No sound from / to incoming calls

system · February 4, 2011, 12:53am

Hello all,

I have, what I hope is a simple problem.

I have various soft phone extensions which all authenticate properly with an Ubuntu 10.10 box running Asterisk 1.8.

The following calls work correctly:

Extension to extension
Extension to external
DID to dialplan (Adhearsion)

The problem is that there is no sound from any incoming DID calls to any extension.

Does this ring a bell with anyone?

Is this a codec or NATing issue?

For what should I be looking in the logs?

Any help would be greatly appreciated.

sip.conf

[general]
context=incoming
format=gsm                

externhost=liffysbox.dyndns.org
externrefresh=600

localnet=192.168.1.0/255.255.255.0

nat=yes

udpbindaddr=0.0.0.0             

bindaddr=0.0.0.0

tcpbindaddr=0.0.0.0             
srvlookup=yes                   

[basic-options](!)                ; a template
        dtmfmode=rfc2833
        context=local
        type=friend
        host=dynamic

[natted-phone](!,basic-options)   ; another template inheriting basic-options
        nat=yes 		  ; firewall between client and server
        directmedia=no

[public-phone](!,basic-options)   ; another template inheriting basic-options
        nat=no			  ; no firewall between client and server
        directmedia=yes

[my-codecs](!)                    ; a template for my preferred codecs
        disallow=all
        allow=ilbc
        allow=g729
        allow=gsm
        allow=g723
        allow=ulaw

[ulaw-phone](!)                   ; and another one for ulaw-only
        disallow=all
        allow=ulaw


[100](basic-options)
dtmfmode=inband
relaxdtmf=yes
md5secret=abd81fb84a2f01a02ba644d9c170008a

[121](basic-options)
md5secret=153e1b0ab3d13dee9af60d0ca02a1302

[122](basic-options)
md5secret=1fe889629a57abf811944049a6968155

[123](basic-options)
md5secret=3ab72802f01f6d3b84ece8292958a676

[124](basic-options)
md5secret=a0b99ae5827aa0a03175b82955f00aa8

dejanst · February 4, 2011, 10:48am

You really complicated the sip.conf file. Why do you need interconnected templatest? This is the first time I see this kind of settings 8) .

For a debug do a “sip set debug on” in the Asterisk CLI and have a look at the SIP trace. You will find the answer there.

system · February 9, 2011, 11:59pm

So after reviewing the debug output, I’m still not entirely sure what is wrong, other than a codec mismatch, and even then it seems as if they both share RTP audio format 0 (PCMU)?

[code]apollo*CLI> core show codecs audio
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INT BINARY HEX TYPE NAME DESCRIPTION

              1 (1 <<  0)                (0x1)  audio       g723   (G.723.1)
              2 (1 <<  1)                (0x2)  audio        gsm   (GSM)
              4 (1 <<  2)                (0x4)  audio       ulaw   (G.711 u-law)
              8 (1 <<  3)                (0x8)  audio       alaw   (G.711 A-law)
             16 (1 <<  4)               (0x10)  audio   g726aal2   (G.726 AAL2)
             32 (1 <<  5)               (0x20)  audio      adpcm   (ADPCM)
             64 (1 <<  6)               (0x40)  audio       slin   (16 bit Signed Linear PCM)
            128 (1 <<  7)               (0x80)  audio      lpc10   (LPC10)
            256 (1 <<  8)              (0x100)  audio       g729   (G.729A)
            512 (1 <<  9)              (0x200)  audio      speex   (SpeeX)
           1024 (1 << 10)              (0x400)  audio       ilbc   (iLBC)
           2048 (1 << 11)              (0x800)  audio       g726   (G.726 RFC3551)
           4096 (1 << 12)             (0x1000)  audio       g722   (G722)
           8192 (1 << 13)             (0x2000)  audio     siren7   (ITU G.722.1 (Siren7, licensed from Polycom))
          16384 (1 << 14)             (0x4000)  audio    siren14   (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
          32768 (1 << 15)             (0x8000)  audio     slin16   (16 bit Signed Linear PCM (16kHz))
     4294967296 (1 << 32)        (0x100000000)  audio       g719   (ITU G.719)
     8589934592 (1 << 33)        (0x200000000)  audio    speex16   (SpeeX 16khz)

…
140737488355328 (1 << 47) (0x800000000000) audio testlaw (G.711 test-law)[/code]

Ekiga is enabled with (in order of preference) the following codecs:
PCMU
PCMA
Speex 16 kHz
gsm
Speex 8 kHz
G726-32
G722

Here is the result of “sip set debug on”:

apollo*CLI> sip set debug on

<--- SIP read from UDP:88.198.19.91:5060 --->
INVITE sip:12129015252@liffysbox.dyndns.org SIP/2.0
Via: SIP/2.0/UDP 88.198.19.91:5060;branch=z9hG4bK7220d1bb;rport
From: "16466432143" <sip:16466432143@88.198.19.91>;tag=as05f0825b
To: <sip:12129015252@liffysbox.dyndns.org>
Contact: <sip:16466432143@88.198.19.91:5060>
Call-ID: 7c919ef45223bd1c4b81daad77955be5@88.198.19.91
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 09 Feb 2011 23:13:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 334

v=0
o=root 30476 30476 IN IP4 88.198.19.91
s=session
c=IN IP4 88.198.19.91
t=0 0
m=audio 16606 RTP/AVP 0 8 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 16 lines) ---

apollo*CLI> 
Sending to 88.198.19.91:5060 (NAT)
Using INVITE request as basis request - 7c919ef45223bd1c4b81daad77955be5@88.198.19.91

apollo*CLI> 
No matching peer for '16466432143' from '88.198.19.91:5060'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 88.198.19.91:16606
Looking for 12129015252 in incoming (domain liffysbox.dyndns.org)

apollo*CLI> 
list_route: hop: <sip:16466432143@88.198.19.91:5060>

<--- Transmitting (NAT) to 88.198.19.91:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 88.198.19.91:5060;branch=z9hG4bK7220d1bb;received=88.198.19.91;rport=5060
From: "16466432143" <sip:16466432143@88.198.19.91>;tag=as05f0825b
To: <sip:12129015252@liffysbox.dyndns.org>
Call-ID: 7c919ef45223bd1c4b81daad77955be5@88.198.19.91
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:12129015252@69.86.247.130:5060>
Content-Length: 0

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

apollo*CLI> 
Audio is at 5060
Adding codec 0x2 (gsm) to SDP

apollo*CLI> 
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 88.198.19.91:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.198.19.91:5060;branch=z9hG4bK7220d1bb;received=88.198.19.91;rport=5060
From: "16466432143" <sip:16466432143@88.198.19.91>;tag=as05f0825b
To: <sip:12129015252@liffysbox.dyndns.org>;tag=as7cf47926
Call-ID: 7c919ef45223bd1c4b81daad77955be5@88.198.19.91
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
C
apollo*CLI> 
ontact: <sip:12129015252@69.86.247.130:5060>
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 1625353470 1625353470 IN IP4 69.86.247.130
s=Asterisk PBX 1.8.1.1
c=IN IP4 69.86.247.130
t=0 0
m=audio 16768 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>

apollo*CLI> 

<--- SIP read from UDP:88.198.19.91:5060 --->
ACK sip:12129015252@69.86.247.130:5060 SIP/2.0
Via: SIP/2.0/UDP 88.198.19.91:5060;branch=z9hG4bK3e2b0f75;rport
From: "16466432143" <sip:16466432143@88.198.19.91>;tag=as05f0825b
To: <sip:12129015252@liffysbox.dyndns.org>;tag=as7cf47926
Contact: <sip:16466432143@88.198.19.91:5060>
Call-ID: 7c919ef45223bd1c4b81daad77955be5@88.198.19.91
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

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

apollo*CLI> 

<--- SIP read from UDP:192.168.1.2:5060 --->

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

apollo*CLI> 
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP

apollo*CLI> 
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

apollo*CLI> 
Reliably Transmitting (NAT) to 192.168.1.3:5060:
INVITE sip:121@192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2574c59f;rport
Max-Forwards: 70
From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82
To: <sip:121@192.168.1.3>
Contact: <sip:16466432143@192.168.1.9:5060>
Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.1.1
Date: Wed, 09 Feb 2011 23:21:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 1641366843 1641366843 IN IP4 192.168.1.9
s=Asterisk PBX 1.8.1.1
c=IN IP4 192.168.1.9
t=0 0
m=audio 14494 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=ptime:20
a=sendrecv
---

<--- SIP read from UDP:192.168.1.3:5060 --->
SIP/2.0 100 Trying
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2574c59f;rport
From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82
Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060
To: <sip:121@192.168.1.3>
Contact: <sip:121@192.168.1.3>
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.3:5060 --->
SIP/2.0 180 Ringing
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2574c59f;rport
User-Agent: Ekiga/3.2.6
From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82
Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060
To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.3:5060 --->
SIP/2.0 200 OK
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2574c59f;rport
User-Agent: Ekiga/3.2.6
From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82
Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060
To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0
Contact: <sip:121@192.168.1.3>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 213

v=0
o=- 1297293224 1 IN IP4 192.168.1.3
s=Opal SIP Session
c=IN IP4 192.168.1.3
t=0 0
m=audio 5066 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
<------------->
--- (11 headers 10 lines) ---

apollo*CLI> 
Found RTP audio format 0
Found RTP audio format 101

apollo*CLI> 
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.3:5066
list_route: hop: <sip:121@192.168.1.3>
set_destination: Parsing <sip:121@192.168.1.3> for address/port to send to

apollo*CLI> 
set_destination: set destination to 192.168.1.3:5060
Transmitting (NAT) to 192.168.1.3:5060:
ACK sip:121@192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK3a0153c7;rport
Max-Forwards: 70
From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82
To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0
Contact: <sip:16466432143@192.168.1.9:5060>
Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.1.1
Content-Length: 0
---

apollo*CLI> 
    -- Remotely bridging SIP/88.198.19.91-00000006 and SIP/121-00000007

apollo*CLI> 
set_destination: Parsing <sip:16466432143@88.198.19.91:5060> for address/port to send to

apollo*CLI> 
set_destination: set destination to 88.198.19.91:5060

apollo*CLI> 
Audio is at 5060

apollo*CLI> 
Adding codec 0x4 (ulaw) to SDP

apollo*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP

apollo*CLI> 
Reliably Transmitting (NAT) to 88.198.19.91:5060:
INVITE sip:16466432143@88.198.19.91:5060 SIP/2.0
Via: SIP/2.0/UDP 69.86.247.130:5060;branch=z9hG4bK689c31ae;rport
Max-Forwards: 70
From: <sip:12129015252@liffysbox.dyndns.org>;tag=as7cf47926
To: "16466432143" <sip:16466432143@88.198.19.91>;tag=as05f0825b
Contact: <sip:12129015252@69.86.247.130:5060>
Call-ID: 7c919ef45223bd1c4b81daad77955be5@88.198.19.91
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 233

v=0
o=root 1625353470 1625353471 IN IP4 192.168.1.3
s=Asterisk PBX 1.8.1.1
c=IN IP4 192.168.1.3
t=0 0
m=audio 5066 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
apollo*CLI> 
set_destination: Parsing <sip:121@192.168.1.3> for address/port to send to

apollo*CLI> 
set_destination: set destination to 192.168.1.3:5060

apollo*CLI> 
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.3:5060:
INVITE sip:121@192.168.1.3 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK313b9feb;rport

Max-Forwards: 70

From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82

To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0

Contact: <sip:16466432143@192.168.1.9:5060>

Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060

CSeq: 103 INVITE

User-Agent: Asterisk PBX 1.8.1.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

X-asterisk-Info: SIP re-invite (External RTP bridge)

Content-Type: application/sdp

Content-Length: 236



v=0

o=root 1641366843 1641366844 IN IP4 88.198.19.91

s=Asterisk PBX 1.8.1.1

c=IN IP4 88.198.19.91

t=0 0

m=audio 15582 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv


---

apollo*CLI> 

<--- SIP read from UDP:192.168.1.3:5060 --->
SIP/2.0 100 Trying
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK313b9feb;rport
From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82
Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060
To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0
Contact: <sip:121@192.168.1.3>
Content-Length: 0

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

apollo*CLI> 

<--- SIP read from UDP:192.168.1.3:5060 --->
SIP/2.0 200 OK
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK313b9feb;rport
User-Agent: Ekiga/3.2.6
From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82
Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060
To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0
Contact: <sip:121@192.168.1.3>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 213

v=0
o=- 1297293224 2 IN IP4 192.168.1.3
s=Opal SIP Session
c=IN IP4 192.168.1.3
t=0 0
m=audio 5066 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
<------------->
--- (11 headers 10 lines) ---

apollo*CLI> 
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.3:5066

apollo*CLI> 
set_destination: Parsing <sip:121@192.168.1.3> for address/port to send to

apollo*CLI> 
set_destination: set destination to 192.168.1.3:5060
Transmitting (NAT) to 192.168.1.3:5060:
ACK sip:121@192.168.1.3 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1581defa;rport

