================
Call from Ext. 1001 to 2001
==========================================================================================

Retransmitting #5 (no NAT) to 198.23.139.21:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.23.139.21;branch=z9hG4bKfa9.427069e6.0;i=1;received=198.23.139.21
Via: SIP/2.0/TCP 192.168.178.33:21012;received=188.105.112.187;branch=z9hG4bK-d8754z-6e073579cb9fff22-1---d8754z-;rport=59986
Record-Route: <sip:198.23.139.21;r2=on;lr=on;ftag=4f54073e>
Record-Route: <sip:198.23.139.21;transport=tcp;r2=on;lr=on;ftag=4f54073e>
From: "Michael Leuker"<sip:1001 at leuker.me>;tag=4f54073e
To: <sip:2001 at leuker.me>;tag=as592c404a
Call-ID: ZTQ4YzE4NTI5NTEyMTAzNzhmZWI5OTY3ZTM2ZGI5Mjc
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2001 at 198.23.139.21:5080>
Content-Type: application/sdp
Content-Length: 368

v=0
o=root 1998415620 1998415620 IN IP4 198.23.139.21
s=Asterisk PBX SVN-trunk-r389770
c=IN IP4 198.23.139.21
t=0 0
m=audio 10968 RTP/AVP 9 8 0 98 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv