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Nasir Javaid

2010-Jul-28 16:23 UTC

[asterisk-users] Nat issue one way audio on IP dial

hi there,

i have posted earlier on the list but got no satisfying answer. the problem
is not big.

I have asterisk server directly connected with internet (79.80.x.x) and
clients are behind router. clients/users are registered with asterisk and
are using sipura and xlite softphone.

Now problem is that when a user calls other by dialing his IP:Port (sip
uri), call is connected fine and he can hear the called user but the called
user can not here the caller voice.

If the caller calls the other user by username instead of IP:Port , the
voice is perfect both ways.

what i have noticed is that IP:Port dial is missing a parameter
"rinstance"
in "Contact" , "To" headers for adf. what is
"rinstance" for? Also something
with "Contact" header seems fishy. or RTP issue.

that is

Dial(SIP/adf,30,r) works fine with bothway audio, but

Dial(SIP/116.18.35.235:28614,30,r) has one way audio.
            /                                 \
            |                                  |
             this is IP:Port of of adf

please help as it's almost 2 weeks and i have found to suitable answer from
any forum. I nead to know what can i do to modify Headers or settings in
conf files to correct this problem.

Below is the conf of calling user

[pepsi]
username=pepsi
type=friend
secret=123456
qualify=yes
nat=no
insecure=port,invite
incominglimit=1
outgoinglimit=1
host=dynamic
dtmfmode=rfc2833
context=out
canreinvite=yes
callerid="pepsi coke" <12345678901>
accountcode=6:0:pepsi
amaflags=default
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm

Below is the conf of called user

[adf]
username=adf
type=friend
secret=123456
qualify=yes
nat=yes
insecure=port,invite
incominglimit=2
outgoinglimit=2
host=dynamic
dtmfmode=rfc2833
context=user
canreinvite=yes
callerid="adf xyz" <11223344556>
accountcode=1:0:adf
amaflags=default
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm



below is my sip debug after dialing

Audio is at 79.80.x.x port 16238
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 116.18.35.235:28614:
INVITE sip:adf at 116.18.35.235:28614 SIP/2.0
Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport
From: "pepsi coke" <sip:12345678901 at
79.80.x.x:5678>;tag=as42ec768c
To: <sip:adf at 116.18.35.235:28614>
Contact: <sip:12345678901 at 79.80.x.x:5678>
Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 21 Jul 2010 15:10:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 9626 9626 IN IP4 79.80.x.x
s=session
c=IN IP4 79.80.x.x
t=0 0
m=audio 16238 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Jul 21 11:10:22] WARNING[23814]: chan_sip.c:2872 sip_call: Setting
auto-congest time to 15000 ms.
    -- Called adf at 116.18.35.235:28614
<------------>
ast-server*CLI>
<--- SIP read from 116.18.35.235:28614 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678
Contact: <sip:adf at 116.18.35.235:28614>
To: <sip:adf at 116.18.35.235:28614>;tag=d54e632c
From: "pepsi coke"<sip:12345678901 at
79.80.x.x:5678>;tag=as42ec768c
Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
CSeq: 102 INVITE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- SIP/116.18.35.235:28614-007f4660 is ringing
ast-server*CLI>
<--- SIP read from 116.18.35.235:28614 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678
Contact: <sip:adf at 116.18.35.235:28614>
To: <sip:adf at 116.18.35.235:28614>;tag=d54e632c
From: "pepsi coke"<sip:12345678901 at
79.80.x.x:5678>;tag=as42ec768c
Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 185