Max-Forwards: 70

From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82

To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0

Contact: <sip:16466432143@192.168.1.9:5060>

Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060

CSeq: 103 ACK

User-Agent: Asterisk PBX 1.8.1.1

Content-Length: 0




---

apollo*CLI> 

<--- SIP read from UDP:88.198.19.91:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 69.86.247.130:5060;branch=z9hG4bK689c31ae;received=69.86.247.130;rport=5060
From: <sip:12129015252@liffysbox.dyndns.org>;tag=as7cf47926
To: "16466432143" <sip:16466432143@88.198.19.91>;tag=as05f0825b
Call-ID: 7c919ef45223bd1c4b81daad77955be5@88.198.19.91
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:16466432143@88.198.19.91:5060>
Content-Length: 0

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

apollo*CLI> 

<--- SIP read from UDP:88.198.19.91:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 69.86.247.130:5060;branch=z9hG4bK689c31ae;received=69.86.247.130;rport=5060
From: <sip:12129015252@liffysbox.dyndns.org>;tag=as7cf47926
To: "16466432143" <sip:16466432143@88.198.19.91>;tag=as05f0825b
Call-ID: 7c919ef45223bd1c4b81daad77955be5@88.198.19.91
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:16466432143@88.198.19.91:5060>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 30476 30477 IN IP4 88.198.19.91
s=session
c=IN IP4 88.198.19.91
t=0 0
m=audio 16606 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 12 lines) ---

apollo*CLI> 
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 88.198.19.91:16606

apollo*CLI> 
set_destination: Parsing <sip:16466432143@88.198.19.91:5060> for address/port to send to
set_destination: set destination to 88.198.19.91:5060
Transmitting (NAT) to 88.198.19.91:5060:
ACK sip:16466432143@88.198.19.91:5060 SIP/2.0

Via: SIP/2.0/UDP 69.86.247.130:5060;branch=z9hG4bK5e6a6366;rport

Max-Forwards: 70

From: <sip:12129015252@liffysbox.dyndns.org>;tag=as7cf47926

To: "16466432143" <sip:16466432143@88.198.19.91>;tag=as05f0825b

Contact: <sip:12129015252@69.86.247.130:5060>

Call-ID: 7c919ef45223bd1c4b81daad77955be5@88.198.19.91

CSeq: 102 ACK

User-Agent: Asterisk PBX 1.8.1.1

Content-Length: 0




---
set_destination: Parsing <sip:121@192.168.1.3> for address/port to send to
set_destination: set destination to 192.168.1.3:5060

apollo*CLI> 
Audio is at 5060

apollo*CLI> 
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

apollo*CLI> 
Reliably Transmitting (NAT) to 192.168.1.3:5060:
INVITE sip:121@192.168.1.3 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4684a98d;rport

Max-Forwards: 70

From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82

To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0

Contact: <sip:16466432143@192.168.1.9:5060>

Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060

CSeq: 104 INVITE

User-Agent: Asterisk PBX 1.8.1.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

X-asterisk-Info: SIP re-invite (External RTP bridge)

Content-Type: application/sdp

Content-Length: 236



v=0

o=root 1641366843 1641366845 IN IP4 88.198.19.91

s=Asterisk PBX 1.8.1.1

c=IN IP4 88.198.19.91

t=0 0

m=audio 16606 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv


---

apollo*CLI> 

<--- SIP read from UDP:192.168.1.3:5060 --->
SIP/2.0 100 Trying
CSeq: 104 INVITE
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4684a98d;rport
From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82
Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060
To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0
Contact: <sip:121@192.168.1.3>
Content-Length: 0

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

apollo*CLI> 

<--- SIP read from UDP:192.168.1.3:5060 --->
SIP/2.0 200 OK
CSeq: 104 INVITE
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4684a98d;rport
User-Agent: Ekiga/3.2.6
From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82
Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060
To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0
Contact: <sip:121@192.168.1.3>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 213

v=0
o=- 1297293224 3 IN IP4 192.168.1.3
s=Opal SIP Session
c=IN IP4 192.168.1.3
t=0 0
m=audio 5066 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
<------------->
--- (11 headers 10 lines) ---

apollo*CLI> 
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.3:5066

apollo*CLI> 
set_destination: Parsing <sip:121@192.168.1.3> for address/port to send to
set_destination: set destination to 192.168.1.3:5060
Transmitting (NAT) to 192.168.1.3:5060:
ACK sip:121@192.168.1.3 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK7ce3c5d6;rport

Max-Forwards: 70

From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82

To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0

Contact: <sip:16466432143@192.168.1.9:5060>

Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060

CSeq: 104 ACK

User-Agent: Asterisk PBX 1.8.1.1

Content-Length: 0




---

apollo*CLI> 

<--- SIP read from UDP:192.168.1.2:5060 --->

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

apollo*CLI> 

<--- SIP read from UDP:88.198.19.91:5060 --->
BYE sip:12129015252@69.86.247.130:5060 SIP/2.0
Via: SIP/2.0/UDP 88.198.19.91:5060;branch=z9hG4bK7e2dc226;rport
From: "16466432143" <sip:16466432143@88.198.19.91>;tag=as05f0825b
To: <sip:12129015252@liffysbox.dyndns.org>;tag=as7cf47926
Call-ID: 7c919ef45223bd1c4b81daad77955be5@88.198.19.91
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

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

apollo*CLI> 
Sending to 88.198.19.91:5060 (NAT)

apollo*CLI> 
Scheduling destruction of SIP dialog '7c919ef45223bd1c4b81daad77955be5@88.198.19.91' in 32000 ms (Method: BYE)

apollo*CLI> 

<--- Transmitting (NAT) to 88.198.19.91:5060 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 88.198.19.91:5060;branch=z9hG4bK7e2dc226;received=88.198.19.91;rport=5060

From: "16466432143" <sip:16466432143@88.198.19.91>;tag=as05f0825b

To: <sip:12129015252@liffysbox.dyndns.org>;tag=as7cf47926

Call-ID: 7c919ef45223bd1c4b81daad77955be5@88.198.19.91

CSeq: 103 BYE

Server: Asterisk PBX 1.8.1.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0




