Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Joshua Colp]

On Fri, Apr 13, 2018, at 11:56 AM, Benjamin Marty wrote:
> The current behaviour is that Earlymedia video isn't working when NAT's in
> between are involved. The source/destination IP's are correct. So the
> client is sending Early media video + Early media audio to the Asterisk
> Server "in the cloud" and the Asterisk Server "in the cloud" is sending
> both to the IP where the Client is located. But strangely just the Early
> media audio is passing the NAT to the recipent.
> 
> My guess is that the NAT traversal for Early media audio is fine, but the
> one for Early media video not yet. Can you propably comprehend something in
> that direction? Or can you guide me to the code part where Asterisk is
> doing the Port change when a NAT is detected and the Client itself is
> sending "fake" RTP Early media traffic to get a NAT Binding for incoming
> RTP Early media traffic?

The code is in res_rtp_asterisk[1]. It's not complex and despite the comment is 
not specific to video. Without logs showing where things are coming from and 
going I don't really have anything else to add.

[1] 
https://github.com/asterisk/asterisk/blob/master/res/res_rtp_asterisk.c#L6140

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Benjamin Marty]

The current behaviour is that Earlymedia video isn't working when NAT's in
between are involved. The source/destination IP's are correct. So the
client is sending Early media video + Early media audio to the Asterisk
Server "in the cloud" and the Asterisk Server "in the cloud" is sending
both to the IP where the Client is located. But strangely just the Early
media audio is passing the NAT to the recipent.

My guess is that the NAT traversal for Early media audio is fine, but the
one for Early media video not yet. Can you propably comprehend something in
that direction? Or can you guide me to the code part where Asterisk is
doing the Port change when a NAT is detected and the Client itself is
sending "fake" RTP Early media traffic to get a NAT Binding for incoming
RTP Early media traffic?

Benjamin

2018-04-11 11:50 GMT+02:00 Joshua Colp :

> On Wed, Apr 11, 2018, at 4:33 AM, Benjamin Marty wrote:
> > I added the bind_rtp_to_media_address=yes on all endpoints but still the
> > same behaviour. The funny thing is that the G711 audio early media works
> > and doesn't have that Private IP issue. I was also able to cross check
> with
> > chan_sip on Asterisk 15, exactly the same wrong behaviour. See following
> > capture (PJSIP):
>
> As I stated previously in order for media to go to the source IP address
> and port, media has to be received from the endpoint. If this doesn't
> happen then you'll see exactly this behavior - we'll send to the IP address
> and port they told us. There's nothing that Asterisk itself can do in that
> instance, the endpoint has to send media or place the correct IP address
> and port in the messages.
>
> Was any media received from it?
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Joshua Colp]

On Wed, Apr 11, 2018, at 4:33 AM, Benjamin Marty wrote:
> I added the bind_rtp_to_media_address=yes on all endpoints but still the
> same behaviour. The funny thing is that the G711 audio early media works
> and doesn't have that Private IP issue. I was also able to cross check with
> chan_sip on Asterisk 15, exactly the same wrong behaviour. See following
> capture (PJSIP):

As I stated previously in order for media to go to the source IP address and 
port, media has to be received from the endpoint. If this doesn't happen then 
you'll see exactly this behavior - we'll send to the IP address and port they 
told us. There's nothing that Asterisk itself can do in that instance, the 
endpoint has to send media or place the correct IP address and port in the 
messages.

Was any media received from it?

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Benjamin Marty]

I added the bind_rtp_to_media_address=yes on all endpoints but still the
same behaviour. The funny thing is that the G711 audio early media works
and doesn't have that Private IP issue. I was also able to cross check with
chan_sip on Asterisk 15, exactly the same wrong behaviour. See following
capture (PJSIP):

No. Time  Source
Destination   Protocol Length Info
187 2018-04-11 07:19:56.735967159.89.XX.XX
192.168.1.185 H264 943PT=H264, SSRC=0x3A7AF929, Seq=27144,
Time=1248011648 FU-A

Frame 187: 943 bytes on wire (7544 bits), 943 bytes captured (7544 bits)
Ethernet II, Src: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7), Dst:
IETF-VRRP-VRID_6e (00:00:5e:00:01:6e)
Internet Protocol Version 4, Src: 159.89.XX.XX, Dst: 192.168.1.185
User Datagram Protocol, Src Port: 11502, Dst Port: 5022
Real-Time Transport Protocol
H.264

No. Time  Source
Destination   Protocol Length Info
188 2018-04-11 07:19:56.735993159.89.XX.XX
192.168.1.185 H264 943PT=H264, SSRC=0x3A7AF929, Seq=27145,
Time=1248011648, Mark FU-A End

Frame 188: 943 bytes on wire (7544 bits), 943 bytes captured (7544 bits)
Ethernet II, Src: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7), Dst:
IETF-VRRP-VRID_6e (00:00:5e:00:01:6e)
Internet Protocol Version 4, Src: 159.89.XX.XX, Dst: 192.168.1.185
User Datagram Protocol, Src Port: 11502, Dst Port: 5022
Real-Time Transport Protocol
H.264

No. Time  Source
Destination   Protocol Length Info
189 2018-04-11 07:19:56.738966178.82.XX.XX
159.89.XX.XXRTP  214PT=ITU-T G.711 PCMU, SSRC=0x2A1A1C31,
Seq=1820, Time=1104983225

Frame 189: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits)
Ethernet II, Src: JuniperN_34:67:f0 (40:a6:77:34:67:f0), Dst:
da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7)
Internet Protocol Version 4, Src: 178.82.XX.XX, Dst: 159.89.XX.XX
User Datagram Protocol, Src Port: 5020, Dst Port: 16130
Real-Time Transport Protocol

No. Time  Source
Destination   Protocol Length Info
190 2018-04-11 07:19:56.738975178.82.XX.XX
159.89.XX.XXRTP  214PT=ITU-T G.722, SSRC=0x49CD55FD,
Seq=26679, Time=470333826

Frame 190: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits)
Ethernet II, Src: JuniperN_4f:3f:f0 (40:a6:77:4f:3f:f0), Dst:
da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7)
Internet Protocol Version 4, Src: 178.82.XX.XX, Dst: 159.89.XX.XX
User Datagram Protocol, Src Port: 5004, Dst Port: 18280
Real-Time Transport Protocol

2018-04-11 9:11 GMT+02:00 Floimair Florian <f.floim...@commend.com>:

> I did a quick check between what I have set and your settings below.
>
>
>
> You can try the following and see if it helps
>
>
>
> In your endpoint:
> bind_rtp_to_media_address=yes
>
>
>
>
>
>
>
>
>
> With best regards
>
>
>
> *Florian Floimair *Innovation - Software-Development -  VoIP & DevOps
>
>
> *COMMEND INTERNATIONAL GMBH *A-5020 Salzburg, Saalachstraße 51
> Tel: +43-662-85 62 25
> Fax: +43-662-85 62 26
> http://www.commend.com
>
>
>
> *Security and Communication by Commend *FN 178618z | LG Salzburg
>
>
>
> *Von:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] *Im Auftrag von *Benjamin Marty
> *Gesendet:* Mittwoch, 11. April 2018 08:55
> *An:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
>
> *Betreff:* Re: [asterisk-users] Asterisk behind NAT Early Media Video
>
>
>
> I think I found the root cause. The H264 Early Media video is received
> successfully on the Asterisk Server. It also seems to get processed. But
> it's send to the private IP of the receipent SIP phone.
>
> For clarification:
>
> 178.82.XX.XX is my Public IP of my Internet access. Both phones use this
> as Public IP via standard Source NAT.
>
> 159.89.XX.XX is the IP of the Asterisk Server. For this test I used a
> Server without Destination NAT. So the eth0 interface has this IP.
>
> Packet capture:
>
> No. Time  Source
> Destination   Protocol Length Info
> 141 2018-04-11 06:40:03.306561178.82.XX.XX  159.89.XX.XX
>H264 64 PT=H264, SSRC=0x3CB1E12D, Seq=19561, Time=319121408
> SPS
>
> Frame 141: 64 bytes on wire (512 bits), 64 bytes captured (512 bits)
> Ethernet II, Src: JuniperN_34:67:f0 (40:a6:77:34:67:f0), Dst:
> da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7)
> Internet Protocol Version 4, Src: 178.82.169.0, Dst: 159.89.104.193
> User Datagram Protocol, Src Port: 5006, Dst Port: 13182
> Real-Time Transport Protocol
> H.264
>
> No. Time  Source
> Destination   Protocol Length Info
> 142 2018-

Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Floimair Florian]

I did a quick check between what I have set and your settings below.

You can try the following and see if it helps

In your endpoint:
bind_rtp_to_media_address=yes




With best regards

Florian Floimair
Innovation - Software-Development -  VoIP & DevOps

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
Tel: +43-662-85 62 25
Fax: +43-662-85 62 26
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Benjamin Marty
Gesendet: Mittwoch, 11. April 2018 08:55
An: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Video

I think I found the root cause. The H264 Early Media video is received 
successfully on the Asterisk Server. It also seems to get processed. But it's 
send to the private IP of the receipent SIP phone.
For clarification:
178.82.XX.XX is my Public IP of my Internet access. Both phones use this as 
Public IP via standard Source NAT.
159.89.XX.XX is the IP of the Asterisk Server. For this test I used a Server 
without Destination NAT. So the eth0 interface has this IP.
Packet capture:
No. Time  SourceDestination 
  Protocol Length Info
141 2018-04-11 06:40:03.306561178.82.XX.XX  159.89.XX.XX
H264 64 PT=H264, SSRC=0x3CB1E12D, Seq=19561, Time=319121408 SPS

Frame 141: 64 bytes on wire (512 bits), 64 bytes captured (512 bits)
Ethernet II, Src: JuniperN_34:67:f0 (40:a6:77:34:67:f0), Dst: da:81:42:3d:d0:e7 
(da:81:42:3d:d0:e7)
Internet Protocol Version 4, Src: 178.82.169.0, Dst: 159.89.104.193
User Datagram Protocol, Src Port: 5006, Dst Port: 13182
Real-Time Transport Protocol
H.264

No. Time  SourceDestination 
  Protocol Length Info
142 2018-04-11 06:40:03.306682159.89.XX.XX192.168.XX.XX 
H264 64 PT=H264, SSRC=0x5EE97C55, Seq=30572, Time=319121408 SPS

Frame 142: 64 bytes on wire (512 bits), 64 bytes captured (512 bits)
Ethernet II, Src: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7), Dst: IETF-VRRP-VRID_6e 
(00:00:5e:00:01:6e)
Internet Protocol Version 4, Src: 159.89.104.193, Dst: 192.168.1.185
User Datagram Protocol, Src Port: 10298, Dst Port: 5022
Real-Time Transport Protocol
H.264
PJSIP.conf:
[7004]
type = endpoint
context = internal
rewrite_contact = yes
direct_media = no
rtp_symmetric = yes
;force_rport = yes
disallow = all
allow = g722, alaw, ulaw, gsm, ilbc, h264
aors = 7004
auth = auth7004

[7004]
type = aor
max_contacts = 2

[auth7004]
type=auth
auth_type=userpass
password=1234
username=7004
extensions.conf:
[internal]
exten => _700X,1,Dial(PJSIP/${EXTEN})


2018-04-10 16:43 GMT+02:00 Benjamin Marty 
<benjamin.ma...@gmail.com<mailto:benjamin.ma...@gmail.com>>:
I just noticed, the calling device isn't even sending the early media video 
stream. It just sends an early media audio stream. Is there propably a change 
in the signaling needed?
(On another P2P SIP Server the early media video works.)

2018-04-10 12:29 GMT+02:00 Benjamin Marty 
<benjamin.ma...@gmail.com<mailto:benjamin.ma...@gmail.com>>:
Hi Florian
I already have the external_media_address set in the PJSIP setup. Also the 
external_signaling_address is set to the Public IP. If I make a call from an 
Early Media (video) capable device to an Early Media capable device (also 
video) the Early Media audio works perfectly. But no video. If I sniff 
with wireshark on the recipent device I just see G711 (audio) RTP traffic. The 
h264 RTP traffic is missing before I accept the call. After accepting the call 
the h264 RTP traffic comes through.
The 183 SIP protocoll comes through. Even Asterisk is noticing it:
-- PJSIP/6002-0013 is making progress passing it to PJSIP/6001-0012

I tried both Asterisk 15 with pjsip.conf configuration and Asterisk 13 with 
sip.conf (chan_sip). In both cases I just put the both case AST_FRAME_VIDEO: 
statements before the two voice cases, like in your diff and 
recompiled/reinstalled.
Regards
Benjamin


2018-04-10 9:37 GMT+02:00 Floimair Florian 
<f.floim...@commend.com<mailto:f.floim...@commend.com>>:
Hi Benjamin!

You're obviously using a similar scenario that I have in place for testing.
I initially had issues with early media (not only video also audio) as well in 
that scenario. What I had to do was to additionally set

external_media_address=

in pjsip.conf

Also, as I wrote the patch for early-media video I'd be interested in any 
feedback from it.




With best regards

Florian Floimair
Innovation - Software-Development -  VoIP & DevOps

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
Tel: +43-662-85 62 25
Fax: +43-662-85 62 26
http://www.commend.com<https://linkprotect.cudasvc.com/url?a=http%3a%2f%2fwww.commend.

Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Benjamin Marty]

I think I found the root cause. The H264 Early Media video is received
successfully on the Asterisk Server. It also seems to get processed. But
it's send to the private IP of the receipent SIP phone.

For clarification:
178.82.XX.XX is my Public IP of my Internet access. Both phones use this as
Public IP via standard Source NAT.
159.89.XX.XX is the IP of the Asterisk Server. For this test I used a
Server without Destination NAT. So the eth0 interface has this IP.

Packet capture:
No. Time  Source
Destination   Protocol Length Info
141 2018-04-11 06:40:03.306561178.82.XX.XX  159.89.XX.XX
   H264 64 PT=H264, SSRC=0x3CB1E12D, Seq=19561, Time=319121408
SPS

Frame 141: 64 bytes on wire (512 bits), 64 bytes captured (512 bits)
Ethernet II, Src: JuniperN_34:67:f0 (40:a6:77:34:67:f0), Dst:
da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7)
Internet Protocol Version 4, Src: 178.82.169.0, Dst: 159.89.104.193
User Datagram Protocol, Src Port: 5006, Dst Port: 13182
Real-Time Transport Protocol
H.264

No. Time  Source
Destination   Protocol Length Info
142 2018-04-11 06:40:03.306682159.89.XX.XX
192.168.XX.XX H264 64 PT=H264, SSRC=0x5EE97C55, Seq=30572,
Time=319121408 SPS

Frame 142: 64 bytes on wire (512 bits), 64 bytes captured (512 bits)
Ethernet II, Src: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7), Dst:
IETF-VRRP-VRID_6e (00:00:5e:00:01:6e)
Internet Protocol Version 4, Src: 159.89.104.193, Dst: 192.168.1.185
User Datagram Protocol, Src Port: 10298, Dst Port: 5022
Real-Time Transport Protocol
H.264

PJSIP.conf:
[7004]
type = endpoint
context = internal
rewrite_contact = yes
direct_media = no
rtp_symmetric = yes
;force_rport = yes
disallow = all
allow = g722, alaw, ulaw, gsm, ilbc, h264
aors = 7004
auth = auth7004

[7004]
type = aor
max_contacts = 2

[auth7004]
type=auth
auth_type=userpass
password=1234
username=7004

extensions.conf:
[internal]
exten => _700X,1,Dial(PJSIP/${EXTEN})



2018-04-10 16:43 GMT+02:00 Benjamin Marty <benjamin.ma...@gmail.com>:

> I just noticed, the calling device isn't even sending the early media
> video stream. It just sends an early media audio stream. Is there propably
> a change in the signaling needed?
>
> (On another P2P SIP Server the early media video works.)
>
> 2018-04-10 12:29 GMT+02:00 Benjamin Marty <benjamin.ma...@gmail.com>:
>
>> Hi Florian
>>
>> I already have the external_media_address set in the PJSIP setup. Also
>> the external_signaling_address is set to the Public IP. If I make a call
>> from an Early Media (video) capable device to an Early Media capable
>> device (also video) the Early Media audio works perfectly. But no
>> video. If I sniff with wireshark on the recipent device I just see G711
>> (audio) RTP traffic. The h264 RTP traffic is missing before I accept the
>> call. After accepting the call the h264 RTP traffic comes through.
>>
>> The 183 SIP protocoll comes through. Even Asterisk is noticing it:
>> -- PJSIP/6002-0013 is making progress passing it to
>> PJSIP/6001-0012
>>
>> I tried both Asterisk 15 with pjsip.conf configuration and Asterisk 13
>> with sip.conf (chan_sip). In both cases I just put the both case
>> AST_FRAME_VIDEO: statements before the two voice cases, like in your diff
>> and recompiled/reinstalled.
>>
>> Regards
>>
>> Benjamin
>>
>>
>>
>> 2018-04-10 9:37 GMT+02:00 Floimair Florian <f.floim...@commend.com>:
>>
>>> Hi Benjamin!
>>>
>>> You're obviously using a similar scenario that I have in place for
>>> testing.
>>> I initially had issues with early media (not only video also audio) as
>>> well in that scenario. What I had to do was to additionally set
>>>
>>> external_media_address=
>>>
>>> in pjsip.conf
>>>
>>> Also, as I wrote the patch for early-media video I'd be interested in
>>> any feedback from it.
>>>
>>>
>>>
>>>
>>> With best regards
>>>
>>> Florian Floimair
>>> Innovation - Software-Development -  VoIP & DevOps
>>>
>>> COMMEND INTERNATIONAL GMBH
>>> A-5020 Salzburg, Saalachstraße 51
>>> Tel: +43-662-85 62 25
>>> Fax: +43-662-85 62 26
>>> http://www.commend.com
>>>
>>> Security and Communication by Commend
>>>
>>> FN 178618z | LG Salzburg
>>>
>>> -Ursprüngliche Nachricht-
>>> Von: asterisk-users-boun...@lists.digium.com [mailto:
>>> asterisk-users-boun...@lists.digium.com] Im Auftrag von Joshua Colp
>>> Gesendet: Montag, 9. April 2018 18:15
>>> An: asterisk-users@lis

Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Benjamin Marty]

I just noticed, the calling device isn't even sending the early media video
stream. It just sends an early media audio stream. Is there propably a
change in the signaling needed?