v=0
o=- 6 2 IN IP4 192.168.0.12
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.12
t=0 0
m=audio 55246 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.12:55246
Found description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.12:55246
list_route: hop: <sip:adf at 116.18.35.235:28614>
[Jul 21 11:10:27] DEBUG[9707]: chan_sip.c:5695 reqprep: Strict routing
enforced for session 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
set_destination: Parsing <sip:adf at 116.18.35.235:28614> for address/port
to
send to
set_destination: set destination to 116.18.35.235, port 28614
Transmitting (NAT) to 116.18.35.235:28614:
ACK sip:adf at 116.18.35.235:28614 SIP/2.0
Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK07eb06b5;rport
From: "pepsi coke" <sip:12345678901 at
79.80.x.x:5678>;tag=as42ec768c
To: <sip:adf at 116.18.35.235:28614>;tag=d54e632c
Contact: <sip:12345678901 at 79.80.x.x:5678>
Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- Call on SIP/116.18.35.235:28614-007f4660 left from hold
    -- SIP/116.18.35.235:28614-007f4660 answered SIP/pepsi-9fdfb9a0
ast-server*CLI>
<--- SIP read from 116.18.35.235:28614 --->



<------------->
--- (0 headers 1 lines) ---
ast-server*CLI>
<--- SIP read from 116.18.35.235:28614 --->



<------------->
--- (0 headers 1 lines) ---
Scheduling destruction of SIP dialog
'0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x' in 32000 ms (Method:
INVITE)
[Jul 21 11:11:03] DEBUG[23814]: chan_sip.c:5695 reqprep: Strict routing
enforced for session 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
set_destination: Parsing <sip:adf at 116.18.35.235:28614> for address/port
to
send to
set_destination: set destination to 116.18.35.235, port 28614
Reliably Transmitting (NAT) to 116.18.35.235:28614:
BYE sip:adf at 116.18.35.235:28614 SIP/2.0
Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK36df65b5;rport
From: "pepsi coke" <sip:12345678901 at
79.80.x.x:5678>;tag=as42ec768c
To: <sip:adf at 116.18.35.235:28614>;tag=d54e632c
Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0




Nasir Javaid
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Jim Dickenson

2010-Jul-28 16:36 UTC

head link

[asterisk-users] Nat issue one way audio on IP dial

Do you have your softphone setup to use a stun server so it can send it's
public IP address in the SIP packets? I see in the SIP debug output a 192.168
address for the RTP packets to go to which of course will not work.
-- 
Jim Dickenson
mailto:dickenson at cfmc.com