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

apollo*CLI> 
set_destination: Parsing <sip:121@192.168.1.3> for address/port to send to
set_destination: set destination to 192.168.1.3:5060
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.3:5060:
INVITE sip:121@192.168.1.3 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1305785d;rport

Max-Forwards: 70

From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82

To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0

Contact: <sip:16466432143@192.168.1.9:5060>

Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060

CSeq: 105 INVITE

User-Agent: Asterisk PBX 1.8.1.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

X-asterisk-Info: SIP re-invite (External RTP bridge)

Content-Type: application/sdp

Content-Length: 234



v=0

o=root 1641366843 1641366846 IN IP4 192.168.1.9

s=Asterisk PBX 1.8.1.1

c=IN IP4 192.168.1.9

t=0 0

m=audio 14494 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv


---

apollo*CLI> 

<--- SIP read from UDP:192.168.1.3:5060 --->
SIP/2.0 100 Trying
CSeq: 105 INVITE
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1305785d;rport
From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82
Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060
To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0
Contact: <sip:121@192.168.1.3>
Content-Length: 0

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

apollo*CLI> 

<--- SIP read from UDP:192.168.1.3:5060 --->
SIP/2.0 200 OK
CSeq: 105 INVITE
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1305785d;rport
User-Agent: Ekiga/3.2.6
From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82
Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060
To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0
Contact: <sip:121@192.168.1.3>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 213

v=0
o=- 1297293224 4 IN IP4 192.168.1.3
s=Opal SIP Session
c=IN IP4 192.168.1.3
t=0 0
m=audio 5066 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
<------------->
--- (11 headers 10 lines) ---

apollo*CLI> 
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.3:5066

apollo*CLI> 
set_destination: Parsing <sip:121@192.168.1.3> for address/port to send to
set_destination: set destination to 192.168.1.3:5060
Transmitting (NAT) to 192.168.1.3:5060:
ACK sip:121@192.168.1.3 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2d257a20;rport

Max-Forwards: 70

From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82

To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0

Contact: <sip:16466432143@192.168.1.9:5060>

Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060

CSeq: 105 ACK

User-Agent: Asterisk PBX 1.8.1.1

Content-Length: 0




---

apollo*CLI> 
Scheduling destruction of SIP dialog '6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060' in 32000 ms (Method: INVITE)

apollo*CLI> 
set_destination: Parsing <sip:121@192.168.1.3> for address/port to send to

apollo*CLI> 
set_destination: set destination to 192.168.1.3:5060

apollo*CLI> 
Reliably Transmitting (NAT) to 192.168.1.3:5060:
BYE sip:121@192.168.1.3 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK14d8ed8f;rport

Max-Forwards: 70

From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82

To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0

Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060

CSeq: 106 BYE

User-Agent: Asterisk PBX 1.8.1.1

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16

Content-Length: 0




---

apollo*CLI> 

<--- SIP read from UDP:192.168.1.3:5060 --->
SIP/2.0 200 OK
CSeq: 106 BYE
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK14d8ed8f;rport
From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as5eb27a82
Call-ID: 6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060
To: <sip:121@192.168.1.3>;tag=5cfee2e8-0f33-e011-9361-00e08155f4f0
Content-Length: 0

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

apollo*CLI> 
Really destroying SIP dialog '6e08cb4d7aa9078e6e6167bd4ed30ccb@192.168.1.9:5060' Method: INVITE

apollo*CLI> 
[Feb  9 18:22:22] e[1;31mWARNINGe[0m[14026]: e[1;37mfile.ce[0m:e[1;37m751e[0m e[1;37mast_readaudio_callbacke[0m: Failed to write frame

apollo*CLI> 

<--- SIP read from UDP:192.168.1.2:5060 --->

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

apollo*CLI>

Server side (Asterisk) “tcpdump -i eth0 -n -s0 -v udp port 5060”:

17:22:29.426860 IP (tos 0xc0, ttl 250, id 49369, offset 0, flags [none], proto UDP (17), length 32)
    192.168.1.2.5060 > 192.168.1.9.5060: SIP, length: 4
	\0x00\0x00\0x00[|sip]
17:22:49.421323 IP (tos 0xc0, ttl 250, id 49370, offset 0, flags [none], proto UDP (17), length 32)
    192.168.1.2.5060 > 192.168.1.9.5060: SIP, length: 4
	\0x00\0x00\0x00[|sip]
17:22:57.064196 IP (tos 0x0, ttl 49, id 7667, offset 0, flags [none], proto UDP (17), length 935)
    88.198.19.91.5060 > 192.168.1.9.5060: SIP, length: 907
	INVITE sip:12129015252@liffysbox.dyndns.org SIP/2.0
	Via: SIP/2.0/UDP 88.198.19.91:5060;branch=z9hG4bK75dd9c9b;rport
	From: "16466432143" <sip:16466432143@88.198.19.91>;tag=as3c4fb26a
	To: <sip:12129015252@liffysbox.dyndns.org>
	Contact: <sip:16466432143@88.198.19.91:5060>
	Call-ID: 150f59212aa080f35b59f15956d9c4cd@88.198.19.91
	CSeq: 102 INVITE
	User-Agent: Asterisk PBX
	Max-Forwards: 70
	Date: Wed, 09 Feb 2011 22:14:40 GMT
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
	Supported: replaces
	Content-Type: application/sdp
	Content-Length: 334
	
	v=0
	o=root 30476 30476 IN IP4 88.198.19.91
	s=session
	c=IN IP4 88.198.19.91
	t=0 0
	m=audio 11938 RTP/AVP 0 8 18 3 101
	a=rtpmap:0 PCMU/8000
	a=rtpmap:8 PCMA/8000
	a=rtpmap:18 G729/8000
	a=fmtp:18 annexb=no
	a=rtpmap:3 GSM/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=silenceSupp:off - - - -
	a=ptime:20
	a=sendrecv
	
17:22:57.066451 IP (tos 0x0, ttl 64, id 42179, offset 0, flags [none], proto UDP (17), length 534)
    192.168.1.9.5060 > 88.198.19.91.5060: SIP, length: 506
	SIP/2.0 100 Trying
	Via: SIP/2.0/UDP 88.198.19.91:5060;branch=z9hG4bK75dd9c9b;received=88.198.19.91;rport=5060
	From: "16466432143" <sip:16466432143@88.198.19.91>;tag=as3c4fb26a
	To: <sip:12129015252@liffysbox.dyndns.org>
	Call-ID: 150f59212aa080f35b59f15956d9c4cd@88.198.19.91
	CSeq: 102 INVITE
	Server: Asterisk PBX 1.8.1.1
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	Contact: <sip:12129015252@69.86.247.130:5060>
	Content-Length: 0
	