(On another P2P SIP Server the early media video works.)

2018-04-10 12:29 GMT+02:00 Benjamin Marty <benjamin.ma...@gmail.com>:

> Hi Florian
>
> I already have the external_media_address set in the PJSIP setup. Also the
> external_signaling_address is set to the Public IP. If I make a call from
> an Early Media (video) capable device to an Early Media capable
> device (also video) the Early Media audio works perfectly. But no
> video. If I sniff with wireshark on the recipent device I just see G711
> (audio) RTP traffic. The h264 RTP traffic is missing before I accept the
> call. After accepting the call the h264 RTP traffic comes through.
>
> The 183 SIP protocoll comes through. Even Asterisk is noticing it:
> -- PJSIP/6002-0013 is making progress passing it to PJSIP/6001-0012
>
> I tried both Asterisk 15 with pjsip.conf configuration and Asterisk 13
> with sip.conf (chan_sip). In both cases I just put the both case
> AST_FRAME_VIDEO: statements before the two voice cases, like in your diff
> and recompiled/reinstalled.
>
> Regards
>
> Benjamin
>
>
>
> 2018-04-10 9:37 GMT+02:00 Floimair Florian <f.floim...@commend.com>:
>
>> Hi Benjamin!
>>
>> You're obviously using a similar scenario that I have in place for
>> testing.
>> I initially had issues with early media (not only video also audio) as
>> well in that scenario. What I had to do was to additionally set
>>
>> external_media_address=
>>
>> in pjsip.conf
>>
>> Also, as I wrote the patch for early-media video I'd be interested in any
>> feedback from it.
>>
>>
>>
>>
>> With best regards
>>
>> Florian Floimair
>> Innovation - Software-Development -  VoIP & DevOps
>>
>> COMMEND INTERNATIONAL GMBH
>> A-5020 Salzburg, Saalachstraße 51
>> Tel: +43-662-85 62 25
>> Fax: +43-662-85 62 26
>> http://www.commend.com
>>
>> Security and Communication by Commend
>>
>> FN 178618z | LG Salzburg
>>
>> -----Ursprüngliche Nachricht-
>> Von: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] Im Auftrag von Joshua Colp
>> Gesendet: Montag, 9. April 2018 18:15
>> An: asterisk-users@lists.digium.com
>> Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Video
>>
>> On Mon, Apr 9, 2018, at 1:05 PM, Benjamin Marty wrote:
>> > wohoo, so if I unterstand it correctly with that patch early media
>> > video works over the Asterisk server? In other words the Asterisk
>> > server get's able to (process/)forward the early media video stream
>> with that patch?
>>
>> The patch forwards video while in an early media state before the call is
>> answered and bridged, yes.
>>
>> --
>> Joshua Colp
>> Digium, Inc. | Senior Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
>> https://linkprotect.cudasvc.com/url?a=https%3a%2f%2fwww.digi
>> um.com=E,1,fYho2t3OGEPSC6ILhV9IAhfyqyv57q-c2eodmmoTlhRYCnE
>> pbgeqpqYbk39h-m_lDWff7UIltd0zakv3XGb858ysVJbX0qeWGwdsbcgvduN
>> naBqVCDk,=1 & www.asterisk.org
>>
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>> m/url?a=http%3a%2f%2fwww.api-digital.com=E,1,XToemLgPy6NQ
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>> UCME4JBQMFPmNOCpc6ESOin_3Al6kti-lRo,=1 --
>>
>> Check out the new Asterisk community forum at:
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>> ysYuWrel9Apl4EqHb4_MpDTQHdQ3lJU3_Zojgbn4stUdMfchlswYSSwVO9jm
>> ol-9H658j2bZr9JmLmb9WCM5OXKTsb_DsBIYKACtBorWRSU6-q1FjJkrbc=1
>>
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Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Benjamin Marty]

Hi Florian

I already have the external_media_address set in the PJSIP setup. Also the
external_signaling_address is set to the Public IP. If I make a call from
an Early Media (video) capable device to an Early Media capable
device (also video) the Early Media audio works perfectly. But no
video. If I sniff with wireshark on the recipent device I just see G711
(audio) RTP traffic. The h264 RTP traffic is missing before I accept the
call. After accepting the call the h264 RTP traffic comes through.

The 183 SIP protocoll comes through. Even Asterisk is noticing it:
-- PJSIP/6002-0013 is making progress passing it to PJSIP/6001-0012

I tried both Asterisk 15 with pjsip.conf configuration and Asterisk 13 with
sip.conf (chan_sip). In both cases I just put the both case
AST_FRAME_VIDEO: statements before the two voice cases, like in your diff
and recompiled/reinstalled.

Regards

Benjamin



2018-04-10 9:37 GMT+02:00 Floimair Florian <f.floim...@commend.com>:

> Hi Benjamin!
>
> You're obviously using a similar scenario that I have in place for testing.
> I initially had issues with early media (not only video also audio) as
> well in that scenario. What I had to do was to additionally set
>
> external_media_address=
>
> in pjsip.conf
>
> Also, as I wrote the patch for early-media video I'd be interested in any
> feedback from it.
>
>
>
>
> With best regards
>
> Florian Floimair
> Innovation - Software-Development -  VoIP & DevOps
>
> COMMEND INTERNATIONAL GMBH
> A-5020 Salzburg, Saalachstraße 51
> Tel: +43-662-85 62 25
> Fax: +43-662-85 62 26
> http://www.commend.com
>
> Security and Communication by Commend
>
> FN 178618z | LG Salzburg
>
> -Ursprüngliche Nachricht-
> Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] Im Auftrag von Joshua Colp
> Gesendet: Montag, 9. April 2018 18:15
> An: asterisk-users@lists.digium.com
> Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Video
>
> On Mon, Apr 9, 2018, at 1:05 PM, Benjamin Marty wrote:
> > wohoo, so if I unterstand it correctly with that patch early media
> > video works over the Asterisk server? In other words the Asterisk
> > server get's able to (process/)forward the early media video stream with
> that patch?
>
> The patch forwards video while in an early media state before the call is
> answered and bridged, yes.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
> https://linkprotect.cudasvc.com/url?a=https%3a%2f%2fwww.digium.com=E,1,
> fYho2t3OGEPSC6ILhV9IAhfyqyv57q-c2eodmmoTlhRYCnEpbgeqpqYbk39h-m_
> lDWff7UIltd0zakv3XGb858ysVJbX0qeWGwdsbcgvduNnaBqVCDk,=1 &
> www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by https://linkprotect.cudasvc.
> com/url?a=http%3a%2f%2fwww.api-digital.com=E,1,
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> 2nB67YHjZewMQU1rUCME4JBQMFPmNOCpc6ESOin_3Al6kti-lRo,=1 --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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> 9H658j2bZr9JmLmb9WCM5OXKTsb_DsBIYKACtBorWRSU6-q1FjJkrbc=1
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Benjamin Marty]

I applied the patch to my Asterisk 13.20. But it seems that it still
doesn't forward the early media video stream. Do I need to put something
special into the extensions.conf? I basically just make a Dial. The calling
Client sends the 183 protocol.

[public]
exten => 6001,1,Dial(SIP/${EXTEN})

2018-04-09 18:14 GMT+02:00 Joshua Colp :

> On Mon, Apr 9, 2018, at 1:05 PM, Benjamin Marty wrote:
> > wohoo, so if I unterstand it correctly with that patch early media video
> > works over the Asterisk server? In other words the Asterisk server get's
> > able to (process/)forward the early media video stream with that patch?
>
> The patch forwards video while in an early media state before the call is
> answered and bridged, yes.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Floimair Florian]

Hi Benjamin!

You're obviously using a similar scenario that I have in place for testing.
I initially had issues with early media (not only video also audio) as well in 
that scenario. What I had to do was to additionally set

external_media_address=

in pjsip.conf

Also, as I wrote the patch for early-media video I'd be interested in any 
feedback from it.


 
 
With best regards

Florian Floimair
Innovation - Software-Development -  VoIP & DevOps

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
Tel: +43-662-85 62 25
Fax: +43-662-85 62 26
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Joshua Colp
Gesendet: Montag, 9. April 2018 18:15
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Video

On Mon, Apr 9, 2018, at 1:05 PM, Benjamin Marty wrote:
> wohoo, so if I unterstand it correctly with that patch early media 
> video works over the Asterisk server? In other words the Asterisk 
> server get's able to (process/)forward the early media video stream with that 
> patch?

The patch forwards video while in an early media state before the call is 
answered and bridged, yes.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: 
https://linkprotect.cudasvc.com/url?a=https%3a%2f%2fwww.digium.com=E,1,fYho2t3OGEPSC6ILhV9IAhfyqyv57q-c2eodmmoTlhRYCnEpbgeqpqYbk39h-m_lDWff7UIltd0zakv3XGb858ysVJbX0qeWGwdsbcgvduNnaBqVCDk,=1
 & www.asterisk.org

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Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Joshua Colp]

On Mon, Apr 9, 2018, at 1:05 PM, Benjamin Marty wrote:
> wohoo, so if I unterstand it correctly with that patch early media video
> works over the Asterisk server? In other words the Asterisk server get's
> able to (process/)forward the early media video stream with that patch?

The patch forwards video while in an early media state before the call is 
answered and bridged, yes.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

-- 
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Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Benjamin Marty]

wohoo, so if I unterstand it correctly with that patch early media video
works over the Asterisk server? In other words the Asterisk server get's
able to (process/)forward the early media video stream with that patch?

2018-04-09 17:57 GMT+02:00 Joshua Colp :

> On Mon, Apr 9, 2018, at 12:04 PM, Benjamin Marty wrote:
> > My understanding based on Wireshark analysis is that the signaling works
> > (also the recipent phone is displaying the video frame before accepting
> the
> > call), also the calling phone send video (i see that also via Wireshark)
> > but the recipent phone doesn't get any video from the Asterisk before the
> > call.
>
> Ah yeah video, I forgot that it was a recent change to add support for
> it[1]. It's not yet in any release.
>
> [1] https://gerrit.asterisk.org/#/c/8398/
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Joshua Colp]

On Mon, Apr 9, 2018, at 12:04 PM, Benjamin Marty wrote:
> My understanding based on Wireshark analysis is that the signaling works
> (also the recipent phone is displaying the video frame before accepting the
> call), also the calling phone send video (i see that also via Wireshark)
> but the recipent phone doesn't get any video from the Asterisk before the
> call.

Ah yeah video, I forgot that it was a recent change to add support for it[1]. 
It's not yet in any release.

[1] https://gerrit.asterisk.org/#/c/8398/

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Benjamin Marty]

My understanding based on Wireshark analysis is that the signaling works
(also the recipent phone is displaying the video frame before accepting the
call), also the calling phone send video (i see that also via Wireshark)
but the recipent phone doesn't get any video from the Asterisk before the
call.

2018-04-09 17:02 GMT+02:00 Joshua Colp :

> On Mon, Apr 9, 2018, at 11:53 AM, Benjamin Marty wrote:
> > Yes, media is flowing through Asterisk because both client's are behind
> > different NAT's.
>
> This doesn't answer the question of what is ACTUALLY happening in the
> scenario you describe which is very important.
>
> > Do I need to do something special in the Call Flow? Or anything
> additional
> > to the pjsip.conf?
>
> The "rtp_symmetric" option as you've used causes Asterisk to send media to
> the source of media, but it requires us to receive media. If we don't
> receive it then we send media to where they've told us to send it, which as
> I've mentioned can be wrong.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Joshua Colp]

On Mon, Apr 9, 2018, at 11:53 AM, Benjamin Marty wrote:
> Yes, media is flowing through Asterisk because both client's are behind
> different NAT's.

This doesn't answer the question of what is ACTUALLY happening in the scenario 
you describe which is very important.
 
> Do I need to do something special in the Call Flow? Or anything additional
> to the pjsip.conf?

The "rtp_symmetric" option as you've used causes Asterisk to send media to the 
source of media, but it requires us to receive media. If we don't receive it 
then we send media to where they've told us to send it, which as I've mentioned 
can be wrong.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Benjamin Marty]

Yes, media is flowing through Asterisk because both client's are behind
different NAT's.

Do I need to do something special in the Call Flow? Or anything additional
to the pjsip.conf?

2018-04-09 16:50 GMT+02:00 Joshua Colp :

> On Mon, Apr 9, 2018, at 11:42 AM, Benjamin Marty wrote:
> > Hello,
> >
> > I have an Asterisk 15 with PJSIP behind NAT (Amazon EC2).
> >
> > Now I would like to get Early Media Video working between clients in
> > different NATed networks. The 183 signalling goes trough perfectly, but
> > asterisk doesn't forward the Early Media RTP stream from the caller to
> the
> > recipent.
>
> You would need to examine things specifically and see where media is
> flowing. Is the recipient behind NAT? If so then until we receive media
> from them (wich may or may not occur with early media) we may not have the
> correct target of media.
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Asterisk behind NAT Early Media Video

[Joshua Colp]

On Mon, Apr 9, 2018, at 11:42 AM, Benjamin Marty wrote:
> Hello,
> 
> I have an Asterisk 15 with PJSIP behind NAT (Amazon EC2).
> 
> Now I would like to get Early Media Video working between clients in
> different NATed networks. The 183 signalling goes trough perfectly, but
> asterisk doesn't forward the Early Media RTP stream from the caller to the
> recipent.

You would need to examine things specifically and see where media is flowing. 
Is the recipient behind NAT? If so then until we receive media from them (wich 
may or may not occur with early media) we may not have the correct target of 
media.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] Asterisk behind NAT Early Media Video

[Benjamin Marty]

Hello,

I have an Asterisk 15 with PJSIP behind NAT (Amazon EC2).

Now I would like to get Early Media Video working between clients in
different NATed networks. The 183 signalling goes trough perfectly, but
asterisk doesn't forward the Early Media RTP stream from the caller to the
recipent.

I have the following configuration:

[6001]
type = endpoint
context = internal
rewrite_contact = yes
direct_media = no
rtp_symmetric = yes
force_rport = yes
disallow = all
allow = alaw, ulaw, h264
aors = 6001
auth = auth6001

[6001]
type = aor
max_contacts = 2

[auth6001]
type=auth
auth_type=userpass
password=1234
username=6001

Is there a Solution for an such scenario?

Thanks

Benjamin
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Re: [asterisk-users] Asterisk behind NAT and Kamailio -- Internal IP in SDP and not externip

[Johan Wilfer]

2013-07-01 15:04, Daniel-Constantin Mierla skrev:

Hello,

On 6/28/13 4:29 PM, Johan Wilfer wrote:

Hi,

We have some Asterisk servers that we are moving behind a NAT to
preserve public addresses and make room for growth. This is Asterisk 1.4

NAT works very good with the externip/localnet-setting when we are
connected directly to our teleco. But when I try to use NAT and put
them behind our Kamailio something interesting happens: The
media-address in the SDP is the internal ip and not the external.


This is the setup:

Teleco - Kamailio - Asterisk
  SIP --  1.2.3.4
   10.0.0.1 -- 10.0.0.2

externip=1.2.3.5
localnet=10.0.0.0/255.255.255.0


  RTP  1.2.3.5 (NAT:ed to 10.0.0.2)


On an incomming call from the teleco - to kamailio (public addr) -
to asterisk in the private net. Asterisk responds with the following SDP:

v=0
o=root 1889 1889 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 23344 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Asterisk seems to think that because the proxy is on the localnet, the
media is too, so it doesn't use the externip as the RTP-ip.

This is a incomming call and the RTP ip of the other leg is another
public address. So the RTP-ip should the public address (externip).

If I connect to the teleco directly from the pbx (bypassing kamailio)
Asterisk correctly uses the externip as the rtp-ip in the SDP.


I know this is an old and unsupported version of Asterisk, but any
input on the topic is welcome. If this is supported in later versions
we can maybe work around until we migrate later.



what I did when I had similar scenario was to let asterisk completely
behind NAT, using only the local IP. I used rtpproxy running on the same
host as kamailio to bridge the rtp between external and internal networks.

Cheers,
Daniel



I think that you are right that this should be done with Kamailio.
Maybe the nathelper-module in Kamilio would do the trick in modifying 
the SDP/Contact to the NAT:ed address instead of using rtpproxy.


Thanks for the feedback!

/Johan



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Re: [asterisk-users] Asterisk behind NAT and Kamailio -- Internal IP in SDP and not externip

[Daniel-Constantin Mierla]

Hello,

On 6/28/13 4:29 PM, Johan Wilfer wrote:

Hi,

We have some Asterisk servers that we are moving behind a NAT to 
preserve public addresses and make room for growth. This is Asterisk 1.4


NAT works very good with the externip/localnet-setting when we are 
connected directly to our teleco. But when I try to use NAT and put 
them behind our Kamailio something interesting happens: The 
media-address in the SDP is the internal ip and not the external.



This is the setup:

Teleco - Kamailio - Asterisk
  SIP --  1.2.3.4
   10.0.0.1 -- 10.0.0.2

externip=1.2.3.5
localnet=10.0.0.0/255.255.255.0


  RTP  1.2.3.5 (NAT:ed to 10.0.0.2)


On an incomming call from the teleco - to kamailio (public addr) - 
to asterisk in the private net. Asterisk responds with the following SDP:


v=0
o=root 1889 1889 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 23344 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Asterisk seems to think that because the proxy is on the localnet, the 
media is too, so it doesn't use the externip as the RTP-ip.


This is a incomming call and the RTP ip of the other leg is another 
public address. So the RTP-ip should the public address (externip).


If I connect to the teleco directly from the pbx (bypassing kamailio) 
Asterisk correctly uses the externip as the rtp-ip in the SDP.



I know this is an old and unsupported version of Asterisk, but any 
input on the topic is welcome. If this is supported in later versions 
we can maybe work around until we migrate later.
what I did when I had similar scenario was to let asterisk completely 
behind NAT, using only the local IP. I used rtpproxy running on the same 
host as kamailio to bridge the rtp between external and internal networks.


Cheers,
Daniel

--
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http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


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[asterisk-users] Asterisk behind NAT and Kamailio -- Internal IP in SDP and not externip

[Johan Wilfer]

Hi,

We have some Asterisk servers that we are moving behind a NAT to 
preserve public addresses and make room for growth. This is Asterisk 1.4


NAT works very good with the externip/localnet-setting when we are 
connected directly to our teleco. But when I try to use NAT and put them 
behind our Kamailio something interesting happens: The media-address in 
the SDP is the internal ip and not the external.



This is the setup:

Teleco - Kamailio - Asterisk
  SIP --  1.2.3.4
   10.0.0.1 -- 10.0.0.2

externip=1.2.3.5
localnet=10.0.0.0/255.255.255.0


  RTP  1.2.3.5 (NAT:ed to 10.0.0.2)


On an incomming call from the teleco - to kamailio (public addr) - to 
asterisk in the private net. Asterisk responds with the following SDP:


v=0
o=root 1889 1889 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 23344 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Asterisk seems to think that because the proxy is on the localnet, the 
media is too, so it doesn't use the externip as the RTP-ip.


This is a incomming call and the RTP ip of the other leg is another 
public address. So the RTP-ip should the public address (externip).


If I connect to the teleco directly from the pbx (bypassing kamailio) 
Asterisk correctly uses the externip as the rtp-ip in the SDP.



I know this is an old and unsupported version of Asterisk, but any input 
on the topic is welcome. If this is supported in later versions we can 
maybe work around until we migrate later.


Thanks!

--
Johan Wilfer

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Re: [asterisk-users] asterisk behind nat

[Leif Neland]

Den 02-03-2011 16:12, Jeremy Kister skrev:

On 3/2/2011 9:46 AM, Leif Neland wrote:

Some of the phones are being disconnected with Asterisk saying no reply
to critical packet


What kind of phones are they?  I might have nothing to do with your 
network configuration;  try adding to sip.conf [general]:


session-timers=refuse


Did no change.

A Budgetone 200 always gets disconnected, appearently not answering this:
Retransmitting #5 (no NAT) to 192.168.5.140:5060:
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP 
192.168.5.140:5060;branch=z9hG4bK9fd529935f5b4f0e;received=192.168.5.140^M

From: Merethe Neland sip:mere...@arnold.neland.dk;tag=9c97c540dba5aceb^M
To: sip:6...@arnold.neland.dk;tag=as4d2cf5b3^M
Call-ID: 13bca406eacc2ef8@192.168.5.140^M
CSeq: 5145 INVITE^M
Server: Asterisk PBX 1.8.2.4^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH^M

Supported: replaces^M
Contact: sip:6000@94.18.45.10:5060^M
Content-Type: application/sdp^M
Content-Length: 204^M
^M
v=0^M
o=root 1348141594 1348141594 IN IP4 94.18.45.10^M
s=Asterisk PBX 1.8.2.4^M
c=IN IP4 94.18.45.10^M
t=0 0^M
m=audio 14144 RTP/AVP 3^M
a=rtpmap:3 GSM/8000^M
a=silenceSupp:off - - - -^M
a=ptime:20^M
a=sendrecv^M

It is a call from phone 192.168.5.140 to echotest (6000 on 94.18.45.10)
The intro from echotest is heard until asterisk disconnects.

On a Budgetone 100, it works,
getting this line on the console -- Locally bridging SIP/9-0006 and 
SIP/musimi-0007



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[asterisk-users] asterisk behind nat

[Leif Neland]

I'm running asterisk on a Freebsd with 2 Nic's.

Inside NIC is 192.168.5.x where the phones are.
Outside NIC used to be a public IP with the ISP's device set to 
bridging, but the new WiMAX router only offers me the public ip 
94.18.x.x on the outside,

and forwarding everything to 192.168.1.50 on the Outside NIC

Some of the phones are being disconnected with Asterisk saying no reply 
to critical packet


How is Asterisk supposed to be configured?

Currently this:
externip = 94.18.x.x  ; Address that we're going to put in outbound SIP 
messages

; if we're behind a NAT
localnet = 192.168.5.0  ; Internal NETWORK address
localmask = 255.255.255.0   ; Internal netmask
; The externip, localnet and localmask 
is used
; when registering and communication 
with other proxies

; that we're registered with


tcpbindaddr=0.0.0.0
bindaddr = 0.0.0.0

Leif



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Re: [asterisk-users] asterisk behind nat

[Jeremy Kister]

On 3/2/2011 9:46 AM, Leif Neland wrote:

Some of the phones are being disconnected with Asterisk saying no reply
to critical packet


What kind of phones are they?  I might have nothing to do with your 
network configuration;  try adding to sip.conf [general]:


session-timers=refuse

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http://jeremy.kister.net./

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[asterisk-users] Asterisk behind NAT, Polycom behind NAT (SIP), how to work?

[bilal ghayyad]

Hi All;

I succeeded to have a success call from Polycom behind NAT while Asterisk has 
public IP address, but I was not able to have a succeed call (it was 
established, but no voice running, and then the call disconnected) if Asterisk 
behind NAT and Polycom behind NAT.

When Asterisk behind NAT and Polycom behind NAT, I forwarded the 5060 UDP to 
asterisk (at asterisk router) and to Polycom IP Phone at polycomg router site, 
but the problem stayed. Also I was use nat=yes in the sip.conf

Also I forwarded the udp rtp ports (that configured in rtp.conf) to the 
asterisk IP address, and did not succeed.

What else I have to do?
Any advise?
Regards
Bilal


  

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Re: [asterisk-users] Asterisk behind NAT, Polycom behind NAT (SIP), how to work?

[Gordon Henderson]

On Mon, 14 Jul 2008, bilal ghayyad wrote:

 Hi All;

 I succeeded to have a success call from Polycom behind NAT while 
 Asterisk has public IP address, but I was not able to have a succeed 
 call (it was established, but no voice running, and then the call 
 disconnected) if Asterisk behind NAT and Polycom behind NAT.

 When Asterisk behind NAT and Polycom behind NAT, I forwarded the 5060 
 UDP to asterisk (at asterisk router) and to Polycom IP Phone at polycomg 
 router site, but the problem stayed. Also I was use nat=yes in the 
 sip.conf

 Also I forwarded the udp rtp ports (that configured in rtp.conf) to the 
 asterisk IP address, and did not succeed.

 What else I have to do? Any advise? Regards Bilal

Did you set

   localnet=
and
   externip=

in sip.conf too?

(See the voip-wiki for their meanings and what to put in there)

I don't know poloycoms, but I'd suggest not having any port-forwarding at 
the phone end and get the phones to use a STUN server if they can. Only 
put port-forwarding at the asterisk side.

Gordon

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Re: [asterisk-users] Asterisk behind NAT, Polycom behind NAT (SIP), how to work?

[Noah Miller]

Hi Bilal -

 When Asterisk behind NAT and Polycom behind NAT, I forwarded the 5060 UDP to 
 asterisk
 (at asterisk router) and to Polycom IP Phone at polycomg router site, but the 
 problem stayed.
 Also I was use nat=yes in the sip.conf

 Also I forwarded the udp rtp ports (that configured in rtp.conf) to the 
 asterisk IP address, and did
 not succeed.

Only forward ports (UDP 5060 and RTP) at the asterisk end.  Do not
forward any ports at the phone end.

- Noah

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[asterisk-users] Asterisk behind NAT

[Nitesh Divecha]

Hello All,

Has anyone implemented Asterisk behind D-Link Router?
Got one pain in butt customer who wants to setup * system behind D-Link 
router model DI-624?


Can anyone share their conf?

Thanks,
Nitesh


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Re: [asterisk-users] Asterisk behind NAT

[Anthony Francis]

Nitesh Divecha wrote:

Hello All,

Has anyone implemented Asterisk behind D-Link Router?
Got one pain in butt customer who wants to setup * system behind 
D-Link router model DI-624?


Can anyone share their conf?

Thanks,
Nitesh


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just use nat=yes and make sure the other side's configuration is 
expecting NAT and then forward the porper ports throught the firewall. 
that is if this box needs to connect via sip to anything on the net.

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Re: [asterisk-users] Asterisk behind NAT and without portforwarding forrtp

[Thomas Winter]

Am Sunday 29 October 2006 01:31 schrieb Dovid B:
 Half asleep. Sorry for my last post. I believe you still need port
 forwarding for IAX. Time to keep to my bed time.

If works as long as you have notransfer=no at both ends.

Iam concerned that with SIP Asterisk is bridging up and I do not receive the 
audio stream.
Asterisk should Hangup the line if Audio stream is announced to com from 
another IP.

Iam wonderung that there is no setting for this.
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[asterisk-users] Asterisk behind NAT and without portforwarding for rtp

[Thomas Winter]

Hi,
I have an Asterisk behind NAT.
NAT=yes and canreinvite=no in globals and for the peer.

I call an peer. The peer advice to use another IP for the audio and my 
Asterisk is sending audio stream to the Audio server.
Because of missing port forwarding I will not receive the audio stream and 
hear nothing.

I would expect that Asterisk will cancel the connection, but this didnt 
happened. Asterisk will follow the reinvite from the peer.

Any solution except portforwarding?

best regards

Thomas
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Re: [asterisk-users] Asterisk behind NAT and without portforwarding forrtp

[Dovid B]

yup. use IAX
- Original Message - 
From: Thomas Winter [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Sunday, October 29, 2006 1:26 AM
Subject: [asterisk-users] Asterisk behind NAT and without portforwarding 
forrtp




Hi,
I have an Asterisk behind NAT.
NAT=yes and canreinvite=no in globals and for the peer.

I call an peer. The peer advice to use another IP for the audio and my
Asterisk is sending audio stream to the Audio server.
Because of missing port forwarding I will not receive the audio stream and
hear nothing.

I would expect that Asterisk will cancel the connection, but this didnt
happened. Asterisk will follow the reinvite from the peer.

Any solution except portforwarding?

best regards

Thomas
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Re: [asterisk-users] Asterisk behind NAT and without portforwarding forrtp

[Dovid B]

Half asleep. Sorry for my last post. I believe you still need port 
forwarding for IAX. Time to keep to my bed time.



- Original Message - 
From: Thomas Winter [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Sunday, October 29, 2006 1:26 AM
Subject: [asterisk-users] Asterisk behind NAT and without portforwarding 
forrtp




Hi,
I have an Asterisk behind NAT.
NAT=yes and canreinvite=no in globals and for the peer.

I call an peer. The peer advice to use another IP for the audio and my
Asterisk is sending audio stream to the Audio server.
Because of missing port forwarding I will not receive the audio stream and
hear nothing.

I would expect that Asterisk will cancel the connection, but this didnt
happened. Asterisk will follow the reinvite from the peer.

Any solution except portforwarding?

best regards

Thomas
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[Asterisk-Users] Asterisk behind NAT

[Joao Pereira]

Hello to all
Can we put Asterisk in a company that has an ADSL connection with just 
one public IP address? Because with just one public IP, Asterisk must 
have a private (NATed) IP... but the idea is to make him dial other SIP 
domains.


Can Asterisk work behing NAT, and still route calls to the Internet?
And he can still receive calls from the Internet?

Thanks
Joao Pereira
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RE: [Asterisk-Users] Asterisk behind NAT

[Kerry Garrison]

Yes.

In Sip.conf you need the following lines:

externip=xxx.xxx.xxx.xxx ; put public ip address here
localnet=192.168.10.0/255.255.255.0 ; edit as appropriate

In your firewall, add the following mappings to your server:

5060-5061 UDP
10,000 - 20,000 UDP

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com



 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Joao Pereira
 Sent: Thursday, April 06, 2006 8:05 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Asterisk behind NAT
 
 Hello to all
 Can we put Asterisk in a company that has an ADSL connection 
 with just one public IP address? Because with just one public 
 IP, Asterisk must have a private (NATed) IP... but the idea 
 is to make him dial other SIP domains.
 
 Can Asterisk work behing NAT, and still route calls to the Internet?
 And he can still receive calls from the Internet?
 
 Thanks
 Joao Pereira
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Re: [Asterisk-Users] Asterisk behind NAT

[Giridhar Reddy Bandi]

IT works fine behind firewall .enable NAT in sip.conf and it works fine.Giridhar BandiOn 4/6/06, Joao Pereira 
[EMAIL PROTECTED] wrote:Hello to allCan we put Asterisk in a company that has an ADSL connection with just
one public IP address? Because with just one public IP, Asterisk musthave a private (NATed) IP... but the idea is to make him dial other SIPdomains.Can Asterisk work behing NAT, and still route calls to the Internet?
And he can still receive calls from the Internet?ThanksJoao Pereira___--Bandwidth and Colocation provided by Easynews.com
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Re: [Asterisk-Users] Asterisk behind NAT

[Joao Pereira]

Thank you very much
And if I put the correct SRV records in the DNS, can Asterisk receive 
calls??


How does the router knows, that the call must be delivered to Asterisk? 
Can I map all the requests that reach the router port 5060, to be 
delivered in 192.168.0.50 ?


Did someone implemented successfully a SIP domain in Asterisk behind NAT?
Thanks
Joao Pereira


Kerry Garrison wrote:


Yes.

In Sip.conf you need the following lines:

externip=xxx.xxx.xxx.xxx ; put public ip address here
localnet=192.168.10.0/255.255.255.0 ; edit as appropriate

In your firewall, add the following mappings to your server:

5060-5061 UDP
10,000 - 20,000 UDP

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com



 


-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Joao Pereira

Sent: Thursday, April 06, 2006 8:05 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk behind NAT

Hello to all
Can we put Asterisk in a company that has an ADSL connection 
with just one public IP address? Because with just one public 
IP, Asterisk must have a private (NATed) IP... but the idea 
is to make him dial other SIP domains.


Can Asterisk work behing NAT, and still route calls to the Internet?
And he can still receive calls from the Internet?

Thanks
Joao Pereira
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RE: [Asterisk-Users] Asterisk behind NAT

[hgaillac-sip]

Hello,

I advise you to install open(ser) with natelper
module.

Harry
--- Kerry Garrison [EMAIL PROTECTED] a écrit
:

 Yes.
 
 In Sip.conf you need the following lines:
 
 externip=xxx.xxx.xxx.xxx ; put public ip address
 here
 localnet=192.168.10.0/255.255.255.0 ; edit as
 appropriate
 
 In your firewall, add the following mappings to your
 server:
 
 5060-5061 UDP
 10,000 - 20,000 UDP
 
 Kerry Garrison
 Director of Technical Services
 Tech Data Pros - Orange County's Mobile IT Service
 Provider
 (949) 502-7819 x200 - [EMAIL PROTECTED]
 http://www.techdatapros.com
 
 
 
  -Original Message-
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]
 On Behalf Of 
  Joao Pereira
  Sent: Thursday, April 06, 2006 8:05 AM
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] Asterisk behind NAT
  
  Hello to all
  Can we put Asterisk in a company that has an ADSL
 connection 
  with just one public IP address? Because with just
 one public 
  IP, Asterisk must have a private (NATed) IP... but
 the idea 
  is to make him dial other SIP domains.
  
  Can Asterisk work behing NAT, and still route
 calls to the Internet?
  And he can still receive calls from the Internet?
  
  Thanks
  Joao Pereira
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Re: [Asterisk-Users] Asterisk behind NAT

[Andre Ruiz]

Kerry Garrison said exactly that three messages before in this thread.

You need to map 5060 and all the RTP ports (1 to 2 unless you
changed that).

BTW, Giridhar said two messages before that you should set nat=yes in
sip.conf. Notice that this is for when the *peer* is behind nat and
reaches asterisk outside nat. It will not help when asterisk is inside
nat and being reached from outside.

andre

On 4/6/06, Joao Pereira [EMAIL PROTECTED] wrote:
 Thank you very much
 And if I put the correct SRV records in the DNS, can Asterisk receive
 calls??

 How does the router knows, that the call must be delivered to Asterisk?
 Can I map all the requests that reach the router port 5060, to be
 delivered in 192.168.0.50 ?

 Did someone implemented successfully a SIP domain in Asterisk behind NAT?
 Thanks
 Joao Pereira


 Kerry Garrison wrote:

 Yes.
 
 In Sip.conf you need the following lines:
 
 externip=xxx.xxx.xxx.xxx ; put public ip address here
 localnet=192.168.10.0/255.255.255.0 ; edit as appropriate
 
 In your firewall, add the following mappings to your server:
 
 5060-5061 UDP
 10,000 - 20,000 UDP
 
 Kerry Garrison
 Director of Technical Services
 Tech Data Pros - Orange County's Mobile IT Service Provider
 (949) 502-7819 x200 - [EMAIL PROTECTED]
 http://www.techdatapros.com
 
 
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Joao Pereira
 Sent: Thursday, April 06, 2006 8:05 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Asterisk behind NAT
 
 Hello to all
 Can we put Asterisk in a company that has an ADSL connection
 with just one public IP address? Because with just one public
 IP, Asterisk must have a private (NATed) IP... but the idea
 is to make him dial other SIP domains.
 
 Can Asterisk work behing NAT, and still route calls to the Internet?
 And he can still receive calls from the Internet?
 
 Thanks
 Joao Pereira
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 To UNSUBSCRIBE or update options visit:
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--
Andre Ruiz  [EMAIL PROTECTED]
Curitiba, PR, Brasil
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Re: [Asterisk-Users] Asterisk behind NAT

[Giridhar Reddy Bandi]

so that means that a sip client can access asterisk server which is behind NAT ( assuming that SIP and RTP ports are properly farwarded ) even is nat=no in sip.conf thanks,Giridhar Bandi.
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Re: [Asterisk-Users] Asterisk behind nat

[Thameem Ansari]

As for the current release asterisk will not support STUN. You will
have problems when you run asterisk behind NAT and try to configure a
remote extension. Refer voxilla.com forums for more details. 
On 10/4/05, Anders Svensson [EMAIL PROTECTED] wrote:















Hi!

How do I configure my * to have a remote extension if
the asterisk is behind a nat?





Regards

Anders Svensson













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[Asterisk-Users] Asterisk behind nat

[Anders Svensson]

Hi!

How do I configure my * to have a remote extension if
the asterisk is behind a nat?





Regards

Anders Svensson












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Re: [Asterisk-Users] Asterisk behind NAT

[Mark Phillips]

Try putting nat=yes in the stanza that deals with your provider. I do 
this with broadvoice and galaxyvoice and they both work fine. Take it 
out and they don't work.

Mark
Oswaldo Arratia wrote:
I have...   Externip=x.x.x.xand nothing... Does not seem to help in
anything. Still my provider sees the private IP. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev
Sent: Friday, April 15, 2005 9:05 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT
Try setting externip=(asterisk public ip address)
Hth
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 12:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk behind NAT
Hi List,
I've spent hours researching on this topic, found tons of info, so far it
doesn't work yet.
Here's the scenario
Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to
send calls to an outside provider.
My SIP phones (outside * NAT) are able to register with no problem.
The problem comes when I send a call out to my provider I get no audio in
either way. My provider sees only my private IP.  How can I send my provider
the public IP??
I have in my sip.conf:
[general]
port=5060   ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
nat=yes
externip = x.x.x.x
localnet=192.168.1.0/24
(remember I enabled DMZ on the router so all ports are being forwarded to *)
Despite of what I do, the externip and the localnet my provider only gets my
* private IP.
Any tip how to solve this??
Thanks
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RE: [Asterisk-Users] Asterisk behind NAT

[Alex Vishnev]

Try setting externip=(asterisk public ip address)

Hth
Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 12:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk behind NAT

Hi List,
I've spent hours researching on this topic, found tons of info, so far it
doesn't work yet.

Here's the scenario

Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to
send calls to an outside provider.

My SIP phones (outside * NAT) are able to register with no problem.


The problem comes when I send a call out to my provider I get no audio in
either way. My provider sees only my private IP.  How can I send my provider
the public IP??


I have in my sip.conf:

[general]
port=5060   ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
nat=yes
externip = x.x.x.x
localnet=192.168.1.0/24

(remember I enabled DMZ on the router so all ports are being forwarded to *)
Despite of what I do, the externip and the localnet my provider only gets my
* private IP.

Any tip how to solve this??

Thanks


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RE: [Asterisk-Users] Asterisk behind NAT

[Oswaldo Arratia]

I have...   Externip=x.x.x.xand nothing... Does not seem to help in
anything. Still my provider sees the private IP. 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev
Sent: Friday, April 15, 2005 9:05 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

Try setting externip=(asterisk public ip address)

Hth
Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 12:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk behind NAT

Hi List,
I've spent hours researching on this topic, found tons of info, so far it
doesn't work yet.

Here's the scenario

Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to
send calls to an outside provider.

My SIP phones (outside * NAT) are able to register with no problem.


The problem comes when I send a call out to my provider I get no audio in
either way. My provider sees only my private IP.  How can I send my provider
the public IP??


I have in my sip.conf:

[general]
port=5060   ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
nat=yes
externip = x.x.x.x
localnet=192.168.1.0/24

(remember I enabled DMZ on the router so all ports are being forwarded to *)
Despite of what I do, the externip and the localnet my provider only gets my
* private IP.

Any tip how to solve this??

Thanks


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RE: [Asterisk-Users] Asterisk behind NAT

[Alex Vishnev]

Can you show your outbound peer configuration? If you are registering,
please include that as well.

Thanks

Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 9:44 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

I have...   Externip=x.x.x.xand nothing... Does not seem to help in
anything. Still my provider sees the private IP. 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev
Sent: Friday, April 15, 2005 9:05 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

Try setting externip=(asterisk public ip address)

Hth
Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 12:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk behind NAT

Hi List,
I've spent hours researching on this topic, found tons of info, so far it
doesn't work yet.

Here's the scenario

Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to
send calls to an outside provider.

My SIP phones (outside * NAT) are able to register with no problem.


The problem comes when I send a call out to my provider I get no audio in
either way. My provider sees only my private IP.  How can I send my provider
the public IP??


I have in my sip.conf:

[general]
port=5060   ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
nat=yes
externip = x.x.x.x
localnet=192.168.1.0/24

(remember I enabled DMZ on the router so all ports are being forwarded to *)
Despite of what I do, the externip and the localnet my provider only gets my
* private IP.

Any tip how to solve this??

Thanks


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RE: [Asterisk-Users] Asterisk behind NAT

[Jim Sturtevant]

Do you have any phones connected to your * on the internal subnet?  Can they
make outbound calls?  



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev
Sent: Friday, April 15, 2005 10:15 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

Can you show your outbound peer configuration? If you are registering,
please include that as well.

Thanks

Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 9:44 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

I have...   Externip=x.x.x.xand nothing... Does not seem to help in
anything. Still my provider sees the private IP. 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev
Sent: Friday, April 15, 2005 9:05 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

Try setting externip=(asterisk public ip address)

Hth
Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 12:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk behind NAT

Hi List,
I've spent hours researching on this topic, found tons of info, so far it
doesn't work yet.

Here's the scenario

Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to
send calls to an outside provider.

My SIP phones (outside * NAT) are able to register with no problem.


The problem comes when I send a call out to my provider I get no audio in
either way. My provider sees only my private IP.  How can I send my provider
the public IP??


I have in my sip.conf:

[general]
port=5060   ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
nat=yes
externip = x.x.x.x
localnet=192.168.1.0/24

(remember I enabled DMZ on the router so all ports are being forwarded to *)
Despite of what I do, the externip and the localnet my provider only gets my
* private IP.

Any tip how to solve this??

Thanks


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RE: [Asterisk-Users] Asterisk behind NAT

[Oswaldo Arratia]

I am not registering, only sending calls, here is the config for the general
section and for that provider (gw2).

[general]
context=default ; Default context for incoming calls
recordhistory=yes   ; Record SIP history by default
; (see sip history / sip no history)
realm=asterisk  ; Realm for digest authentication
; defaults to asterisk
; Realms MUST be globally unique according
to RFC 3261
; Set this to your host name or domain name
port=5060   ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=no; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the
Internet

maxexpirey=3600 ; Max length of incoming registration we
allow
defaultexpirey=3600 ; Default length of incoming/outoing
registration

disallow=all
;allow=ulaw
allow=g729

language=en ; Default language setting for all
users/peers
; This may also be set for individual
users/peers
rtptimeout=300   ; Terminate call if 60 seconds of no RTP
activity
; when we're not on hold
rtpholdtimeout=300  ; Terminate call if 300 seconds of no RTP
activity
; when we're on hold (must be  rtptimeout)
;progressinband=no  ; If we should generate in-band ringing
always

useragent=Asterisk  ; Allows you to change the user
agent string

nat=yes

externip = 1.3.5.7
localnet=192.168.1.0/255.255.255.0


[gw2]
type=peer
port=5060
host=2.4.6.8
disallow=all
defaultip=2.4.6.8
allow=g729

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev
Sent: Friday, April 15, 2005 1:15 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

Can you show your outbound peer configuration? If you are registering,
please include that as well.

Thanks

Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 9:44 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

I have...   Externip=x.x.x.xand nothing... Does not seem to help in
anything. Still my provider sees the private IP. 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev
Sent: Friday, April 15, 2005 9:05 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

Try setting externip=(asterisk public ip address)

Hth
Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 12:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk behind NAT

Hi List,
I've spent hours researching on this topic, found tons of info, so far it
doesn't work yet.

Here's the scenario

Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to
send calls to an outside provider.

My SIP phones (outside * NAT) are able to register with no problem.


The problem comes when I send a call out to my provider I get no audio in
either way. My provider sees only my private IP.  How can I send my provider
the public IP??


I have in my sip.conf:

[general]
port=5060   ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
nat=yes
externip = x.x.x.x
localnet=192.168.1.0/24

(remember I enabled DMZ on the router so all ports are being forwarded to *)
Despite of what I do, the externip and the localnet my provider only gets my
* private IP.

Any tip how to solve this??

Thanks


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RE: [Asterisk-Users] Asterisk behind NAT

[Oswaldo Arratia]

No phones in local LAN but I can try that, let me do that and I'll get back
to you. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Sturtevant
Sent: Friday, April 15, 2005 1:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

Do you have any phones connected to your * on the internal subnet?  Can they
make outbound calls?  



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev
Sent: Friday, April 15, 2005 10:15 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

Can you show your outbound peer configuration? If you are registering,
please include that as well.

Thanks

Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 9:44 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

I have...   Externip=x.x.x.xand nothing... Does not seem to help in
anything. Still my provider sees the private IP. 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev
Sent: Friday, April 15, 2005 9:05 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT

Try setting externip=(asterisk public ip address)

Hth
Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 12:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk behind NAT

Hi List,
I've spent hours researching on this topic, found tons of info, so far it
doesn't work yet.

Here's the scenario

Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to
send calls to an outside provider.

My SIP phones (outside * NAT) are able to register with no problem.


The problem comes when I send a call out to my provider I get no audio in
either way. My provider sees only my private IP.  How can I send my provider
the public IP??


I have in my sip.conf:

[general]
port=5060   ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
nat=yes
externip = x.x.x.x
localnet=192.168.1.0/24

(remember I enabled DMZ on the router so all ports are being forwarded to *)
Despite of what I do, the externip and the localnet my provider only gets my
* private IP.

Any tip how to solve this??

Thanks


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Re: [Asterisk-Users] Asterisk behind NAT

[Michiel van Baak]

On 13:45, Fri 15 Apr 05, Oswaldo Arratia wrote:
 [gw2]
 type=peer
 port=5060
 host=2.4.6.8
 disallow=all
 defaultip=2.4.6.8
 allow=g729
 

Hi,

Put this line in there:
canreinvite=no

That fixed a lot of nat issues for me.

-- 
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Two of the most famous products of Berkeley are LSD and BSD. I don't think 
that this is a coincidence.

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RE: [Asterisk-Users] Asterisk behind NAT

[Oswaldo Arratia]

I put it in there, it worked.  I took it off and continued working (I reload
of course).
Those * misteries!!!  I'll continue testing and let you know what I see.

Thanks for the pointer!
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent: Friday, April 15, 2005 2:01 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk behind NAT

On 13:45, Fri 15 Apr 05, Oswaldo Arratia wrote:
 [gw2]
 type=peer
 port=5060
 host=2.4.6.8
 disallow=all
 defaultip=2.4.6.8
 allow=g729
 

Hi,

Put this line in there:
canreinvite=no

That fixed a lot of nat issues for me.

--
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Two of the most famous products of Berkeley are LSD and BSD. I don't think
that this is a coincidence.

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[Asterisk-Users] Asterisk behind NAT

[Oswaldo Arratia]

Hi List,
I've spent hours researching on this topic, found tons of info, so far it
doesn't work yet.

Here's the scenario

Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to
send calls to an outside provider.

My SIP phones (outside * NAT) are able to register with no problem.


The problem comes when I send a call out to my provider I get no audio in
either way. My provider sees only my private IP.  How can I send my provider
the public IP??


I have in my sip.conf:

[general]
port=5060   ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
nat=yes
externip = x.x.x.x
localnet=192.168.1.0/24

(remember I enabled DMZ on the router so all ports are being forwarded to *)
Despite of what I do, the externip and the localnet my provider only gets my
* private IP.

Any tip how to solve this??

Thanks


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RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file

[David J Carter]

I have used the Draytek 2600V router in a few locations where only 1 or 2
phones are required.
The router has 2 FXS ports and can be used locally to an * box or via the
VPN to a remote * box.
The VPN built into the routers just works, and I have 1 user who has had 3
VPN circuits up and running now for 6 months solid.
Not bad in this day and age for an ADSL to stay functional for so long
without interruptions.

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Anton Krall
Sent: 05 March 2005 04:56
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file


The VPN approach might resolv a lot of nat issues I guess... Depending on
the scenario I guess.. You could put another * box inside the second nat and
interconnect using IAX, or if using a single phone, just use your setup, and
finally, if using 2 or more phones and cant put a second * box, well, the
vpn solution, I wonder how to do it if you have ATAs and nost softphone on
the second NATted LAN.. Well... In time I guess :)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rudolf
Ladyzhenskii
Sent: Viernes, 04 de Marzo de 2005 10:20 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk behind NAT -- SIP config file

Yes, only port 5060. If you do not forward 5060, you can not call this
 phone
from outside. Seem to work OK without other ports being forwarded.

 You mean on the remote sip phone firewall? What if there arem ore than
 1 sip phone on that network behidn that firewall?

Then you are in trouble. Asterisk only sees single public IP address. As far
as it concerns there is only single phone out there.
If you get multiple phones working, let me know.

Another option, I think, may be using VPN, but I have not tried that. Then
you can potentially have remote SIP phones to be on the virtual network.


 Don't you need to forward ports 1-2 for voice? Or does the sip
 phones just open up the ports from inside (by doing the in to out
 calls and keep alives)?


I have mot tried to sniff on the traffic in details. I think, other ports
are opened in responce to connection on port 5060. The only port listens at
is port 5060.

Rudolf

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RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file

[Anton Krall]

Good success story.. I'll keep in mind that router just in case.

Thx David. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J Carter
Sent: Sábado, 05 de Marzo de 2005 04:18 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file


I have used the Draytek 2600V router in a few locations where only 1 or 2
phones are required.
The router has 2 FXS ports and can be used locally to an * box or via the
VPN to a remote * box.
The VPN built into the routers just works, and I have 1 user who has had 3
VPN circuits up and running now for 6 months solid.
Not bad in this day and age for an ADSL to stay functional for so long
without interruptions.

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Anton Krall
Sent: 05 March 2005 04:56
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file


The VPN approach might resolv a lot of nat issues I guess... Depending on
the scenario I guess.. You could put another * box inside the second nat and
interconnect using IAX, or if using a single phone, just use your setup, and
finally, if using 2 or more phones and cant put a second * box, well, the
vpn solution, I wonder how to do it if you have ATAs and nost softphone on
the second NATted LAN.. Well... In time I guess :)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rudolf
Ladyzhenskii
Sent: Viernes, 04 de Marzo de 2005 10:20 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk behind NAT -- SIP config file

Yes, only port 5060. If you do not forward 5060, you can not call this
 phone
from outside. Seem to work OK without other ports being forwarded.

 You mean on the remote sip phone firewall? What if there arem ore than
 1 sip phone on that network behidn that firewall?

Then you are in trouble. Asterisk only sees single public IP address. As far
as it concerns there is only single phone out there.
If you get multiple phones working, let me know.

Another option, I think, may be using VPN, but I have not tried that. Then
you can potentially have remote SIP phones to be on the virtual network.


 Don't you need to forward ports 1-2 for voice? Or does the sip 
 phones just open up the ports from inside (by doing the in to out 
 calls and keep alives)?


I have mot tried to sniff on the traffic in details. I think, other ports
are opened in responce to connection on port 5060. The only port listens at
is port 5060.

Rudolf

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[Asterisk-Users] Asterisk behind NAT -- SIP config file

[Rudolf Ladyzhenskii]

Hi, all
This is the souktion that worked for me.
Here is my config again
PHONE  1 -- * BOX
|
 NAT/Firewall
|
|
  NAT/Firewall
   |
   |
 PHONE 2
Firewall on Asterisk end is Linux RH9 with iptables.
I have set it up to forward ports 5060, 1-2 to Asterisk.
Firewall at PHONE 2 end is an off-the-shelf router. Firewall was disabled 
and I port forwarded port 5060 to the phone.

Here is my sip.conf file: (PHONE1 is ext101, PHONE2 is ext102).
; SIP configuration file
[general]
port=5060
bindaddr=0.0.0.0
context=default
externip=my poublic ip
localnet=192.168.1.0/24

[ext101]
type=user
host=dynamic
secret=ext101
context=default
[ext101]
type=peer
secret=ext101
host=dynamic
context=default
callerid=Ext 101

[ext102]
type=user
nat=yes
host=dynamic
secret=ext102
context=default
canreinvite=no
[ext102]
type=peer
nat=yes
secret=ext102
host=dynamic
context=default
callerid=Ext 102
canreinvite=no

Hope it helps.

Rudolf
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RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file

[Anton Krall]

Why are the sip.conf extensions mentioned twice each?

Also, if you * box is behind another firewall, by forward ports 5060 and
1-2 and maybe 5004 from the firewall to the * box will that help on
the NAT issue? 

If phone 2 is behind another firewall, do you need to forward port 5060 only
to that phone? Or some other ports...?

I have read a lot of stuff about NAT and all the mayor flavors, still, Im
having some problems with nat and some networks..  I need to do more testing
using ethereal and other tools but I wanted to hear some basic thought on
the subject.

Thx! 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rudolf
Ladyzhenskii
Sent: Viernes, 04 de Marzo de 2005 08:41 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk behind NAT -- SIP config file

Hi, all

This is the souktion that worked for me.
Here is my config again

 PHONE  1 -- * BOX
 |
  NAT/Firewall
 |
 |
   NAT/Firewall
|
|
  PHONE 2

Firewall on Asterisk end is Linux RH9 with iptables.

I have set it up to forward ports 5060, 1-2 to Asterisk.

Firewall at PHONE 2 end is an off-the-shelf router. Firewall was disabled
and I port forwarded port 5060 to the phone.

Here is my sip.conf file: (PHONE1 is ext101, PHONE2 is ext102).
; SIP configuration file

[general]

port=5060

bindaddr=0.0.0.0

context=default

externip=my poublic ip

localnet=192.168.1.0/24



[ext101]

type=user

host=dynamic

secret=ext101

context=default

[ext101]

type=peer

secret=ext101

host=dynamic

context=default

callerid=Ext 101



[ext102]

type=user

nat=yes

host=dynamic

secret=ext102

context=default

canreinvite=no

[ext102]

type=peer

nat=yes

secret=ext102

host=dynamic

context=default

callerid=Ext 102

canreinvite=no



Hope it helps.



Rudolf

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Re: [Asterisk-Users] Asterisk behind NAT -- SIP config file

[Rudolf Ladyzhenskii]

Why are the sip.conf extensions mentioned twice each?
I am using Polycom SP300 phones. You have to separate 'user' and 'peer' part 
of it to get it working. Search the wiki for description of the problem.

Also, if you * box is behind another firewall, by forward ports 5060 and
1-2 and maybe 5004 from the firewall to the * box will that help 
on
the NAT issue?
You have to forward port 5060 so that phone from outside can register and 
call. And ports 1-2 do that voice can go through. Actual port ranfge 
is isn filr rtp.conf. 1-2 is  the default range

If phone 2 is behind another firewall, do you need to forward port 5060 
only
to that phone? Or some other ports...?
Yes, only port 5060. If you do not forward 5060, you can not call this phone 
from outside. Seem to work OK without other ports being forwarded.

Rudolf 

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RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file

[Anton Krall]

I am using Polycom SP300 phones. You have to separate 'user' and 'peer'
part of it to get it working. Search the wiki for description of the
problem.

Nice to know ... I don't own any of those but its good general knowledge.

You have to forward port 5060 so that phone from outside can register and
call. And ports 1-2 do that voice can go through. Actual port
ranfge is isn filr rtp.conf. 1-2 is  the default range

Ive done this on the firewall infront of our * box. 

Yes, only port 5060. If you do not forward 5060, you can not call this
phone 
from outside. Seem to work OK without other ports being forwarded.

You mean on the remote sip phone firewall? What if there arem ore than 1 sip
phone on that network behidn that firewall?

Don't you need to forward ports 1-2 for voice? Or does the sip
phones just open up the ports from inside (by doing the in to out calls and
keep alives)?


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Re: [Asterisk-Users] Asterisk behind NAT -- SIP config file

[Rudolf Ladyzhenskii]

Yes, only port 5060. If you do not forward 5060, you can not call this
phone
from outside. Seem to work OK without other ports being forwarded.
You mean on the remote sip phone firewall? What if there arem ore than 1 
sip
phone on that network behidn that firewall?
Then you are in trouble. Asterisk only sees single public IP address. As far 
as it concerns there is only single phone out there.
If you get multiple phones working, let me know.

Another option, I think, may be using VPN, but I have not tried that. Then 
you can potentially have remote SIP phones to be on the virtual network.

Don't you need to forward ports 1-2 for voice? Or does the sip
phones just open up the ports from inside (by doing the in to out calls 
and
keep alives)?

I have mot tried to sniff on the traffic in details. I think, other ports 
are opened in responce to connection on port 5060. The only port listens at 
is port 5060.

Rudolf 

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RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file

[Anton Krall]

The VPN approach might resolv a lot of nat issues I guess... Depending on
the scenario I guess.. You could put another * box inside the second nat and
interconnect using IAX, or if using a single phone, just use your setup, and
finally, if using 2 or more phones and cant put a second * box, well, the
vpn solution, I wonder how to do it if you have ATAs and nost softphone on
the second NATted LAN.. Well... In time I guess :) 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rudolf
Ladyzhenskii
Sent: Viernes, 04 de Marzo de 2005 10:20 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk behind NAT -- SIP config file

Yes, only port 5060. If you do not forward 5060, you can not call this
 phone
from outside. Seem to work OK without other ports being forwarded.

 You mean on the remote sip phone firewall? What if there arem ore than 
 1 sip phone on that network behidn that firewall?

Then you are in trouble. Asterisk only sees single public IP address. As far
as it concerns there is only single phone out there.
If you get multiple phones working, let me know.

Another option, I think, may be using VPN, but I have not tried that. Then
you can potentially have remote SIP phones to be on the virtual network.


 Don't you need to forward ports 1-2 for voice? Or does the sip 
 phones just open up the ports from inside (by doing the in to out 
 calls and keep alives)?


I have mot tried to sniff on the traffic in details. I think, other ports 
are opened in responce to connection on port 5060. The only port listens at 
is port 5060.

Rudolf 

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[Asterisk-Users] Asterisk Behind NAT

[sammy ominsky]

Hi all,
I've done quite a bit of reading, and I see that it's going to be 
difficult, but as a last-ditch effort before implementing a suggestion 
I don't like at all, I figured I'd ask...

Has anyone successfully put an asterisk box on an internal network 
behind a NAT device and been able to connect with SIP from outside?  
The real point behind all this is to implement QoS for the voice 
traffic, and putting a third box in front of the asterisk and NAT boxes 
has been deemed too expensive.

Currently, asterisk has a public IP, as does the NAT box behind which 
all the office machines sit.  If it can be done, the NAT box would be 
the best place to do the QoS, so why not ask, right?

Alternatively, I'm open to any suggestions that would work.  I've been 
handed this challenge on my first day on a new job... :/

Thanks,
---sambo
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RE: [Asterisk-Users] Asterisk Behind NAT

[C. Tomlinson]

I know this is possible using IAX easily, although I guess that is not an
option for you.

I have no firsthand experience, but believe some have got it working via
careful setup e.g noreinvites and other things.

If you setup a linux router, you could maybe have a separate DMZ to the *
box, but still use QoS?

Hope someone else can help you.

Best of luck with the new job.

C

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of sammy ominsky
Sent: 28 February 2005 21:04
To: Asterisk Users
Subject: [Asterisk-Users] Asterisk Behind NAT

Hi all,

I've done quite a bit of reading, and I see that it's going to be 
difficult, but as a last-ditch effort before implementing a suggestion 
I don't like at all, I figured I'd ask...

Has anyone successfully put an asterisk box on an internal network 
behind a NAT device and been able to connect with SIP from outside?  
The real point behind all this is to implement QoS for the voice 
traffic, and putting a third box in front of the asterisk and NAT boxes 
has been deemed too expensive.

Currently, asterisk has a public IP, as does the NAT box behind which 
all the office machines sit.  If it can be done, the NAT box would be 
the best place to do the QoS, so why not ask, right?

Alternatively, I'm open to any suggestions that would work.  I've been 
handed this challenge on my first day on a new job... :/

Thanks,

---sambo

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Re: [Asterisk-Users] Asterisk Behind NAT

[Carlos Chavez]

On Mon, 28 Feb 2005 16:03:30 -0500, sammy ominsky wrote
 Hi all,
 
 I've done quite a bit of reading, and I see that it's going to be 
 difficult, but as a last-ditch effort before implementing a 
 suggestion I don't like at all, I figured I'd ask...
 
 Has anyone successfully put an asterisk box on an internal network 
 behind a NAT device and been able to connect with SIP from outside?  
 The real point behind all this is to implement QoS for the voice 
 traffic, and putting a third box in front of the asterisk and NAT 
 boxes has been deemed too expensive.
 
 Currently, asterisk has a public IP, as does the NAT box behind 
 which all the office machines sit.  If it can be done, the NAT box 
 would be the best place to do the QoS, so why not ask, right?
 
 I have serveral servers running this way.

 Asterisk can be behind a NAT as long as you redirect all the necessary
ports to the server from the gateway.  The best way to do that is to put the *
server on the DMZ, but if that is not possible just redirect the ports needed
for SIP, IAX2 and RTP to the server.

 In the sip.conf file make sure you edit the externip and localnet options
to reflect your configuration.  Externip should be the external public IP.
 

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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RE: [Asterisk-Users] Asterisk Behind NAT

[Bill Seddon]

Title: [Asterisk-Users] Asterisk Behind NAT






Alternatively, I'm open to any suggestions that would 
work

Like you I read about and NAT and the problems. 
After a few days unsuccessful battling I gave up. Instead of using SIP 
directly, we've taken SIP numbers with a VoIP service provider and receive calls 
using IAX from the VoIP provider.

I guess you could do the same yourself: have an 
instance of Asterisk outside your firewall holding just SIP definitions and a 
simple dialplan to direct calls to and an Asterisk instance within a firewall 
using IAX that has a complete dialplan. I'm sure the VoIP providers that 
offer SIP-IAX and IAX-SIP, such as the one we use, are doing more and 
that there are some gotchas. But its an idea.

Bill Seddon


From: [EMAIL PROTECTED] on 
behalf of sammy ominskySent: Mon 2/28/2005 9:03 PMTo: 
Asterisk UsersSubject: [Asterisk-Users] Asterisk Behind 
NAT

Hi all,I've done quite a bit of reading, and I see that 
it's going to bedifficult, but as a last-ditch effort before implementing a 
suggestionI don't like at all, I figured I'd ask...Has anyone 
successfully put an asterisk box on an internal networkbehind a NAT device 
and been able to connect with SIP from outside?The real point behind 
all this is to implement QoS for the voicetraffic, and putting a third box 
in front of the asterisk and NAT boxeshas been deemed "too 
expensive".Currently, asterisk has a public IP, as does the NAT box 
behind whichall the office machines sit. If it can be done, the NAT 
box would bethe best place to do the QoS, so why not ask, 
right?Alternatively, I'm open to any suggestions that would work. 
I've beenhanded this challenge on my first day on a new job... 
:/Thanks,---sambo___Asterisk-Users 
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Re: [Asterisk-Users] Asterisk Behind NAT

[Steve Clark]

sammy ominsky wrote:
Hi all,
I've done quite a bit of reading, and I see that it's going to be 
difficult, but as a last-ditch effort before implementing a suggestion 
I don't like at all, I figured I'd ask...

Has anyone successfully put an asterisk box on an internal network 
behind a NAT device and been able to connect with SIP from outside?  
The real point behind all this is to implement QoS for the voice 
traffic, and putting a third box in front of the asterisk and NAT boxes 
has been deemed too expensive.

Currently, asterisk has a public IP, as does the NAT box behind which 
all the office machines sit.  If it can be done, the NAT box would be 
the best place to do the QoS, so why not ask, right?

Alternatively, I'm open to any suggestions that would work.  I've been 
handed this challenge on my first day on a new job... :/

Thanks,
---sambo
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What is the nat box? Linux, BSD, etc.
Steve
--
They that give up essential liberty to obtain temporary safety,
deserve neither liberty nor safety.  (Ben Franklin)
The course of history shows that as a government grows, liberty
decreases.  (Thomas Jefferson)
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Re: [Asterisk-Users] Asterisk Behind NAT

[rudolfl]

Hi,

I am working on exact same problem now and open to any suggestions.

So far I :
1. Made my NAT device to forward port 5060 to Asterisk server. 
2. Added line 'nat=yes' to the sip.conf for the user that is on outside.

At the moment, outside phone registers with Asterisk, but I can only place 
calls in
one direction and when cal is established, no sound path exist. Asterisk tries 
to
talk to the remote phone using its local IP address and this does not work. 

Let us know if you get anywhere and I will keep you posted too.
Rudolf


 sammy ominsky [EMAIL PROTECTED] wrote:
 
 Hi all,
 
 I've done quite a bit of reading, and I see that it's going to be 
 difficult, but as a last-ditch effort before implementing a suggestion 
 I don't like at all, I figured I'd ask...
 
 Has anyone successfully put an asterisk box on an internal network 
 behind a NAT device and been able to connect with SIP from outside?  
 The real point behind all this is to implement QoS for the voice 
 traffic, and putting a third box in front of the asterisk and NAT boxes 
 
 has been deemed too expensive.
 
 Currently, asterisk has a public IP, as does the NAT box behind which 
 all the office machines sit.  If it can be done, the NAT box would be 
 the best place to do the QoS, so why not ask, right?
 
 Alternatively, I'm open to any suggestions that would work.  I've been 
 handed this challenge on my first day on a new job... :/
 
 Thanks,
 
 ---sambo
 
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Re: [Asterisk-Users] Asterisk Behind NAT

[sammy ominsky]

On Feb 28, 2005, at 16:48, Steve Clark wrote:
What is the nat box? Linux, BSD, etc.
Linux.  Gibraltar firewall.
http://www.gibraltar.at/
---sambo
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Re: [Asterisk-Users] Asterisk Behind NAT

[Steve Rawlings]

- Original Message - 
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, February 28, 2005 9:50 PM
Subject: Re: [Asterisk-Users] Asterisk Behind NAT


Hi,
I am working on exact same problem now and open to any suggestions.
So far I :
1. Made my NAT device to forward port 5060 to Asterisk server.
2. Added line 'nat=yes' to the sip.conf for the user that is on outside.
At the moment, outside phone registers with Asterisk, but I can only place 
calls in
one direction and when cal is established, no sound path exist. Asterisk 
tries to
talk to the remote phone using its local IP address and this does not work.

Let us know if you get anywhere and I will keep you posted too.
Rudolf

Check rtp.conf and set to your nat device to forward these ports also to 
your * server, maybe reduce the number of ports too.

Steve 

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Re: [Asterisk-Users] Asterisk Behind NAT

[Steve Clark]

[EMAIL PROTECTED] wrote:
Hi,
I am working on exact same problem now and open to any suggestions.
So far I :
1. Made my NAT device to forward port 5060 to Asterisk server. 
2. Added line 'nat=yes' to the sip.conf for the user that is on outside.

At the moment, outside phone registers with Asterisk, but I can only place calls in
one direction and when cal is established, no sound path exist. Asterisk tries to
talk to the remote phone using its local IP address and this does not work. 

Let us know if you get anywhere and I will keep you posted too.
Rudolf

sammy ominsky [EMAIL PROTECTED] wrote:
Hi all,
I've done quite a bit of reading, and I see that it's going to be 
difficult, but as a last-ditch effort before implementing a suggestion 
I don't like at all, I figured I'd ask...

Has anyone successfully put an asterisk box on an internal network 
behind a NAT device and been able to connect with SIP from outside?  
The real point behind all this is to implement QoS for the voice 
traffic, and putting a third box in front of the asterisk and NAT boxes 

has been deemed too expensive.
Currently, asterisk has a public IP, as does the NAT box behind which 
all the office machines sit.  If it can be done, the NAT box would be 
the best place to do the QoS, so why not ask, right?

Alternatively, I'm open to any suggestions that would work.  I've been 
handed this challenge on my first day on a new job... :/

Thanks,
---sambo
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I believe you will also have to set externip in sip.conf to you public ip 
address.
Then you have allow rtp packet thru the fw and have them natted without altering 
the ports. You should then be able to call out and have sound.
When you call in you should get answered but probably wont have sound until the 
inside phone starts sending rtp packets.

HTH,
Steve
--
They that give up essential liberty to obtain temporary safety,
deserve neither liberty nor safety.  (Ben Franklin)
The course of history shows that as a government grows, liberty
decreases.  (Thomas Jefferson)
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RE: [Asterisk-Users] Asterisk Behind NAT

[Tom Rymes]

This has already been mentioned, but I remembered this froma  little
while back (sorry forget the original poster):

Begin Quoted message
Thanks to Pau (the original person to pose the question on this list),
it's fixed.  The firewall was getting in the way.  I needed to open up
UDP ports 1 to 2 for RTP traffic.

See the following for more info:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20rtp.con
f
http://www.voip-info.org/wiki-Asterisk+firewall+rules

end Quote

Tom

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 [EMAIL PROTECTED]
 Sent: Monday, February 28, 2005 4:51 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Asterisk Behind NAT



 Hi,

 I am working on exact same problem now and open to any suggestions.

 So far I :
 1. Made my NAT device to forward port 5060 to Asterisk server.
 2. Added line 'nat=yes' to the sip.conf for the user that is
 on outside.

 At the moment, outside phone registers with Asterisk, but I
 can only place calls in one direction and when cal is
 established, no sound path exist. Asterisk tries to talk to
 the remote phone using its local IP address and this does not work.

 Let us know if you get anywhere and I will keep you posted too. Rudolf


  sammy ominsky [EMAIL PROTECTED] wrote:
 
  Hi all,
 
  I've done quite a bit of reading, and I see that it's going to be
  difficult, but as a last-ditch effort before implementing a
 suggestion
  I don't like at all, I figured I'd ask...
 
  Has anyone successfully put an asterisk box on an internal network
  behind a NAT device and been able to connect with SIP from
 outside?
  The real point behind all this is to implement QoS for the voice
  traffic, and putting a third box in front of the asterisk
 and NAT boxes
 
  has been deemed too expensive.
 
  Currently, asterisk has a public IP, as does the NAT box
 behind which
  all the office machines sit.  If it can be done, the NAT
 box would be
  the best place to do the QoS, so why not ask, right?
 
  Alternatively, I'm open to any suggestions that would work.
  I've been
  handed this challenge on my first day on a new job... :/
 
  Thanks,
 
  ---sambo
 
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Re: [Asterisk-Users] Asterisk behind NAT How to do it.

[Balaji NJL]

Hi All,

i just applied this patch. i need to test whether its
working. Can someone
connect to my server and leave me a vm at extension
2000.

Server : ojoobala.com

Phone
Extension : 2005
pwd   : mytest
auth: md5.

pl leave a vm on extension 2000.

thanks a lot,
-B
- Original Message - 
From: listas iPfone [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 09, 2003 2:10 AM
Subject: Re: [Asterisk-Users] Asterisk behind NAT 
How to do it.


 Hi

 The version 1.260 of chan_sip.c already have that
patch?:


http://bugs.digium.com/file_download.php?file_id=430type=bug

 thanks!

 Miklos


 - Original Message - 
 From: Leif Madsen [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Friday, November 28, 2003 2:10 AM
 Subject: [Asterisk-Users] Asterisk behind NAT  How
to do it.


  Thanks to ww and his patch on bug #104, I have
successfully implemented
  Asterisk behind NAT without using STUN or anything
crazy.  It's quite
  straight forward.
 
  Until this gets tested enough and put into CVS,
you will have to patch
  your chan_sip.c file to do this.  I'm sure within
the next few days this
  will get put merged into CVS if no one finds any
problems.
 
  I tried this on chan_sip.c version 1.249 (the
version the patch was
  written for) and the latest as of today 1.258. 
Both work great.
 
  Open ports 5060 and your RTP range (found in
/etc/asterisk/rtp.conf).
  Default is 1 - 2
 
  Forward ports 5060 and your RTP range to your
internal Asterisk box.
 
  For your sip.conf, you need to add three lines:
 
  ; sip.conf snippet
  [general]
  port=5060   ; make sure you
have this line :)
  inside_net=192.168.1.100; this is the
internal ip address of
  the;
  asterisk server
  inside_mask=255.255.255.0   ; internal ip
mask.  /24 as this example
  outside_addr=216.239.33.100 ; this can also be
a FQDN! ie.
  ; my.domain.com
  ; ... plus whatever else you have in your sip.conf
 
  Download the patch at:
 
http://bugs.digium.com/file_download.php?file_id=430type=bug
 
  Either update your Asterisk or verify you have at
least version 1.249 of
  chan_sip.c:
 
  cd /usr/src/asterisk/channels/
  cvs status chan_sip.c
 
 
===
  File: chan_sip.cStatus: Locally Modified
 
 Working revision:1.258
 Repository revision: 1.258
  /usr/cvsroot/asterisk/channels/chan_sip.c,v
 
  While in pwd /usr/src/asterisk/channels/
  patch -p0  /path/to/patch
 
  Nothing should fail.
 
  cd /usr/src/asterisk/
  make
  cp /usr/src/asterisk/channels/chan_sip.so
/usr/lib/asterisk/modules/
 
  Restart your Asterisk and try it.  If you want to
call a NAT'd Asterisk
  box, my Free World Dialup number is 18924. 
Currently online.
 
  -- 
  Leif Madsen [EMAIL PROTECTED]
  http://www.hacklocalhost.com
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Re: [Asterisk-Users] Asterisk behind NAT How to do it.

[Balaji NJL]

i like the idea of not requiring to open 1 ports
in the firewall.

Do i need to change rtf.conf to from 1 - 2 to
16384 and 16394.

thanks,
-B 
- Original Message - 
From: Craig Waddington [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 27, 2003 3:43 AM
Subject: RE: [Asterisk-Users] Asterisk behind NAT 
How to do it.


 Hi
 
 I have SIP working on NAT using X-lite phones. 
 
 On my Cisco 827H ADSL router I forwarded ports 5060,
16394, 16384 to my
 * (10.1.0.0).
 
 16394,16384 being RTP.
 
 In X-lite set the RTP port to use 16394 instead of
the default 8000.
 
 Works great over the internet. Didn't need patches
or anything else.
 
 I hope that helps you.
 
 -C
 
 
 www.ntfs.org
 
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
Behalf Of Balaji NJL
 Sent: 27 December 2003 08:34
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Asterisk behind NAT 
How to do it.
 
 Hi All,
 
 i tried to apply this patch and i got the following
 error. The chan_sip.c
 version i hv is 1.265
 
 hv any one tried this patch on this latest chan_sip
 version.
 
 thanks,
 -B
 
 chan_sip.o: In function `load_module':
 chan_sip.o(.text+0x15ebf): undefined reference to
 `ast_rtp_proto_register'
 chan_sip.o(.text+0x15ee0): undefined reference to
 `ast_register_application'
 chan_sip.o: In function `delete_users':
 chan_sip.o(.text+0x15fc1): undefined reference to
 `ast_free_ha'
 chan_sip.o(.text+0x1604d): undefined reference to
 `ast_sched_del'
 chan_sip.o: In function `prune_peers':
 chan_sip.o(.text+0x16167): undefined reference to
 `ast_sched_del'
 chan_sip.o(.text+0x1618d): undefined reference to
 `ast_sched_del'
 chan_sip.o: In function `unload_module':
 chan_sip.o(.text+0x162bd): undefined reference to
 `ast_channel_unregister'
 chan_sip.o(.text+0x162ce): undefined reference to
 `ast_unregister_application'
 chan_sip.o(.text+0x16337): undefined reference to
 `ast_softhangup'
 chan_sip.o(.text+0x1636c): undefined reference to
 `ast_log'
 chan_sip.o(.text+0x163ab): undefined reference to
 `pthread_cancel'
 chan_sip.o(.text+0x163be): undefined reference to
 `pthread_kill'
 chan_sip.o(.text+0x163d1): undefined reference to
 `pthread_join'
 chan_sip.o(.text+0x16418): undefined reference to
 `ast_log'
 chan_sip.o(.text+0x164b8): undefined reference to
 `ast_log'
 collect2: ld returned 1 exit status
 make: *** [chan_sip.so] Error 1
 
 - Original Message - 
 From: listas iPfone [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Tuesday, December 09, 2003 2:10 AM
 Subject: Re: [Asterisk-Users] Asterisk behind NAT 
 How to do it.
 
 
  Hi
 
  The version 1.260 of chan_sip.c already have that
 patch?:
 
 

http://bugs.digium.com/file_download.php?file_id=430type=bug
 
  thanks!
 
  Miklos
 
 
  - Original Message - 
  From: Leif Madsen [EMAIL PROTECTED]
  To: [EMAIL PROTECTED]
  Sent: Friday, November 28, 2003 2:10 AM
  Subject: [Asterisk-Users] Asterisk behind NAT 
How
 to do it.
 
 
   Thanks to ww and his patch on bug #104, I have
 successfully implemented
   Asterisk behind NAT without using STUN or
anything
 crazy.  It's quite
   straight forward.
  
   Until this gets tested enough and put into CVS,
 you will have to patch
   your chan_sip.c file to do this.  I'm sure
within
 the next few days this
   will get put merged into CVS if no one finds any
 problems.
  
   I tried this on chan_sip.c version 1.249 (the
 version the patch was
   written for) and the latest as of today 1.258. 
 Both work great.
  
   Open ports 5060 and your RTP range (found in
 /etc/asterisk/rtp.conf).
   Default is 1 - 2
  
   Forward ports 5060 and your RTP range to your
 internal Asterisk box.
  
   For your sip.conf, you need to add three lines:
  
   ; sip.conf snippet
   [general]
   port=5060   ; make sure you
 have this line :)
   inside_net=192.168.1.100; this is the
 internal ip address of
   the;
   asterisk server
   inside_mask=255.255.255.0   ; internal ip
 mask.  /24 as this example
   outside_addr=216.239.33.100 ; this can also
be
 a FQDN! ie.
   ; my.domain.com
   ; ... plus whatever else you have in your
sip.conf
  
   Download the patch at:
  

http://bugs.digium.com/file_download.php?file_id=430type=bug
  
   Either update your Asterisk or verify you have
at
 least version 1.249 of
   chan_sip.c:
  
   cd /usr/src/asterisk/channels/
   cvs status chan_sip.c
  
  

===
   File: chan_sip.cStatus: Locally Modified
  
  Working revision:1.258
  Repository revision: 1.258
   /usr/cvsroot/asterisk/channels/chan_sip.c,v
  
   While in pwd /usr/src/asterisk/channels/
   patch -p0  /path/to/patch
  
   Nothing should fail.
  
   cd /usr/src/asterisk/
   make
   cp /usr/src/asterisk/channels/chan_sip.so
 /usr/lib/asterisk/modules/
  
   Restart your Asterisk and try

RE: [Asterisk-Users] Asterisk behind NAT How to do it.

[Craig Waddington]

Balaji.

I just left rtf.conf at default. Though I guess it wouldn't hurt to
change it to test.

Does it currently work for you with the settings I provided?

Craig.


www.ntfs.org


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Balaji NJL
Sent: 11 January 2004 10:35
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk behind NAT  How to do it.

i like the idea of not requiring to open 1 ports
in the firewall.

Do i need to change rtf.conf to from 1 - 2 to
16384 and 16394.

thanks,
-B 
- Original Message - 
From: Craig Waddington [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 27, 2003 3:43 AM
Subject: RE: [Asterisk-Users] Asterisk behind NAT 
How to do it.


 Hi
 
 I have SIP working on NAT using X-lite phones. 
 
 On my Cisco 827H ADSL router I forwarded ports 5060,
16394, 16384 to my
 * (10.1.0.0).
 
 16394,16384 being RTP.
 
 In X-lite set the RTP port to use 16394 instead of
the default 8000.
 
 Works great over the internet. Didn't need patches
or anything else.
 
 I hope that helps you.
 
 -C
 
 
 www.ntfs.org
 
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
Behalf Of Balaji NJL
 Sent: 27 December 2003 08:34
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Asterisk behind NAT 
How to do it.
 
 Hi All,
 
 i tried to apply this patch and i got the following
 error. The chan_sip.c
 version i hv is 1.265
 
 hv any one tried this patch on this latest chan_sip
 version.
 
 thanks,
 -B
 
 chan_sip.o: In function `load_module':
 chan_sip.o(.text+0x15ebf): undefined reference to
 `ast_rtp_proto_register'
 chan_sip.o(.text+0x15ee0): undefined reference to
 `ast_register_application'
 chan_sip.o: In function `delete_users':
 chan_sip.o(.text+0x15fc1): undefined reference to
 `ast_free_ha'
 chan_sip.o(.text+0x1604d): undefined reference to
 `ast_sched_del'
 chan_sip.o: In function `prune_peers':
 chan_sip.o(.text+0x16167): undefined reference to
 `ast_sched_del'
 chan_sip.o(.text+0x1618d): undefined reference to
 `ast_sched_del'
 chan_sip.o: In function `unload_module':
 chan_sip.o(.text+0x162bd): undefined reference to
 `ast_channel_unregister'
 chan_sip.o(.text+0x162ce): undefined reference to
 `ast_unregister_application'
 chan_sip.o(.text+0x16337): undefined reference to
 `ast_softhangup'
 chan_sip.o(.text+0x1636c): undefined reference to
 `ast_log'
 chan_sip.o(.text+0x163ab): undefined reference to
 `pthread_cancel'
 chan_sip.o(.text+0x163be): undefined reference to
 `pthread_kill'
 chan_sip.o(.text+0x163d1): undefined reference to
 `pthread_join'
 chan_sip.o(.text+0x16418): undefined reference to
 `ast_log'
 chan_sip.o(.text+0x164b8): undefined reference to
 `ast_log'
 collect2: ld returned 1 exit status
 make: *** [chan_sip.so] Error 1
 
 - Original Message - 
 From: listas iPfone [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Tuesday, December 09, 2003 2:10 AM
 Subject: Re: [Asterisk-Users] Asterisk behind NAT 
 How to do it.
 
 
  Hi
 
  The version 1.260 of chan_sip.c already have that
 patch?:
 
 

http://bugs.digium.com/file_download.php?file_id=430type=bug
 
  thanks!
 
  Miklos
 
 
  - Original Message - 
  From: Leif Madsen [EMAIL PROTECTED]
  To: [EMAIL PROTECTED]
  Sent: Friday, November 28, 2003 2:10 AM
  Subject: [Asterisk-Users] Asterisk behind NAT 
How
 to do it.
 
 
   Thanks to ww and his patch on bug #104, I have
 successfully implemented
   Asterisk behind NAT without using STUN or
anything
 crazy.  It's quite
   straight forward.
  
   Until this gets tested enough and put into CVS,
 you will have to patch
   your chan_sip.c file to do this.  I'm sure
within
 the next few days this
   will get put merged into CVS if no one finds any
 problems.
  
   I tried this on chan_sip.c version 1.249 (the
 version the patch was
   written for) and the latest as of today 1.258. 
 Both work great.
  
   Open ports 5060 and your RTP range (found in
 /etc/asterisk/rtp.conf).
   Default is 1 - 2
  
   Forward ports 5060 and your RTP range to your
 internal Asterisk box.
  
   For your sip.conf, you need to add three lines:
  
   ; sip.conf snippet
   [general]
   port=5060   ; make sure you
 have this line :)
   inside_net=192.168.1.100; this is the
 internal ip address of
   the;
   asterisk server
   inside_mask=255.255.255.0   ; internal ip
 mask.  /24 as this example
   outside_addr=216.239.33.100 ; this can also
be
 a FQDN! ie.
   ; my.domain.com
   ; ... plus whatever else you have in your
sip.conf
  
   Download the patch at:
  

http://bugs.digium.com/file_download.php?file_id=430type=bug
  
   Either update your Asterisk or verify you have
at
 least version 1.249 of
   chan_sip.c:
  
   cd /usr/src/asterisk/channels/
   cvs status chan_sip.c
  
  

===
   File: chan_sip.c

RE: [Asterisk-Users] Asterisk behind NAT How to do it.

[Craig Waddington]

Hi

I have SIP working on NAT using X-lite phones. 

On my Cisco 827H ADSL router I forwarded ports 5060, 16394, 16384 to my
* (10.1.0.0).

16394,16384 being RTP.

In X-lite set the RTP port to use 16394 instead of the default 8000.

Works great over the internet. Didn't need patches or anything else.

I hope that helps you.

-C


www.ntfs.org




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Balaji NJL
Sent: 27 December 2003 08:34
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk behind NAT  How to do it.

Hi All,

i tried to apply this patch and i got the following
error. The chan_sip.c
version i hv is 1.265

hv any one tried this patch on this latest chan_sip
version.

thanks,
-B

chan_sip.o: In function `load_module':
chan_sip.o(.text+0x15ebf): undefined reference to
`ast_rtp_proto_register'
chan_sip.o(.text+0x15ee0): undefined reference to
`ast_register_application'
chan_sip.o: In function `delete_users':
chan_sip.o(.text+0x15fc1): undefined reference to
`ast_free_ha'
chan_sip.o(.text+0x1604d): undefined reference to
`ast_sched_del'
chan_sip.o: In function `prune_peers':
chan_sip.o(.text+0x16167): undefined reference to
`ast_sched_del'
chan_sip.o(.text+0x1618d): undefined reference to
`ast_sched_del'
chan_sip.o: In function `unload_module':
chan_sip.o(.text+0x162bd): undefined reference to
`ast_channel_unregister'
chan_sip.o(.text+0x162ce): undefined reference to
`ast_unregister_application'
chan_sip.o(.text+0x16337): undefined reference to
`ast_softhangup'
chan_sip.o(.text+0x1636c): undefined reference to
`ast_log'
chan_sip.o(.text+0x163ab): undefined reference to
`pthread_cancel'
chan_sip.o(.text+0x163be): undefined reference to
`pthread_kill'
chan_sip.o(.text+0x163d1): undefined reference to
`pthread_join'
chan_sip.o(.text+0x16418): undefined reference to
`ast_log'
chan_sip.o(.text+0x164b8): undefined reference to
`ast_log'
collect2: ld returned 1 exit status
make: *** [chan_sip.so] Error 1

- Original Message - 
From: listas iPfone [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 09, 2003 2:10 AM
Subject: Re: [Asterisk-Users] Asterisk behind NAT 
How to do it.


 Hi

 The version 1.260 of chan_sip.c already have that
patch?:


http://bugs.digium.com/file_download.php?file_id=430type=bug

 thanks!

 Miklos


 - Original Message - 
 From: Leif Madsen [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Friday, November 28, 2003 2:10 AM
 Subject: [Asterisk-Users] Asterisk behind NAT  How
to do it.


  Thanks to ww and his patch on bug #104, I have
successfully implemented
  Asterisk behind NAT without using STUN or anything
crazy.  It's quite
  straight forward.
 
  Until this gets tested enough and put into CVS,
you will have to patch
  your chan_sip.c file to do this.  I'm sure within
the next few days this
  will get put merged into CVS if no one finds any
problems.
 
  I tried this on chan_sip.c version 1.249 (the
version the patch was
  written for) and the latest as of today 1.258. 
Both work great.
 
  Open ports 5060 and your RTP range (found in
/etc/asterisk/rtp.conf).
  Default is 1 - 2
 
  Forward ports 5060 and your RTP range to your
internal Asterisk box.
 
  For your sip.conf, you need to add three lines:
 
  ; sip.conf snippet
  [general]
  port=5060   ; make sure you
have this line :)
  inside_net=192.168.1.100; this is the
internal ip address of
  the;
  asterisk server
  inside_mask=255.255.255.0   ; internal ip
mask.  /24 as this example
  outside_addr=216.239.33.100 ; this can also be
a FQDN! ie.
  ; my.domain.com
  ; ... plus whatever else you have in your sip.conf
 
  Download the patch at:
 
http://bugs.digium.com/file_download.php?file_id=430type=bug
 
  Either update your Asterisk or verify you have at
least version 1.249 of
  chan_sip.c:
 
  cd /usr/src/asterisk/channels/
  cvs status chan_sip.c
 
 
===
  File: chan_sip.cStatus: Locally Modified
 
 Working revision:1.258
 Repository revision: 1.258
  /usr/cvsroot/asterisk/channels/chan_sip.c,v
 
  While in pwd /usr/src/asterisk/channels/
  patch -p0  /path/to/patch
 
  Nothing should fail.
 
  cd /usr/src/asterisk/
  make
  cp /usr/src/asterisk/channels/chan_sip.so
/usr/lib/asterisk/modules/
 
  Restart your Asterisk and try it.  If you want to
call a NAT'd Asterisk
  box, my Free World Dialup number is 18924. 
Currently online.
 
  -- 
  Leif Madsen [EMAIL PROTECTED]
  http://www.hacklocalhost.com
  ___
  Asterisk-Users mailing list
  [EMAIL PROTECTED]
 
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]

http://lists.digium.com/mailman/listinfo/asterisk-users


__
Do you Yahoo

Re: [Asterisk-Users] Asterisk behind NAT How to do it.

[Balaji NJL]

thats cool. i ll try that too. Whats ur * version.

if thats the case what is this patch for. Is bug 104
already approved and in
production.

-B

- Original Message - 
From: Craig Waddington [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 27, 2003 3:43 AM
Subject: RE: [Asterisk-Users] Asterisk behind NAT 
How to do it.


 Hi

 I have SIP working on NAT using X-lite phones.

 On my Cisco 827H ADSL router I forwarded ports 5060,
16394, 16384 to my
 * (10.1.0.0).

 16394,16384 being RTP.

 In X-lite set the RTP port to use 16394 instead of
the default 8000.

 Works great over the internet. Didn't need patches
or anything else.

 I hope that helps you.

 -C


 www.ntfs.org




 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
Behalf Of Balaji NJL
 Sent: 27 December 2003 08:34
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Asterisk behind NAT 
How to do it.

 Hi All,

 i tried to apply this patch and i got the following
 error. The chan_sip.c
 version i hv is 1.265

 hv any one tried this patch on this latest chan_sip
 version.

 thanks,
 -B

 chan_sip.o: In function `load_module':
 chan_sip.o(.text+0x15ebf): undefined reference to
 `ast_rtp_proto_register'
 chan_sip.o(.text+0x15ee0): undefined reference to
 `ast_register_application'
 chan_sip.o: In function `delete_users':
 chan_sip.o(.text+0x15fc1): undefined reference to
 `ast_free_ha'
 chan_sip.o(.text+0x1604d): undefined reference to
 `ast_sched_del'
 chan_sip.o: In function `prune_peers':
 chan_sip.o(.text+0x16167): undefined reference to
 `ast_sched_del'
 chan_sip.o(.text+0x1618d): undefined reference to
 `ast_sched_del'
 chan_sip.o: In function `unload_module':
 chan_sip.o(.text+0x162bd): undefined reference to
 `ast_channel_unregister'
 chan_sip.o(.text+0x162ce): undefined reference to
 `ast_unregister_application'
 chan_sip.o(.text+0x16337): undefined reference to
 `ast_softhangup'
 chan_sip.o(.text+0x1636c): undefined reference to
 `ast_log'
 chan_sip.o(.text+0x163ab): undefined reference to
 `pthread_cancel'
 chan_sip.o(.text+0x163be): undefined reference to
 `pthread_kill'
 chan_sip.o(.text+0x163d1): undefined reference to
 `pthread_join'
 chan_sip.o(.text+0x16418): undefined reference to
 `ast_log'
 chan_sip.o(.text+0x164b8): undefined reference to
 `ast_log'
 collect2: ld returned 1 exit status
 make: *** [chan_sip.so] Error 1

 - Original Message - 
 From: listas iPfone [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Tuesday, December 09, 2003 2:10 AM
 Subject: Re: [Asterisk-Users] Asterisk behind NAT 
 How to do it.


  Hi
 
  The version 1.260 of chan_sip.c already have that
 patch?:
 
 

http://bugs.digium.com/file_download.php?file_id=430type=bug
 
  thanks!
 
  Miklos
 
 
  - Original Message - 
  From: Leif Madsen [EMAIL PROTECTED]
  To: [EMAIL PROTECTED]
  Sent: Friday, November 28, 2003 2:10 AM
  Subject: [Asterisk-Users] Asterisk behind NAT 
How
 to do it.
 
 
   Thanks to ww and his patch on bug #104, I have
 successfully implemented
   Asterisk behind NAT without using STUN or
anything
 crazy.  It's quite
   straight forward.
  
   Until this gets tested enough and put into CVS,
 you will have to patch
   your chan_sip.c file to do this.  I'm sure
within
 the next few days this
   will get put merged into CVS if no one finds any
 problems.
  
   I tried this on chan_sip.c version 1.249 (the
 version the patch was
   written for) and the latest as of today 1.258.
 Both work great.
  
   Open ports 5060 and your RTP range (found in
 /etc/asterisk/rtp.conf).
   Default is 1 - 2
  
   Forward ports 5060 and your RTP range to your
 internal Asterisk box.
  
   For your sip.conf, you need to add three lines:
  
   ; sip.conf snippet
   [general]
   port=5060   ; make sure you
 have this line :)
   inside_net=192.168.1.100; this is the
 internal ip address of
   the;
   asterisk server
   inside_mask=255.255.255.0   ; internal ip
 mask.  /24 as this example
   outside_addr=216.239.33.100 ; this can also
be
 a FQDN! ie.
   ; my.domain.com
   ; ... plus whatever else you have in your
sip.conf
  
   Download the patch at:
  

http://bugs.digium.com/file_download.php?file_id=430type=bug
  
   Either update your Asterisk or verify you have
at
 least version 1.249 of
   chan_sip.c:
  
   cd /usr/src/asterisk/channels/
   cvs status chan_sip.c
  
  

===
   File: chan_sip.cStatus: Locally Modified
  
  Working revision:1.258
  Repository revision: 1.258
   /usr/cvsroot/asterisk/channels/chan_sip.c,v
  
   While in pwd /usr/src/asterisk/channels/
   patch -p0  /path/to/patch
  
   Nothing should fail.
  
   cd /usr/src/asterisk/
   make
   cp /usr/src/asterisk/channels/chan_sip.so
 /usr/lib/asterisk/modules/
  
   Restart your Asterisk and try it.  If you want
to
 call a NAT'd

[Asterisk-Users] asterisk behind NAT

[Patrick Cantwell]

I know this issue has been covered with at least 2 different patches, and
probably a dozen different discussions, however I'm a bit unclear as to what
my options are.

I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall
doing 1:1 NAT for machines behind the firewall.  My asterisk box is one of
these machines, and I'd like to allow foreign SIP clients
(softphones/hardware phones) to register to my Asterisk box -WITHOUT-
breaking internal connectivity.

A brief example of my setup works like this:

asterisk box - openbsd firewall --- internet
(192.168.250.7)|
  --
-- other internal networks  (192.168.0.0/16)

The OpenBSD firewall provides a 1:1 NAT mapping for the asterisk box to
216.254.114.221 so ports/port forwarding is a non issue.

I also have several other internal subnets hanging off of the OpenBSD
firewall, all using 192.168.0.0/16 address space, and I do have some
hardware/software clients running internally.

I've also noticed that in newer CVS versions, there are provisions for
'externip', but nothing for internal net/netmask, so I suspect this will
break my internal clients.

My question is, first off, do I need to apply a patch, and if so, which one?
Second, once I apply said patch, what options do I need to supply in
sip.conf?

I could also run something on the openbsd firewall (maybe a SIP proxy?),
I've seen references to 'STUN' but haven't found enough info on it to know
if it will help me.

Thanks,
Pat

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] asterisk behind NAT

[Brian West]

bug 104 on bugs.digium.com take a look at it.  Also your setup is EVIL

bkw

On Thu, 18 Dec 2003, Patrick Cantwell wrote:

 I know this issue has been covered with at least 2 different patches, and
 probably a dozen different discussions, however I'm a bit unclear as to what
 my options are.

 I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall
 doing 1:1 NAT for machines behind the firewall.  My asterisk box is one of
 these machines, and I'd like to allow foreign SIP clients
 (softphones/hardware phones) to register to my Asterisk box -WITHOUT-
 breaking internal connectivity.

 A brief example of my setup works like this:

 asterisk box - openbsd firewall --- internet
 (192.168.250.7)|
   --
 -- other internal networks  (192.168.0.0/16)

 The OpenBSD firewall provides a 1:1 NAT mapping for the asterisk box to
 216.254.114.221 so ports/port forwarding is a non issue.

 I also have several other internal subnets hanging off of the OpenBSD
 firewall, all using 192.168.0.0/16 address space, and I do have some
 hardware/software clients running internally.

 I've also noticed that in newer CVS versions, there are provisions for
 'externip', but nothing for internal net/netmask, so I suspect this will
 break my internal clients.

 My question is, first off, do I need to apply a patch, and if so, which one?
 Second, once I apply said patch, what options do I need to supply in
 sip.conf?

 I could also run something on the openbsd firewall (maybe a SIP proxy?),
 I've seen references to 'STUN' but haven't found enough info on it to know
 if it will help me.

 Thanks,
 Pat

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk behind NAT How to do it.

[Leif Madsen]

On Tue, 2003-12-09 at 05:10, listas iPfone wrote:
 Hi
 
 The version 1.260 of chan_sip.c already have that patch?:
 
 http://bugs.digium.com/file_download.php?file_id=430type=bug

That link didn't work for me, but the NAT patch has not been put into
CVS yet.  It needs to be TESTED more, so if you guys want this added,
then you need to go and apply the patch and comment on it in the bug
tracker.  If something doesn't work, SAY SO!

Thanks :)

-- 
Leif Madsen [EMAIL PROTECTED]
http://www.hacklocalhost.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Asterisk behind NAT How to do it.

[David J Carter]

Hi

I have applied the patch, I can register a Grandstream 100 from another
internet connection but I get no audio and a timeout line drop after 5
seconds.
If I call my SipPhone number 17476691936 I hear my welcome message and again
the line times out and drops after 5 seconds.
I notice that the connection is trying to do a native bridge even though I
have reinvite=no  canreinvite=no in the sip.conf.
Any help would be welcome.

Regards

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen
Sent: 12 December 2003 08:24
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk behind NAT  How to do it.

On Tue, 2003-12-09 at 05:10, listas iPfone wrote:
 Hi

 The version 1.260 of chan_sip.c already have that patch?:

 http://bugs.digium.com/file_download.php?file_id=430type=bug

That link didn't work for me, but the NAT patch has not been put into
CVS yet.  It needs to be TESTED more, so if you guys want this added,
then you need to go and apply the patch and comment on it in the bug
tracker.  If something doesn't work, SAY SO!

Thanks :)

--
Leif Madsen [EMAIL PROTECTED]
http://www.hacklocalhost.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk behind NAT How to do it.

[listas iPfone]

Hi

The version 1.260 of chan_sip.c already have that patch?:

http://bugs.digium.com/file_download.php?file_id=430type=bug

thanks!

Miklos


- Original Message - 
From: Leif Madsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, November 28, 2003 2:10 AM
Subject: [Asterisk-Users] Asterisk behind NAT  How to do it.


 Thanks to ww and his patch on bug #104, I have successfully implemented
 Asterisk behind NAT without using STUN or anything crazy.  It's quite
 straight forward.
 
 Until this gets tested enough and put into CVS, you will have to patch
 your chan_sip.c file to do this.  I'm sure within the next few days this
 will get put merged into CVS if no one finds any problems.
 
 I tried this on chan_sip.c version 1.249 (the version the patch was
 written for) and the latest as of today 1.258.  Both work great.
 
 Open ports 5060 and your RTP range (found in /etc/asterisk/rtp.conf). 
 Default is 1 - 2
 
 Forward ports 5060 and your RTP range to your internal Asterisk box.
 
 For your sip.conf, you need to add three lines:
 
 ; sip.conf snippet
 [general]
 port=5060   ; make sure you have this line :)
 inside_net=192.168.1.100; this is the internal ip address of
 the;
 asterisk server
 inside_mask=255.255.255.0   ; internal ip mask.  /24 as this example
 outside_addr=216.239.33.100 ; this can also be a FQDN! ie.
 ; my.domain.com
 ; ... plus whatever else you have in your sip.conf
 
 Download the patch at:
 http://bugs.digium.com/file_download.php?file_id=430type=bug
 
 Either update your Asterisk or verify you have at least version 1.249 of
 chan_sip.c:
 
 cd /usr/src/asterisk/channels/
 cvs status chan_sip.c
 
 ===
 File: chan_sip.cStatus: Locally Modified
  
Working revision:1.258
Repository revision: 1.258  
 /usr/cvsroot/asterisk/channels/chan_sip.c,v
 
 While in pwd /usr/src/asterisk/channels/
 patch -p0  /path/to/patch
 
 Nothing should fail.
 
 cd /usr/src/asterisk/
 make
 cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/
 
 Restart your Asterisk and try it.  If you want to call a NAT'd Asterisk
 box, my Free World Dialup number is 18924.  Currently online.
 
 -- 
 Leif Madsen [EMAIL PROTECTED]
 http://www.hacklocalhost.com
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
___
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Re: [Asterisk-Users] Asterisk behind NAT How to do it.

[Leif Madsen]

On Wed, 2003-12-03 at 15:34, William Waites wrote:

   localnet= internal ip of * machine?
 
 localnet should be the internal network address not the internal
 ip address. i.e. if your asterisk server is 192.168.0.245, localnet
 should be 192.168.0.0

Agreed, I was wrong before :)

-- 
Leif Madsen [EMAIL PROTECTED]
http://www.hacklocalhost.com
___
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Re: [Asterisk-Users] Asterisk behind NAT How to do it.

[robert ivanc]

Arnold Ligtvoet wrote:

  Leif wrote:
  
  
Awesome!  Have you tried the newer patch / diff for 1.259 (which as of
right now is the newest chan_sip file).  If you goto bugs.digium.com and
login anonymously and jump to bug 104, then you can get the newest
patch.  Same instructions as before.

  
  
  

this patch seems to break my GS phones that are connecting to * via
NAT. The one before that works ok - 249 or something? They can't
connect anymore - get a Not Found error back.

Regards,

 Robert


  Installed the new patch, no errors here. Ran make and copied chan_sip.o. All
went fine.

  
  
I just updated it to test the new sip.conf structure which is

externip=
localnet=
localmask=

  
  
Updated my sip.conf to match these settings. The result seems to be better,
yesterday I noticed a slight delay in the setup of the audio channel, the
speaking clock would only start at the second word, this is now gone.

  
  
Still working great for me here!

BTW!   Can you login to the bug tracker and post a comment ?  Thanks!

  
  
I do have one strange issue. I have a test setup here which is very simple.
* server and one windows machine. * is connected via ISDN (chan_i4l) to my
home pbx. On my windows machine I installed Diax, SjPhone and SIPPS. The
strange thing I now have is that both skinny clients are able to receive
audio but not send any when I call an extension on my pbx (so external for
*). I first thought it was my mic, but diax is working fine.

I have already been looking at my sip.conf for the extensions but I'm not
sure if this is the problem. Anyway my sip.conf now is :
[general]
disallow=all; Disallow all codecs
allow=ulaw  ; Allow codecs in order of preference
allow=alaw
allow=ilbc
allow=gsm

; for fix 1.259
externip=212.238.144.173
localnet=192.168.0.100
localmask=255.255.255.0

[phone1]
type=friend
host=dynamic
defaultip=192.168.0.2
dtmfmode=inband
mailbox=1000 ; Mailbox for message waiting indicator
context=default
callerid="Me" 2124
;reinvite=no
;canreinvite=no
;nat=yes
;insecure=yes

I'll wait your reply for the one-way sound 'issue' (probably me!) before
posting to the bugtracker. Hopefully someone has some clue as to why my sip
clients are not able to send sound.

Thanks,
Arnold Ligtvoet.

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Re: [Asterisk-Users] Asterisk behind NAT How to do it.

[William Waites]

On Thu, Dec 04, 2003 at 07:56:56PM +0100, robert ivanc wrote:

 this patch seems to break my GS phones that are connecting to * via NAT. 
 The one before that works ok - 249 or something? They can't connect 
 anymore - get a Not Found error back.

That is very strange -- the *only* difference between those two versions
of the patch is the variable naming. Can you give me some more debugging
information? Some more information on your setup and perhaps a trace of
the SIP conversation? I don't have a GS phone to test with here.

Thanks,
-w
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 X No Word docs in email
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RE: [Asterisk-Users] Asterisk behind NAT How to do it.

[Leif Madsen]

On Tue, 2003-12-02 at 15:55, Arnold Ligtvoet wrote:

 Hi Leif,
 
 I tried the patch. Installed it exactly as described per your email. Thought
 that you might be interested that it works for me as well. Like a charm, I
 can finally call FWD numbers like 10001 and 612 (speaking clock demo).
 
 BTW: For anybody wanting to install this, if your version of chan_sip.c is
 older than the one described, first use 'cvs update -C
 asterisk/channels/chan_sip.c'.

Awesome!  Have you tried the newer patch / diff for 1.259 (which as of
right now is the newest chan_sip file).  If you goto bugs.digium.com and
login anonymously and jump to bug 104, then you can get the newest
patch.  Same instructions as before.

I just updated it to test the new sip.conf structure which is

externip=
localnet=
localmask=

Still working great for me here!

BTW!   Can you login to the bug tracker and post a comment ?  Thanks!

-- 
Leif Madsen [EMAIL PROTECTED]
http://www.hacklocalhost.com
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Re: [Asterisk-Users] Asterisk behind NAT How to do it.

[listas iPfone]

Hi!

I  need help to undestand the options:

 externip= static/ dynamic ip? can be a domain?
 localnet= internal ip of * machine?
 localmask= 255.255.255.0 ?

Thanks


- Original Message - 
From: Leif Madsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 03, 2003 7:25 AM
Subject: RE: [Asterisk-Users] Asterisk behind NAT  How to do it.


 On Tue, 2003-12-02 at 15:55, Arnold Ligtvoet wrote:

  Hi Leif,
 
  I tried the patch. Installed it exactly as described per your email.
Thought
  that you might be interested that it works for me as well. Like a charm,
I
  can finally call FWD numbers like 10001 and 612 (speaking clock demo).
 
  BTW: For anybody wanting to install this, if your version of chan_sip.c
is
  older than the one described, first use 'cvs update -C
  asterisk/channels/chan_sip.c'.

 Awesome!  Have you tried the newer patch / diff for 1.259 (which as of
 right now is the newest chan_sip file).  If you goto bugs.digium.com and
 login anonymously and jump to bug 104, then you can get the newest
 patch.  Same instructions as before.

 I just updated it to test the new sip.conf structure which is

 externip=
 localnet=
 localmask=

 Still working great for me here!

 BTW!   Can you login to the bug tracker and post a comment ?  Thanks!

 -- 
 Leif Madsen [EMAIL PROTECTED]
 http://www.hacklocalhost.com
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Re: [Asterisk-Users] Asterisk behind NAT How to do it.

[William Waites]

On Wed, Dec 03, 2003 at 05:47:59PM -0200, listas iPfone wrote:
 Hi!
 
 I  need help to undestand the options:
 

hello.

  externip= static/ dynamic ip? can be a domain?

externip can by an IP address or a domain. it uses gethostbyname(3)
in the code.

  localnet= internal ip of * machine?

localnet should be the internal network address not the internal
ip address. i.e. if your asterisk server is 192.168.0.245, localnet
should be 192.168.0.0

  localmask= 255.255.255.0 ?

that is correct. (unless you have a different netmasks of course)

cheers,
-w
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RE: [Asterisk-Users] Asterisk behind NAT How to do it.

[Arnold Ligtvoet]

Leif wrote:
 Awesome!  Have you tried the newer patch / diff for 1.259 (which as of
 right now is the newest chan_sip file).  If you goto bugs.digium.com and
 login anonymously and jump to bug 104, then you can get the newest
 patch.  Same instructions as before.

Installed the new patch, no errors here. Ran make and copied chan_sip.o. All
went fine.

 I just updated it to test the new sip.conf structure which is

 externip=
 localnet=
 localmask=

Updated my sip.conf to match these settings. The result seems to be better,
yesterday I noticed a slight delay in the setup of the audio channel, the
speaking clock would only start at the second word, this is now gone.

 Still working great for me here!

 BTW!   Can you login to the bug tracker and post a comment ?  Thanks!

I do have one strange issue. I have a test setup here which is very simple.
* server and one windows machine. * is connected via ISDN (chan_i4l) to my
home pbx. On my windows machine I installed Diax, SjPhone and SIPPS. The
strange thing I now have is that both skinny clients are able to receive
audio but not send any when I call an extension on my pbx (so external for
*). I first thought it was my mic, but diax is working fine.

I have already been looking at my sip.conf for the extensions but I'm not
sure if this is the problem. Anyway my sip.conf now is :
[general]
disallow=all; Disallow all codecs
allow=ulaw  ; Allow codecs in order of preference
allow=alaw
allow=ilbc
allow=gsm

; for fix 1.259
externip=212.238.144.173
localnet=192.168.0.100
localmask=255.255.255.0

[phone1]
type=friend
host=dynamic
defaultip=192.168.0.2
dtmfmode=inband
mailbox=1000 ; Mailbox for message waiting indicator
context=default
callerid=Me 2124
;reinvite=no
;canreinvite=no
;nat=yes
;insecure=yes

I'll wait your reply for the one-way sound 'issue' (probably me!) before
posting to the bugtracker. Hopefully someone has some clue as to why my sip
clients are not able to send sound.