CfMC
http://www.cfmc.com/



On Jul 28, 2010, at 9:23 AM, Nasir Javaid wrote:
> hi there,
> 
> i have posted earlier on the list but got no satisfying answer. the problem
is not big.
> 
> I have asterisk server directly connected with internet (79.80.x.x) and
clients are behind router. clients/users are registered with asterisk and are
using sipura and xlite softphone.
> 
> Now problem is that when a user calls other by dialing his IP:Port (sip
uri), call is connected fine and he can hear the called user but the called user
can not here the caller voice.
> 
> If the caller calls the other user by username instead of IP:Port , the
voice is perfect both ways.
> 
> what i have noticed is that IP:Port dial is missing a parameter
"rinstance" in "Contact" , "To" headers for adf.
what is "rinstance" for? Also something with "Contact"
header seems fishy. or RTP issue.
>  
> that is 
> 
> Dial(SIP/adf,30,r) works fine with bothway audio, but
> 
> Dial(SIP/116.18.35.235:28614,30,r) has one way audio.
>             /                                 \
>             |                                  |            
>              this is IP:Port of of adf
> 
> please help as it's almost 2 weeks and i have found to suitable answer
from any forum. I nead to know what can i do to modify Headers or settings in
conf files to correct this problem.
> 
> Below is the conf of calling user
> 
> [pepsi]
> username=pepsi
> type=friend
> secret=123456
> qualify=yes
> nat=no
> insecure=port,invite
> incominglimit=1
> outgoinglimit=1
> host=dynamic
> dtmfmode=rfc2833
> context=out
> canreinvite=yes
> callerid="pepsi coke" <12345678901>
> accountcode=6:0:pepsi
> amaflags=default
> disallow=all
> allow=alaw
> allow=ulaw
> allow=g729
> allow=gsm
> 
> Below is the conf of called user
> 
> [adf]
> username=adf
> type=friend
> secret=123456
> qualify=yes
> nat=yes
> insecure=port,invite
> incominglimit=2
> outgoinglimit=2
> host=dynamic
> dtmfmode=rfc2833
> context=user
> canreinvite=yes
> callerid="adf xyz" <11223344556>
> accountcode=1:0:adf
> amaflags=default
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
> allow=gsm
> 
> 
> 
> below is my sip debug after dialing
> 
> Audio is at 79.80.x.x port 16238
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (NAT) to 116.18.35.235:28614:
> INVITE sip:adf at 116.18.35.235:28614 SIP/2.0
> Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport
> From: "pepsi coke" <sip:12345678901 at
79.80.x.x:5678>;tag=as42ec768c
> To: <sip:adf at 116.18.35.235:28614>
> Contact: <sip:12345678901 at 79.80.x.x:5678>
> Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Wed, 21 Jul 2010 15:10:22 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 285
> 
> v=0
> o=root 9626 9626 IN IP4 79.80.x.x
> s=session
> c=IN IP4 79.80.x.x
> t=0 0
> m=audio 16238 RTP/AVP 8 0 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> ---
> [Jul 21 11:10:22] WARNING[23814]: chan_sip.c:2872 sip_call: Setting
auto-congest time to 15000 ms.
>     -- Called adf at 116.18.35.235:28614
> <------------>
> ast-server*CLI> 
> <--- SIP read from 116.18.35.235:28614 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678
> Contact: <sip:adf at 116.18.35.235:28614>
> To: <sip:adf at 116.18.35.235:28614>;tag=d54e632c
> From: "pepsi coke"<sip:12345678901 at
79.80.x.x:5678>;tag=as42ec768c
> Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
> CSeq: 102 INVITE
> User-Agent: X-Lite release 1104o stamp 56125
> Content-Length: 0
> 
> 
> <------------->
> --- (9 headers 0 lines) ---
>     -- SIP/116.18.35.235:28614-007f4660 is ringing
> ast-server*CLI> 
> <--- SIP read from 116.18.35.235:28614 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678
> Contact: <sip:adf at 116.18.35.235:28614>
> To: <sip:adf at 116.18.35.235:28614>;tag=d54e632c
> From: "pepsi coke"<sip:12345678901 at
79.80.x.x:5678>;tag=as42ec768c
> Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
> CSeq: 102 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
> Content-Type: application/sdp
> User-Agent: X-Lite release 1104o stamp 56125
> Content-Length: 185
> 
> v=0
> o=- 6 2 IN IP4 192.168.0.12
> s=CounterPath X-Lite 3.0
> c=IN IP4 192.168.0.12
> t=0 0
> m=audio 55246 RTP/AVP 8 0 101
> a=fmtp:101 0-15
> a=rtpmap:101 telephone-event/8000
> a=sendrecv
> 
> <------------->
> --- (11 headers 9 lines) ---
> Found RTP audio format 8
> Found RTP audio format 0
> Found RTP audio format 101
> Peer audio RTP is at port 192.168.0.12:55246
> Found description format telephone-event for ID 101
> Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
> Peer audio RTP is at port 192.168.0.12:55246
> list_route: hop: <sip:adf at 116.18.35.235:28614>
> [Jul 21 11:10:27] DEBUG[9707]: chan_sip.c:5695 reqprep: Strict routing
enforced for session 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
> set_destination: Parsing <sip:adf at 116.18.35.235:28614> for
address/port to send to
> set_destination: set destination to 116.18.35.235, port 28614
> Transmitting (NAT) to 116.18.35.235:28614:
> ACK sip:adf at 116.18.35.235:28614 SIP/2.0
> Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK07eb06b5;rport
> From: "pepsi coke" <sip:12345678901 at
79.80.x.x:5678>;tag=as42ec768c
> To: <sip:adf at 116.18.35.235:28614>;tag=d54e632c
> Contact: <sip:12345678901 at 79.80.x.x:5678>
> Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
> 
> 
> ---
>     -- Call on SIP/116.18.35.235:28614-007f4660 left from hold
>     -- SIP/116.18.35.235:28614-007f4660 answered SIP/pepsi-9fdfb9a0
> ast-server*CLI> 
> <--- SIP read from 116.18.35.235:28614 --->
> 
> 
> 
> <------------->
> --- (0 headers 1 lines) ---
> ast-server*CLI> 
> <--- SIP read from 116.18.35.235:28614 --->
> 
> 
> 
> <------------->
> --- (0 headers 1 lines) ---
> Scheduling destruction of SIP dialog '0433af7878e3a8067a40f896382cc3a6
at 79.80.x.x' in 32000 ms (Method: INVITE)
> [Jul 21 11:11:03] DEBUG[23814]: chan_sip.c:5695 reqprep: Strict routing
enforced for session 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
> set_destination: Parsing <sip:adf at 116.18.35.235:28614> for
address/port to send to
> set_destination: set destination to 116.18.35.235, port 28614
> Reliably Transmitting (NAT) to 116.18.35.235:28614:
> BYE sip:adf at 116.18.35.235:28614 SIP/2.0
> Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK36df65b5;rport
> From: "pepsi coke" <sip:12345678901 at
79.80.x.x:5678>;tag=as42ec768c
> To: <sip:adf at 116.18.35.235:28614>;tag=d54e632c
> Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
> CSeq: 103 BYE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
> 
> 
> 
> 
> Nasir Javaid
> -- 
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
> 
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Nasir Javaid

2010-Jul-29 08:34 UTC

head link

[asterisk-users] Nat issue one way audio on IP dial

thanks Jim

I will check stun server settings asap,

but i have noticed 192.168.x.x is also present in the debug of successful
call having both way audio. so i don't think this has to do anything with
this.

below is the sip debug of successful call .

---

Audio is at 79.80.154.99 port 14034

Adding codec 0x8 (alaw) to SDP

Adding codec 0x4 (ulaw) to SDP

Adding codec 0x2 (gsm) to SDP

Adding non-codec 0x1 (telephone-event) to SDP

Reliably Transmitting (NAT) to 116.18.35.235:28614:

INVITE sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0

Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport

From: "pepsi coke" <sip:12345678901 at
79.80.154.99:5678>;tag=as12245807

To: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>

Contact: <sip:12345678901 at 79.80.154.99:5678>

Call-ID: 25a6e3604896da0e5482a7565560ce3b at 79.80.154.99

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 21 Jul 2010 15:06:24 GMT

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

Supported: replaces

Content-Type: application/sdp

Content-Length: 285

v=0

o=root 9626 9626 IN IP4 79.80.154.99

s=session

c=IN IP4 79.80.154.99

t=0 0

m=audio 14034 RTP/AVP 8 0 3 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

---

[Jul 21 11:06:24] WARNING[23749]: chan_sip.c:2872 sip_call: Setting
auto-congest time to 15000 ms.

-- Called adf

ast-server*CLI>

<--- SIP read from 116.18.35.235:28614 --->

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport=5678

Contact: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>

To: <sip:adf at
116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=bd6f2350

From: "pepsi coke"<sip:12345678901 at
79.80.154.99:5678>;tag=as12245807

Call-ID: 25a6e3604896da0e5482a7565560ce3b at 79.80.154.99

CSeq: 102 INVITE

User-Agent: X-Lite release 1104o stamp 56125

Content-Length: 0

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

--- (9 headers 0 lines) ---

-- SIP/adf-00794e30 is ringing

ast-server*CLI>

<--- SIP read from 116.18.35.235:28614 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport=5678

Contact: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>

To: <sip:adf at
116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=bd6f2350

From: "pepsi coke"<sip:12345678901 at
79.80.154.99:5678>;tag=as12245807

Call-ID: 25a6e3604896da0e5482a7565560ce3b at 79.80.154.99

CSeq: 102 INVITE

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

Content-Type: application/sdp

User-Agent: X-Lite release 1104o stamp 56125

Content-Length: 185

v=0

o=- 2 2 IN IP4 192.168.0.12

s=CounterPath X-Lite 3.0

c=IN IP4 192.168.0.12

t=0 0

m=audio 15956 RTP/AVP 8 0 101

a=fmtp:101 0-15

a=rtpmap:101 telephone-event/8000

a=sendrecv

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

--- (11 headers 9 lines) ---

Found RTP audio format 8

Found RTP audio format 0

Found RTP audio format 101

Peer audio RTP is at port 192.168.0.12:15956

Found description format telephone-event for ID 101

Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)