	
17:22:57.140900 IP (tos 0x0, ttl 64, id 42180, offset 0, flags [none], proto UDP (17), length 863)
    192.168.1.9.5060 > 88.198.19.91.5060: SIP, length: 835
	SIP/2.0 200 OK
	Via: SIP/2.0/UDP 88.198.19.91:5060;branch=z9hG4bK75dd9c9b;received=88.198.19.91;rport=5060
	From: "16466432143" <sip:16466432143@88.198.19.91>;tag=as3c4fb26a
	To: <sip:12129015252@liffysbox.dyndns.org>;tag=as4d81a35e
	Call-ID: 150f59212aa080f35b59f15956d9c4cd@88.198.19.91
	CSeq: 102 INVITE
	Server: Asterisk PBX 1.8.1.1
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	Contact: <sip:12129015252@69.86.247.130:5060>
	Content-Type: application/sdp
	Content-Length: 285
	
	v=0
	o=root 1449731218 1449731218 IN IP4 69.86.247.130
	s=Asterisk PBX 1.8.1.1
	c=IN IP4 69.86.247.130
	t=0 0
	m=audio 12272 RTP/AVP 3 0 8 101
	a=rtpmap:3 GSM/8000
	a=rtpmap:0 PCMU/8000
	a=rtpmap:8 PCMA/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=ptime:20
	a=sendrecv
	
17:22:57.349099 IP (tos 0x0, ttl 49, id 7668, offset 0, flags [none], proto UDP (17), length 449)
    88.198.19.91.5060 > 192.168.1.9.5060: SIP, length: 421
	ACK sip:12129015252@69.86.247.130:5060 SIP/2.0
	Via: SIP/2.0/UDP 88.198.19.91:5060;branch=z9hG4bK2afcd526;rport
	From: "16466432143" <sip:16466432143@88.198.19.91>;tag=as3c4fb26a
	To: <sip:12129015252@liffysbox.dyndns.org>;tag=as4d81a35e
	Contact: <sip:16466432143@88.198.19.91:5060>
	Call-ID: 150f59212aa080f35b59f15956d9c4cd@88.198.19.91
	CSeq: 102 ACK
	User-Agent: Asterisk PBX
	Max-Forwards: 70
	Content-Length: 0
	
	
17:23:09.415859 IP (tos 0xc0, ttl 250, id 49371, offset 0, flags [none], proto UDP (17), length 32)
    192.168.1.2.5060 > 192.168.1.9.5060: SIP, length: 4
	\0x00\0x00\0x00[|sip]
17:23:18.427758 IP (tos 0x0, ttl 64, id 9545, offset 0, flags [none], proto UDP (17), length 879)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 851
	INVITE sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK06fa03a8;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 102 INVITE
	User-Agent: Asterisk PBX 1.8.1.1
	Date: Wed, 09 Feb 2011 22:23:18 GMT
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	Content-Type: application/sdp
	Content-Length: 281
	
	v=0
	o=root 1578362385 1578362385 IN IP4 192.168.1.9
	s=Asterisk PBX 1.8.1.1
	c=IN IP4 192.168.1.9
	t=0 0
	m=audio 11498 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=ptime:20
	a=sendrecv
	
17:23:18.429978 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 336)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 308
	SIP/2.0 100 Trying
	CSeq: 102 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK06fa03a8;rport
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>
	Contact: <sip:121@192.168.1.3>
	Content-Length: 0
	
	
17:23:18.433705 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 450)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 422
	SIP/2.0 180 Ringing
	CSeq: 102 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK06fa03a8;rport
	User-Agent: Ekiga/3.2.6
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
	Content-Length: 0
	
	
17:23:25.516689 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 723)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 695
	SIP/2.0 200 OK
	CSeq: 102 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK06fa03a8;rport
	User-Agent: Ekiga/3.2.6
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:121@192.168.1.3>
	Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
	Content-Type: application/sdp
	Content-Length: 213
	
	v=0
	o=- 1297289704 1 IN IP4 192.168.1.3
	s=Opal SIP Session
	c=IN IP4 192.168.1.3
	t=0 0
	m=audio 5064 RTP/AVP 0 101
	a=sendrecv
	a=rtpmap:0 PCMU/8000/1
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16,32,36
	
17:23:25.517711 IP (tos 0x0, ttl 64, id 9546, offset 0, flags [none], proto UDP (17), length 452)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 424
	ACK sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK107c1043;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 102 ACK
	User-Agent: Asterisk PBX 1.8.1.1
	Content-Length: 0
	
	
17:23:25.520584 IP (tos 0x0, ttl 64, id 42181, offset 0, flags [none], proto UDP (17), length 895)
    192.168.1.9.5060 > 88.198.19.91.5060: SIP, length: 867
	INVITE sip:16466432143@88.198.19.91:5060 SIP/2.0
	Via: SIP/2.0/UDP 69.86.247.130:5060;branch=z9hG4bK56b56109;rport
	Max-Forwards: 70
	From: <sip:12129015252@liffysbox.dyndns.org>;tag=as4d81a35e
	To: "16466432143" <sip:16466432143@88.198.19.91>;tag=as3c4fb26a
	Contact: <sip:12129015252@69.86.247.130:5060>
	Call-ID: 150f59212aa080f35b59f15956d9c4cd@88.198.19.91
	CSeq: 102 INVITE
	User-Agent: Asterisk PBX 1.8.1.1
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	X-asterisk-Info: SIP re-invite (External RTP bridge)
	Content-Type: application/sdp
	Content-Length: 233
	
	v=0
	o=root 1449731218 1449731219 IN IP4 192.168.1.3
	s=Asterisk PBX 1.8.1.1
	c=IN IP4 192.168.1.3
	t=0 0
	m=audio 5064 RTP/AVP 0 101
	a=rtpmap:0 PCMU/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=ptime:20
	a=sendrecv
	
17:23:25.521741 IP (tos 0x0, ttl 64, id 9547, offset 0, flags [none], proto UDP (17), length 892)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 864
	INVITE sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5cd09ab9;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 103 INVITE
	User-Agent: Asterisk PBX 1.8.1.1
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	X-asterisk-Info: SIP re-invite (External RTP bridge)
	Content-Type: application/sdp
	Content-Length: 236
	
	v=0
	o=root 1578362385 1578362386 IN IP4 88.198.19.91
	s=Asterisk PBX 1.8.1.1
	c=IN IP4 88.198.19.91
	t=0 0
	m=audio 10914 RTP/AVP 0 101
	a=rtpmap:0 PCMU/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=ptime:20
	a=sendrecv
	
17:23:25.523895 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 377)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 349
	SIP/2.0 100 Trying
	CSeq: 103 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5cd09ab9;rport
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:121@192.168.1.3>
	Content-Length: 0
	
	
17:23:25.611182 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 723)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 695
	SIP/2.0 200 OK
	CSeq: 103 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5cd09ab9;rport
	User-Agent: Ekiga/3.2.6
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:121@192.168.1.3>
	Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
	Content-Type: application/sdp
	Content-Length: 213
	
	v=0
	o=- 1297289704 2 IN IP4 192.168.1.3
	s=Opal SIP Session
	c=IN IP4 192.168.1.3
	t=0 0
	m=audio 5064 RTP/AVP 0 101
	a=sendrecv
	a=rtpmap:0 PCMU/8000/1
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16,32,36
	
17:23:25.612287 IP (tos 0x0, ttl 64, id 9548, offset 0, flags [none], proto UDP (17), length 452)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 424
	ACK sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1f4d4a8a;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 103 ACK
	User-Agent: Asterisk PBX 1.8.1.1
	Content-Length: 0
	
	
17:23:25.637602 IP (tos 0x0, ttl 49, id 7669, offset 0, flags [none], proto UDP (17), length 524)
    88.198.19.91.5060 > 192.168.1.9.5060: SIP, length: 496
	SIP/2.0 100 Trying
	Via: SIP/2.0/UDP 69.86.247.130:5060;branch=z9hG4bK56b56109;received=69.86.247.130;rport=5060
	From: <sip:12129015252@liffysbox.dyndns.org>;tag=as4d81a35e
	To: "16466432143" <sip:16466432143@88.198.19.91>;tag=as3c4fb26a
	Call-ID: 150f59212aa080f35b59f15956d9c4cd@88.198.19.91
	CSeq: 102 INVITE
	User-Agent: Asterisk PBX
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
	Supported: replaces
	Contact: <sip:16466432143@88.198.19.91:5060>
	Content-Length: 0
	
	
17:23:25.638554 IP (tos 0x0, ttl 49, id 7670, offset 0, flags [none], proto UDP (17), length 793)
    88.198.19.91.5060 > 192.168.1.9.5060: SIP, length: 765
	SIP/2.0 200 OK
	Via: SIP/2.0/UDP 69.86.247.130:5060;branch=z9hG4bK56b56109;received=69.86.247.130;rport=5060
	From: <sip:12129015252@liffysbox.dyndns.org>;tag=as4d81a35e
	To: "16466432143" <sip:16466432143@88.198.19.91>;tag=as3c4fb26a
	Call-ID: 150f59212aa080f35b59f15956d9c4cd@88.198.19.91
	CSeq: 102 INVITE
	User-Agent: Asterisk PBX
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
	Supported: replaces
	Contact: <sip:16466432143@88.198.19.91:5060>
	Content-Type: application/sdp
	Content-Length: 240
	
	v=0
	o=root 30476 30477 IN IP4 88.198.19.91
	s=session
	c=IN IP4 88.198.19.91
	t=0 0
	m=audio 11938 RTP/AVP 0 101
	a=rtpmap:0 PCMU/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=silenceSupp:off - - - -
	a=ptime:20
	a=sendrecv
	
17:23:25.639798 IP (tos 0x0, ttl 64, id 42182, offset 0, flags [none], proto UDP (17), length 458)
    192.168.1.9.5060 > 88.198.19.91.5060: SIP, length: 430
	ACK sip:16466432143@88.198.19.91:5060 SIP/2.0
	Via: SIP/2.0/UDP 69.86.247.130:5060;branch=z9hG4bK75afcbdc;rport
	Max-Forwards: 70
	From: <sip:12129015252@liffysbox.dyndns.org>;tag=as4d81a35e
	To: "16466432143" <sip:16466432143@88.198.19.91>;tag=as3c4fb26a
	Contact: <sip:12129015252@69.86.247.130:5060>
	Call-ID: 150f59212aa080f35b59f15956d9c4cd@88.198.19.91
	CSeq: 102 ACK
	User-Agent: Asterisk PBX 1.8.1.1
	Content-Length: 0
	
	
17:23:25.640985 IP (tos 0x0, ttl 64, id 9549, offset 0, flags [none], proto UDP (17), length 892)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 864
	INVITE sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4c6be58f;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 104 INVITE
	User-Agent: Asterisk PBX 1.8.1.1
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	X-asterisk-Info: SIP re-invite (External RTP bridge)
	Content-Type: application/sdp
	Content-Length: 236
	
	v=0
	o=root 1578362385 1578362387 IN IP4 88.198.19.91
	s=Asterisk PBX 1.8.1.1
	c=IN IP4 88.198.19.91
	t=0 0
	m=audio 11938 RTP/AVP 0 101
	a=rtpmap:0 PCMU/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=ptime:20
	a=sendrecv
	
17:23:25.642274 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 377)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 349
	SIP/2.0 100 Trying
	CSeq: 104 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4c6be58f;rport
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:121@192.168.1.3>
	Content-Length: 0
	
	
17:23:25.735294 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 723)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 695
	SIP/2.0 200 OK
	CSeq: 104 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4c6be58f;rport
	User-Agent: Ekiga/3.2.6
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:121@192.168.1.3>
	Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
	Content-Type: application/sdp
	Content-Length: 213
	
	v=0
	o=- 1297289704 3 IN IP4 192.168.1.3
	s=Opal SIP Session
	c=IN IP4 192.168.1.3
	t=0 0
	m=audio 5064 RTP/AVP 0 101
	a=sendrecv
	a=rtpmap:0 PCMU/8000/1
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16,32,36
	
17:23:25.737022 IP (tos 0x0, ttl 64, id 9550, offset 0, flags [none], proto UDP (17), length 452)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 424
	ACK sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK43612f99;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 104 ACK
	User-Agent: Asterisk PBX 1.8.1.1
	Content-Length: 0
	