Peer audio RTP is at port 192.168.0.12:15956

list_route: hop: <sip:adf at
116.18.35.235:28614;rinstance=0266b8b94f488588>

[Jul 21 11:06:38] DEBUG[9707]: chan_sip.c:5695 reqprep: Strict routing
enforced for session 25a6e3604896da0e5482a7565560ce3b at 79.80.154.99

set_destination: Parsing
<sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>
for address/port to send to

set_destination: set destination to 116.18.35.235, port 28614

Transmitting (NAT) to 116.18.35.235:28614:

ACK sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0

Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK00fdcc7c;rport

From: "pepsi coke" <sip:12345678901 at
79.80.154.99:5678>;tag=as12245807

To: <sip:adf at
116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=bd6f2350

Contact: <sip:12345678901 at 79.80.154.99:5678>

Call-ID: 25a6e3604896da0e5482a7565560ce3b at 79.80.154.99

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0

 ---

-- Call on SIP/adf-00794e30 left from hold

-- SIP/adf-00794e30 answered SIP/pepsi-9fd06cc0

ast-server*CLI>

<--- SIP read from 116.18.35.235:28614 --->

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

--- (0 headers 1 lines) ---

ast-server*CLI>

<--- SIP read from 116.18.35.235:28614 --->

SUBSCRIBE sip:adf at ast-server.axvoice.com:5678 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.12:28614
;branch=z9hG4bK-d8754z-7039d4338568107f-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:adf at 116.18.35.235:28614>

To: "adf"<sip:adf at ast-server.axvoice.com:5678>

From: "adf"<sip:adf at ast-server.axvoice.com:5678>;tag=5d297f22

Call-ID: MTE5N2M4ZDY1OWRjOGQwMjgyOWEzZjkzYjA3Y2RkYWY.

CSeq: 1 SUBSCRIBE

Expires: 300

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

User-Agent: X-Lite release 1104o stamp 56125

Event: message-summary

Content-Length: 0

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

--- (13 headers 0 lines) ---

Creating new subscription

Sending to 116.18.35.235 : 28614 (NAT)

Found peer 'adf'

Looking for adf in uscan_int (domain ast-server.axvoice.com)

ast-server*CLI>

<--- Transmitting (NAT) to 116.18.35.235:28614 --->

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.0.12:28614
;branch=z9hG4bK-d8754z-7039d4338568107f-1---d8754z-;received=116.18.35.235;rport=28614

From: "adf"<sip:adf at ast-server.axvoice.com:5678>;tag=5d297f22

To: "adf"<sip:adf at ast-server.axvoice.com:5678>;tag=as724c598c

Call-ID: MTE5N2M4ZDY1OWRjOGQwMjgyOWEzZjkzYjA3Y2RkYWY.

CSeq: 1 SUBSCRIBE

User-Agent: Asterisk PBX

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

Supported: replaces

Content-Length: 0

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

Really destroying SIP dialog
'MTE5N2M4ZDY1OWRjOGQwMjgyOWEzZjkzYjA3Y2RkYWY.'
Method: SUBSCRIBE

Reliably Transmitting (NAT) to 116.18.35.235:28614:

OPTIONS sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0

Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK42fde971;rport

From: "asterisk" <sip:asterisk at
79.80.154.99:5678>;tag=as223ef4a7

To: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>

Contact: <sip:asterisk at 79.80.154.99:5678>

Call-ID: 5c66fbdf4234deca50d5c44a18641582 at 79.80.154.99

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 21 Jul 2010 15:07:07 GMT

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

Supported: replaces

Content-Length: 0

 ---

ast-server*CLI>

<--- SIP read from 116.18.35.235:28614 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK42fde971;rport=5678

Contact: <sip:192.168.0.12:28614>

To: <sip:adf at
116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=15133f38