	
17:23:29.410367 IP (tos 0xc0, ttl 250, id 49372, offset 0, flags [none], proto UDP (17), length 32)
    192.168.1.2.5060 > 192.168.1.9.5060: SIP, length: 4
	\0x00\0x00\0x00[|sip]
17:23:41.786372 IP (tos 0x0, ttl 49, id 7672, offset 0, flags [none], proto UDP (17), length 403)
    88.198.19.91.5060 > 192.168.1.9.5060: SIP, length: 375
	BYE sip:12129015252@69.86.247.130:5060 SIP/2.0
	Via: SIP/2.0/UDP 88.198.19.91:5060;branch=z9hG4bK1cc2f39d;rport
	From: "16466432143" <sip:16466432143@88.198.19.91>;tag=as3c4fb26a
	To: <sip:12129015252@liffysbox.dyndns.org>;tag=as4d81a35e
	Call-ID: 150f59212aa080f35b59f15956d9c4cd@88.198.19.91
	CSeq: 103 BYE
	User-Agent: Asterisk PBX
	Max-Forwards: 70
	Content-Length: 0
	
	
17:23:41.788644 IP (tos 0x0, ttl 64, id 42183, offset 0, flags [none], proto UDP (17), length 495)
    192.168.1.9.5060 > 88.198.19.91.5060: SIP, length: 467
	SIP/2.0 200 OK
	Via: SIP/2.0/UDP 88.198.19.91:5060;branch=z9hG4bK1cc2f39d;received=88.198.19.91;rport=5060
	From: "16466432143" <sip:16466432143@88.198.19.91>;tag=as3c4fb26a
	To: <sip:12129015252@liffysbox.dyndns.org>;tag=as4d81a35e
	Call-ID: 150f59212aa080f35b59f15956d9c4cd@88.198.19.91
	CSeq: 103 BYE
	Server: Asterisk PBX 1.8.1.1
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	Content-Length: 0
	
	
17:23:41.789481 IP (tos 0x0, ttl 64, id 9551, offset 0, flags [none], proto UDP (17), length 890)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 862
	INVITE sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4c9674d9;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 105 INVITE
	User-Agent: Asterisk PBX 1.8.1.1
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	X-asterisk-Info: SIP re-invite (External RTP bridge)
	Content-Type: application/sdp
	Content-Length: 234
	
	v=0
	o=root 1578362385 1578362388 IN IP4 192.168.1.9
	s=Asterisk PBX 1.8.1.1
	c=IN IP4 192.168.1.9
	t=0 0
	m=audio 11498 RTP/AVP 0 101
	a=rtpmap:0 PCMU/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=ptime:20
	a=sendrecv
	
17:23:41.791571 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 377)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 349
	SIP/2.0 100 Trying
	CSeq: 105 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4c9674d9;rport
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:121@192.168.1.3>
	Content-Length: 0
	
	
17:23:41.936689 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 723)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 695
	SIP/2.0 200 OK
	CSeq: 105 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4c9674d9;rport
	User-Agent: Ekiga/3.2.6
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:121@192.168.1.3>
	Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
	Content-Type: application/sdp
	Content-Length: 213
	
	v=0
	o=- 1297289704 4 IN IP4 192.168.1.3
	s=Opal SIP Session
	c=IN IP4 192.168.1.3
	t=0 0
	m=audio 5064 RTP/AVP 0 101
	a=sendrecv
	a=rtpmap:0 PCMU/8000/1
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16,32,36
	
17:23:41.937752 IP (tos 0x0, ttl 64, id 9552, offset 0, flags [none], proto UDP (17), length 452)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 424
	ACK sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1c1ef24f;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 105 ACK
	User-Agent: Asterisk PBX 1.8.1.1
	Content-Length: 0
	
	
17:23:42.340748 IP (tos 0x0, ttl 64, id 9553, offset 0, flags [none], proto UDP (17), length 480)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 452
	BYE sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK6f7df4a5;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 106 BYE
	User-Agent: Asterisk PBX 1.8.1.1
	X-Asterisk-HangupCause: Normal Clearing
	X-Asterisk-HangupCauseCode: 16
	Content-Length: 0
	
	
17:23:42.342386 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 338)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 310
	SIP/2.0 200 OK
	CSeq: 106 BYE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK6f7df4a5;rport
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Content-Length: 0

system · February 10, 2011, 12:01am

Client side (softphone) “tcpdump -i eth0 -n -s0 -v udp port 5060”:

17:15:04.532857 IP (tos 0x0, ttl 64, id 9545, offset 0, flags [none], proto UDP (17), length 879)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 851
	INVITE sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK06fa03a8;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 102 INVITE
	User-Agent: Asterisk PBX 1.8.1.1
	Date: Wed, 09 Feb 2011 22:23:18 GMT
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	Content-Type: application/sdp
	Content-Length: 281
	
	v=0
	o=root 1578362385 1578362385 IN IP4 192.168.1.9
	s=Asterisk PBX 1.8.1.1
	c=IN IP4 192.168.1.9
	t=0 0
	m=audio 11498 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=ptime:20
	a=sendrecv
	
17:15:04.534930 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 336)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 308
	SIP/2.0 100 Trying
	CSeq: 102 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK06fa03a8;rport
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>
	Contact: <sip:121@192.168.1.3>
	Content-Length: 0
	
	
17:15:04.538647 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 450)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 422
	SIP/2.0 180 Ringing
	CSeq: 102 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK06fa03a8;rport
	User-Agent: Ekiga/3.2.6
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
	Content-Length: 0
	
	
17:15:11.620726 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 723)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 695
	SIP/2.0 200 OK
	CSeq: 102 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK06fa03a8;rport
	User-Agent: Ekiga/3.2.6
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:121@192.168.1.3>
	Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
	Content-Type: application/sdp
	Content-Length: 213
	
	v=0
	o=- 1297289704 1 IN IP4 192.168.1.3
	s=Opal SIP Session
	c=IN IP4 192.168.1.3
	t=0 0
	m=audio 5064 RTP/AVP 0 101
	a=sendrecv
	a=rtpmap:0 PCMU/8000/1
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16,32,36
	
17:15:11.621859 IP (tos 0x0, ttl 64, id 9546, offset 0, flags [none], proto UDP (17), length 452)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 424
	ACK sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK107c1043;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 102 ACK
	User-Agent: Asterisk PBX 1.8.1.1
	Content-Length: 0
	
	
17:15:11.625919 IP (tos 0x0, ttl 64, id 9547, offset 0, flags [none], proto UDP (17), length 892)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 864
	INVITE sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5cd09ab9;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 103 INVITE
	User-Agent: Asterisk PBX 1.8.1.1
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	X-asterisk-Info: SIP re-invite (External RTP bridge)
	Content-Type: application/sdp
	Content-Length: 236
	
	v=0
	o=root 1578362385 1578362386 IN IP4 88.198.19.91
	s=Asterisk PBX 1.8.1.1
	c=IN IP4 88.198.19.91
	t=0 0
	m=audio 10914 RTP/AVP 0 101
	a=rtpmap:0 PCMU/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=ptime:20
	a=sendrecv
	
17:15:11.627944 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 377)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 349
	SIP/2.0 100 Trying
	CSeq: 103 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5cd09ab9;rport
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:121@192.168.1.3>
	Content-Length: 0
	
	
17:15:11.715206 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 723)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 695
	SIP/2.0 200 OK
	CSeq: 103 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5cd09ab9;rport
	User-Agent: Ekiga/3.2.6
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:121@192.168.1.3>
	Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
	Content-Type: application/sdp
	Content-Length: 213
	
	v=0
	o=- 1297289704 2 IN IP4 192.168.1.3
	s=Opal SIP Session
	c=IN IP4 192.168.1.3
	t=0 0
	m=audio 5064 RTP/AVP 0 101
	a=sendrecv
	a=rtpmap:0 PCMU/8000/1
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16,32,36
	
17:15:11.716431 IP (tos 0x0, ttl 64, id 9548, offset 0, flags [none], proto UDP (17), length 452)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 424
	ACK sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1f4d4a8a;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 103 ACK
	User-Agent: Asterisk PBX 1.8.1.1
	Content-Length: 0
	
	
17:15:11.745142 IP (tos 0x0, ttl 64, id 9549, offset 0, flags [none], proto UDP (17), length 892)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 864
	INVITE sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4c6be58f;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 104 INVITE
	User-Agent: Asterisk PBX 1.8.1.1
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	X-asterisk-Info: SIP re-invite (External RTP bridge)
	Content-Type: application/sdp
	Content-Length: 236
	
	v=0
	o=root 1578362385 1578362387 IN IP4 88.198.19.91
	s=Asterisk PBX 1.8.1.1
	c=IN IP4 88.198.19.91
	t=0 0
	m=audio 11938 RTP/AVP 0 101
	a=rtpmap:0 PCMU/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=ptime:20
	a=sendrecv
	
17:15:11.746305 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 377)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 349
	SIP/2.0 100 Trying
	CSeq: 104 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4c6be58f;rport
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:121@192.168.1.3>
	Content-Length: 0
	
	
17:15:11.839302 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 723)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 695
	SIP/2.0 200 OK
	CSeq: 104 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4c6be58f;rport
	User-Agent: Ekiga/3.2.6
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:121@192.168.1.3>
	Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
	Content-Type: application/sdp
	Content-Length: 213
	
	v=0
	o=- 1297289704 3 IN IP4 192.168.1.3
	s=Opal SIP Session
	c=IN IP4 192.168.1.3
	t=0 0
	m=audio 5064 RTP/AVP 0 101
	a=sendrecv
	a=rtpmap:0 PCMU/8000/1
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16,32,36
	
17:15:11.841140 IP (tos 0x0, ttl 64, id 9550, offset 0, flags [none], proto UDP (17), length 452)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 424
	ACK sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK43612f99;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 104 ACK
	User-Agent: Asterisk PBX 1.8.1.1
	Content-Length: 0
	
	
17:15:27.891564 IP (tos 0x0, ttl 64, id 9551, offset 0, flags [none], proto UDP (17), length 890)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 862
	INVITE sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4c9674d9;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 105 INVITE
	User-Agent: Asterisk PBX 1.8.1.1
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	X-asterisk-Info: SIP re-invite (External RTP bridge)
	Content-Type: application/sdp
	Content-Length: 234
	
	v=0
	o=root 1578362385 1578362388 IN IP4 192.168.1.9
	s=Asterisk PBX 1.8.1.1
	c=IN IP4 192.168.1.9
	t=0 0
	m=audio 11498 RTP/AVP 0 101
	a=rtpmap:0 PCMU/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=ptime:20
	a=sendrecv
	
17:15:27.893515 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 377)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 349
	SIP/2.0 100 Trying
	CSeq: 105 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4c9674d9;rport
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:121@192.168.1.3>
	Content-Length: 0
	
	
17:15:28.038602 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 723)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 695
	SIP/2.0 200 OK
	CSeq: 105 INVITE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4c9674d9;rport
	User-Agent: Ekiga/3.2.6
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:121@192.168.1.3>
	Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
	Content-Type: application/sdp
	Content-Length: 213
	
	v=0
	o=- 1297289704 4 IN IP4 192.168.1.3
	s=Opal SIP Session
	c=IN IP4 192.168.1.3
	t=0 0
	m=audio 5064 RTP/AVP 0 101
	a=sendrecv
	a=rtpmap:0 PCMU/8000/1
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16,32,36
	
17:15:28.039802 IP (tos 0x0, ttl 64, id 9552, offset 0, flags [none], proto UDP (17), length 452)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 424
	ACK sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1c1ef24f;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Contact: <sip:16466432143@192.168.1.9:5060>
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 105 ACK
	User-Agent: Asterisk PBX 1.8.1.1
	Content-Length: 0
	
	
17:15:28.442747 IP (tos 0x0, ttl 64, id 9553, offset 0, flags [none], proto UDP (17), length 480)
    192.168.1.9.5060 > 192.168.1.3.5060: SIP, length: 452
	BYE sip:121@192.168.1.3 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK6f7df4a5;rport
	Max-Forwards: 70
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	CSeq: 106 BYE
	User-Agent: Asterisk PBX 1.8.1.1
	X-Asterisk-HangupCause: Normal Clearing
	X-Asterisk-HangupCauseCode: 16
	Content-Length: 0
	
	
17:15:28.444259 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 338)
    192.168.1.3.5060 > 192.168.1.9.5060: SIP, length: 310
	SIP/2.0 200 OK
	CSeq: 106 BYE
	Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK6f7df4a5;rport
	From: "16466432143" <sip:16466432143@192.168.1.9>;tag=as7699d9f6
	Call-ID: 4e8718d215b5f29d0bc8e6316bdd9fe0@192.168.1.9:5060
	To: <sip:121@192.168.1.3>;tag=1c77c6b6-0733-e011-9361-00e08155f4f0
	Content-Length: 0

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No sound from / to incoming calls

[code]apollo*CLI> core show codecs audio Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME DESCRIPTION

Related topics

[code]apollo*CLI> core show codecs audio
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INT BINARY HEX TYPE NAME DESCRIPTION