From: "asterisk"<sip:asterisk at
79.80.154.99:5678>;tag=as223ef4a7

Call-ID: 5c66fbdf4234deca50d5c44a18641582 at 79.80.154.99

CSeq: 102 OPTIONS

Accept: application/sdp

Accept-Language: en

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

User-Agent: X-Lite release 1104o stamp 56125

Content-Length: 0

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

--- (12 headers 0 lines) ---

Really destroying SIP dialog '5c66fbdf4234deca50d5c44a18641582 at
79.80.154.99'
Method: OPTIONS

ast-server*CLI>

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

Scheduling destruction of SIP dialog '
6514fece69f1718e5cefe72632909c0e at 79.80.154.99' in 23936 ms (Method:
NOTIFY)

Reliably Transmitting (NAT) to 116.18.35.235:28614:

NOTIFY sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0

Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK218cf73a;rport

From: "asterisk" <sip:asterisk at
79.80.154.99:5678>;tag=as756cae64

To: <sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>

Contact: <sip:asterisk at 79.80.154.99:5678>

Call-ID: 6514fece69f1718e5cefe72632909c0e at 79.80.154.99

CSeq: 102 NOTIFY

User-Agent: Asterisk PBX

Max-Forwards: 70

Event: message-summary

Content-Type: application/simple-message-summary

Content-Length: 92

Messages-Waiting: no

Message-Account: sip:asterisk at 79.80.154.99 <sip%3Aasterisk at
79.80.154.99>

Voice-Message: 0/0 (0/0)

---

ast-server*CLI>

<--- SIP read from 116.18.35.235:28614 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK218cf73a;rport=5678

Contact: <sip:192.168.0.12:28614>

To: <sip:adf at
116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=b9541904

From: "asterisk"<sip:asterisk at
79.80.154.99:5678>;tag=as756cae64

Call-ID: 6514fece69f1718e5cefe72632909c0e at 79.80.154.99

CSeq: 102 NOTIFY

User-Agent: X-Lite release 1104o stamp 56125

Content-Length: 0

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

--- (9 headers 0 lines) ---

Really destroying SIP dialog '6514fece69f1718e5cefe72632909c0e at
79.80.154.99'
Method: NOTIFY

[Jul 21 11:07:15] DEBUG[23749]: chan_sip.c:3074 update_call_counter: Call to
peer 'adf' removed from call limit 2

Scheduling destruction of SIP dialog '
25a6e3604896da0e5482a7565560ce3b at 79.80.154.99' in 18624 ms (Method:
INVITE)

[Jul 21 11:07:15] DEBUG[23749]: chan_sip.c:5695 reqprep: Strict routing
enforced for session 25a6e3604896da0e5482a7565560ce3b at 79.80.154.99

set_destination: Parsing
<sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588>
for address/port to send to

set_destination: set destination to 116.18.35.235, port 28614

Reliably Transmitting (NAT) to 116.18.35.235:28614:

BYE sip:adf at 116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0

Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK05cc42e6;rport

From: "pepsi coke" <sip:12345678901 at
79.80.154.99:5678>;tag=as12245807

To: <sip:adf at
116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=bd6f2350

Call-ID: 25a6e3604896da0e5482a7565560ce3b at 79.80.154.99

CSeq: 103 BYE

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0

>Date: Wed, 28 Jul 2010 09:36:51 -0700
>From: Jim Dickenson <dickenson at cfmc.com>
>Subject: Re: [asterisk-users] Nat issue one way audio on IP dial
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>      <asterisk-users at lists.digium.com>
>Message-ID: <353D789C-0987-49E1-988E-F3C98E41F8A7 at cfmc.com>
>Content-Type: text/plain; charset="us-ascii"
>Do you have your softphone setup to use a stun server so it can send
it'spublic IP address in the SIP packets? I see in the SIP >debug output a
192.168 address for the RTP packets to go to which of course will not
work.>--
>Jim Dickenson
>mailto:dickenson at cfmc.com <dickenson at cfmc.com>
>CfMC
>http://www.cfmc.com/-------------- next part --------------
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