Thanks,
Arnold Ligtvoet.

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RE: [Asterisk-Users] Asterisk behind NAT How to do it.

[Arnold Ligtvoet]

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen
 Sent: vrijdag 28 november 2003 5:11
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Asterisk behind NAT  How to do it.


 Thanks to ww and his patch on bug #104, I have successfully implemented
 Asterisk behind NAT without using STUN or anything crazy.  It's quite
 straight forward.

 Until this gets tested enough and put into CVS, you will have to patch
 your chan_sip.c file to do this.  I'm sure within the next few days this
 will get put merged into CVS if no one finds any problems.


Hi Leif,

I tried the patch. Installed it exactly as described per your email. Thought
that you might be interested that it works for me as well. Like a charm, I
can finally call FWD numbers like 10001 and 612 (speaking clock demo).

BTW: For anybody wanting to install this, if your version of chan_sip.c is
older than the one described, first use 'cvs update -C
asterisk/channels/chan_sip.c'.

Thanks,
Arnold Ligtvoet.

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[Asterisk-Users] Asterisk behind NAT How to do it.

[Leif Madsen]

Thanks to ww and his patch on bug #104, I have successfully implemented
Asterisk behind NAT without using STUN or anything crazy.  It's quite
straight forward.

Until this gets tested enough and put into CVS, you will have to patch
your chan_sip.c file to do this.  I'm sure within the next few days this
will get put merged into CVS if no one finds any problems.

I tried this on chan_sip.c version 1.249 (the version the patch was
written for) and the latest as of today 1.258.  Both work great.

Open ports 5060 and your RTP range (found in /etc/asterisk/rtp.conf). 
Default is 1 - 2

Forward ports 5060 and your RTP range to your internal Asterisk box.

For your sip.conf, you need to add three lines:

; sip.conf snippet
[general]
port=5060   ; make sure you have this line :)
inside_net=192.168.1.100; this is the internal ip address of
the;
asterisk server
inside_mask=255.255.255.0   ; internal ip mask.  /24 as this example
outside_addr=216.239.33.100 ; this can also be a FQDN! ie.
; my.domain.com
; ... plus whatever else you have in your sip.conf

Download the patch at:
http://bugs.digium.com/file_download.php?file_id=430type=bug

Either update your Asterisk or verify you have at least version 1.249 of
chan_sip.c:

cd /usr/src/asterisk/channels/
cvs status chan_sip.c

===
File: chan_sip.cStatus: Locally Modified
 
   Working revision:1.258
   Repository revision: 1.258  
/usr/cvsroot/asterisk/channels/chan_sip.c,v

While in pwd /usr/src/asterisk/channels/
patch -p0  /path/to/patch

Nothing should fail.

cd /usr/src/asterisk/
make
cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/

Restart your Asterisk and try it.  If you want to call a NAT'd Asterisk
box, my Free World Dialup number is 18924.  Currently online.

-- 
Leif Madsen [EMAIL PROTECTED]
http://www.hacklocalhost.com
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Re: [Asterisk-Users] Asterisk behind NAT to SIP provider

[Olle E. Johansson]

Jan Janak wrote:

I experimented a little bit and Asterisk behind NAT with SIP works. I created 
an account at iptel.org and use that account for outbound SIP traffic from
Asterisk.
I can confirm that Asterisk behind NAT can call out to IPtel.org
...and users connected to iptel.org can call me, if my server is registred
to iptel.org.
As stated earlier, the iptel.org SIP express router is configured with
a development version of the nathelper module, that assists SIP clients
inside a NAT to keep sessions open, allowing incoming calls. In this
configuration, Asterisk is simply just another SIP phone, seen from
iptel.org's point of view.
I'll update the information on the wiki so you can experiment with this.

Thank you, Jan Janak @iptel.org, for testing with me!

/Olle

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Re: [Asterisk-Users] Asterisk behind NAT to SIP provider

[Rich Adamson]

  I experimented a little bit and Asterisk behind NAT with SIP works. I created 
  an account at iptel.org and use that account for outbound SIP traffic from
  Asterisk.
 I can confirm that Asterisk behind NAT can call out to IPtel.org
 ...and users connected to iptel.org can call me, if my server is registred
 to iptel.org.
 
 As stated earlier, the iptel.org SIP express router is configured with
 a development version of the nathelper module, that assists SIP clients
 inside a NAT to keep sessions open, allowing incoming calls. In this
 configuration, Asterisk is simply just another SIP phone, seen from
 iptel.org's point of view.
 
 I'll update the information on the wiki so you can experiment with this.
 
 Thank you, Jan Janak @iptel.org, for testing with me!

Olle,

That's exactly one of the methods I was referring to in my long-winded
dissertation on asterisk with nat. There are others as well.

It would be nice if some detailed technical explanation was included
in the documentation as to why it works, and not just refer to nathelper
as though everyone reading the doc will understand what that module
is actually doing. (It probably won't help the plug-n-play newbies, but
will certainly enlighten those that keep posting unqualified responses
similar to asterisk won't work behind a nat box.)

If possible, I'd also ensure you test the config with two or more
simultanous conversations (through the nat box) as there are likely
to be some limitations that should probably be noted as well.

Rich


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[Asterisk-Users] Asterisk behind nat with hole, hardcoding solution

[Walter Snel]

Hi,

A brief 6-step guide on how to hardcode a change in the Asterisk source that
will allow it to work from behind a nat device. I know it’s messy, but it
may prove useful to some people.

1. First punch a whole in your nat device. I just forwarded the port 5060
(for sip) and all ports between 1 to 10020 (for rtp) to my asterisk
gateway.
2. Now make sure your /etc/asterisk/rtp.conf correctly reflects the 'rtp'
hole in the nat device (for me that's between 1 and 10020).

Now we need to make three small changes to the file
/usr/src/asterisk/channels/chan_sip.c

3. First find the function build_contact(…) and insert your ‘outside’ ip
address in the right position, as is indicated below (the original line is
commented out):

static void build_contact(struct sip_pvt *p)
{
/* Construct Contact: header */
if (ourport != 5060)
snprintf(p-our_contact, sizeof(p-our_contact),
sip:[EMAIL PROTECTED]:%d, p-exten, inet_ntoa(p-ourip), ourport);
else
//  snprintf(p-our_contact, sizeof(p-our_contact),
sip:[EMAIL PROTECTED], p-exten, inet_ntoa(p-ourip));
snprintf(p-our_contact, sizeof(p-our_contact),
sip:[EMAIL PROTECTED], p-exten, inet_ntoa(p-ourip));
}

4. Now find the function add_sdp(…) and replace the variable strings with
the ‘outside’ ip address (two times) as indicated below:

snprintf(v, sizeof(v), v=0\r\n);
//  snprintf(o, sizeof(o), o=root %d %d IN IP4 %s\r\n, getpid(),
getpid(), inet_ntoa(dest.sin_addr));
snprintf(o, sizeof(o), o=root %d %d IN IP4 213.84.4.39\r\n,
getpid(), getpid());
snprintf(s, sizeof(s), s=session\r\n);
//  snprintf(c, sizeof(c), c=IN IP4 %s\r\n, inet_ntoa(dest.sin_addr));
snprintf(c, sizeof(c), c=IN IP4 213.84.4.39\r\n);
snprintf(t, sizeof(t), t=0 0\r\n);
snprintf(m, sizeof(m), m=audio %d RTP/AVP, ntohs(dest.sin_port));
snprintf(m2, sizeof(m2), m=video %d RTP/AVP,
ntohs(vdest.sin_port));
/* Start by sending our preferred codecs */
cur = prefs;

5. Perform a ‘make’ in this directory, and copy the resulting ‘chan_sip.so’
file to your /usr/lib/asterisk/modules/ directory.
6. Restart asterisk.

It works for me (tested with xten softphone)

This causes Asterisk to use the outside address in all sip connections, as a
result Asterisk may become useless for sip phones on the 'inside' network.
Naturally it would be much better to make this behavior:

1. Configurable.
2. Dependent on something like an ip addressfilter so that only connections
for peers that are actually behind the nat (as indicated by a match with the
filter) are 'redirected' to the outside address.

With kind regards,
Walter Snel

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Re: [Asterisk-Users] Asterisk behind NAT to SIP provider

[Olle E. Johansson]

Jan Janak wrote:

I experimented a little bit and Asterisk behind NAT with SIP works. I created 
an account at iptel.org and use that account for outbound SIP traffic from
Asterisk.
Great! I copied your information for other users to the Wiki.

http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER

Now, I have to check why it doesn't work on my Asterisk. Propably newbie behind
console keyboard, but anyway...
/O

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Re: [Asterisk-Users] Asterisk behind NAT to SIP provider

[Rich Adamson]

  I experimented a little bit and Asterisk behind NAT with SIP works. I created 
  an account at iptel.org and use that account for outbound SIP traffic from
  Asterisk.
 Great! I copied your information for other users to the Wiki.
 
 http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER
 
 Now, I have to check why it doesn't work on my Asterisk. Propably newbie behind
 console keyboard, but anyway...

There has been a fair amount of discussion on the list as to whether nat
works with various/different configurations of sip phones with *. The
exact configuration required is highly dependent on a number of technical
factors that must be well understood before anyone can make a generic
statement relative to whether it works or doesn't work. Without that
understanding, practically every statement made on the list has been 
based on opinion and/or some trial  error methodology that has resulted
in a working example. (Nothing wrong with that, but the majority of the
postings leave out critical info that causes the next person to attempt
the same implementation but fails, and additional questions are generated.)

The critical information needed to understand nat config's include:
1. Is * behind a nat box, sip phone behind a nat box, or both?
2. Is the nat box sip aware?
3. Can the nat box be programmed to forward a static range of ports 
   to the inside?
4. Are there two nat boxes involved (one at each end of an expected
   sip-based connection)?
5. Does the sip phone support nat (eg, play nice with headers)?
6. Does * support nat (eg, play nice with headers) and is it config'ed?
7. Are there timers involved at either end of a nat traversal that
   are intended to keep nat table entries from timing out?
8. If so, what are the actual timeout values used for the specific
   nat box, and are sip end-point timers less then those of the nat
   box? (Don't assume all sip phones with nat functions are equal.)
9. What is the nat impact of a sip phone that has been configured to 
   re-register every 60 seconds?
10. What is the range of rtp ports expected by the sip phone (eg, 7960's
range from 16384 to 32766, but can be changed; xten uses 8000
to 8012 or something like that)?
11. Can the user implement iax (instead of sip) between end points?
12. When nat is found to function correctly, which end originated
the nat traversal (makes a BIG difference)?
And, probably another half dozen technical parameters that I'm forgetting
to mention.

I've spent many years working with corporate clients in more then 40
states diagnosing networking issues, doing protocol analysis, etc, and
have seen a large number of nat boxes. The nat implementations from
various vendors range from very basic translation tables to some rather
sophisticated functions. And, just because a nat implementation comes
from a well-known vendor doesn't mean anything (even Cisco has problems
with no nat timeouts in certain boxes today).

With that said, here's a couple of high-level examples that could 
work but these are not based on actual lab tests, etc.

1. If * is behind a nat box and * inititiates a tcp/udp conversation
   with a non-nat'ed address, some form of timer-based keep alive
   packet will keep the nat-box-table-entries active allowing the
   implementation to work. (Obviously assumes equipment can support
   sip header functions.) What are some of the configuration issues
   that may need to be addressed?
   a. limit the port numbers that can be used by * (rtp.conf)
   b. limit the port numbers that can be used by the sip phone.
   c. may still need to map the specific rtp port range in the nat
  box depending upon the nat box functionality.
   d. probably define nat=yes within *.
   (The real issue here is which end initiated the conversation
   and what is used to keep the nat translations active. I think we've
   already heard some folks doing this with certain Internet-based
   companies, but the postings left out a bunch of technical 
   configuration data on both ends.)

2. * = nat = Internet = nat = sip phone
   Implement a combination of #1, above, at both ends assuming the 
   end-point equipment has the capability to be configured (including
   the sip phone, nat boxes, etc).

What tends to aggravate nat implementations are those NAT boxes that
also implement PAT (port address translation), and the box vendor doesn't
bother to hint at it in their documentation. (There are a very large
number of networking folks that don't understand this, and its probably
safe to assume 99.99% of the user community has never heard of it.)
The PAT issues usually end up with someone suggesting sip phone #1 works 
but #2 doesn't and they are configured exactly the same. Or, call #1 
works but call #2 fails. (And then the next person on the list says
it works fine for them, but doesn't mention who's nat box he's using
or what it's actually doing from a technical perspective.) 

I'd bet a small amount of money that 

Re: [Asterisk-Users] Asterisk behind NAT to SIP provider

[Olle E. Johansson]

Rich Adamson wrote:

I experimented a little bit and Asterisk behind NAT with SIP works. I created 
an account at iptel.org and use that account for outbound SIP traffic from
Asterisk.
Great! I copied your information for other users to the Wiki.

http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER

Now, I have to check why it doesn't work on my Asterisk. Propably newbie behind
console keyboard, but anyway...


There has been a fair amount of discussion on the list as to whether nat
works with various/different configurations of sip phones with *. The
exact configuration required is highly dependent on a number of technical
factors that must be well understood before anyone can make a generic
statement relative to whether it works or doesn't work. Without that
understanding, practically every statement made on the list has been 
based on opinion and/or some trial  error methodology that has resulted
in a working example. (Nothing wrong with that, but the majority of the
postings leave out critical info that causes the next person to attempt
the same implementation but fails, and additional questions are generated.)
Rich,
Thank you for your additional information on the NAT/VoIP issue. Is it ok
with you if I add it to the Wiki?
As you say, we need to collect information and compose a data base of
what works and what's not working in certain circumstances.
Jan got * - SER working, I can't. We have different NAT:s. To try to
solve my problem I made sure his solution was documented so far.
There's no silver bullet here. With NATs, we've built a network without
end-to-end connectivity and we need to  patch it up to get VoIP working
on an IPv4 network with NATs in every corner.
I just hope that IPv6 will make life easier for the next generation of
VoIP users. Right now, we need to understand all variables.
/O

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Re: [Asterisk-Users] Asterisk behind NAT to SIP provider

[rnc Info Lists]

 My asterisk server(s) are behind NAT, and I am a customer of Vonage
 (thrice-over), iconnecthere, and Net2Phone.

 There are still some rough edges (especially with iconnecthere) but
 overall it is not correct to say that they won't work.

 B.

Thats great to hear.  Can you please share your config files that connect
iconnecthere and net2phone via SIP?  I think there are a number of people
here who have tried and not been able to get it to work.

Robert
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Re: [Asterisk-Users] Asterisk behind NAT to SIP provider

[Brian Capouch]

rnc Info Lists wrote:

Thats great to hear.  Can you please share your config files that connect
iconnecthere and net2phone via SIP?  I think there are a number of people
here who have tried and not been able to get it to work.
Here's what I'm using for iconnecthere.  They provide me with both 
origination and termination, btw, so there are clauses that handle each.

***
in sip.conf:
register = 18005551212:[EMAIL PROTECTED]
(first part is my inbound phone number, second is account password)
[iconnect]
type=peer
username=12312312
secret=
callerid = My Name 18005551212
host=213.137.73.140
And in extensions.conf:

exten = _11.,1,Goto,iconn|BYEXTENSION|1

Later on. . .
[iconn]
exten = _11NXXNXX,1,StripMSD,1
exten = _1NXXNXX,2,Prefix,
exten = _1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]||r
For origination:

exten = 15126919417,1,Dial,SIP/ata1|23



Note I'm using the old (deprecated) syntax for the various commands. 
And I don't pretend this is beautiful or optimal syntax.  The  
preceding the number was something they told me to use to get gsm encoding.

FWIW.

B.

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Re: [Asterisk-Users] Asterisk behind NAT to SIP provider

[Jan Janak]

I experimented a little bit and Asterisk behind NAT with SIP works. I created 
an account at iptel.org and use that account for outbound SIP traffic from
Asterisk.

I am using [EMAIL PROTECTED], all the SIP traffic will be sent to
iptel.org proxy and the proxy will take care of NAT traversal. Currently
I forward all numbers begining with 3 to iptel.org beucase I don't know
how to create fall-back rule that will match when there are no other
rules (neither i nor _. works for me).

In the other direction, calls to [EMAIL PROTECTED] get translated to
[EMAIL PROTECTED] and user jan registered at the asterisk box will
receive them.

To able able to call anywhere through iptel.org, From header field must
contain iptel.org so fromdomain parameter is necesarry in [iptel]
section.

Testing scenario was as follows:

[Caller][*]---[NAT][iptel.org (public inet)][NAT]---[Callee]

and vice versa.

sip.conf and extensions.conf follow. I have no previous experience in
configuriing asterisk so maybe the config files are not the best ones, I
simply took John Todd's config files and tweaked them a bit, it seems to
work for me.

To iptel.org proxy asterisk looks like a normal SIP user agent behind
NAT. iptel.org is running SER with extended nathelper and RTP proxy.

  Jan.

;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0  ; Address to bind to
context = from-sip  ; Default for incoming calls
;
register = asterisk:[EMAIL PROTECTED]/jan ; Register with a SIP
provider

[iptel]
type=friend
username=asterisk
secret=password
fromdomain=iptel.org
host=iptel.org

[jan]
type=friend
username=jan
host=dynamic
canreinvite=no


extensions.conf:

[from-sip]
exten = jan,1,Dial(SIP/jan)
exten = jan,2,Hangup
exten = _3.,1,SetCallerID(jan)
exten = _3.,2,SetCIDName(Jan Janak)
exten = _3.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _3.,4,Playback(invalid)
exten = _3.,5,Hangup

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[Asterisk-Users] Asterisk behind NAT to SIP provider

[Jonathan Hogg]

Hi all,

OK. I've tried trawling the archives, but I'm not getting very far. I've got
an Asterisk box behind a NAT which I want to register with a SIP provider.

In my sip.conf I have (edited to protect the innocent):

-
[general]
port = 5060
bindaddr = 0.0.0.0
disallow = all
allow = alaw
allow = ulaw
allow = gsm
context = bogus-calls
tos = lowdelay
nat = yes
register = 8703405315:[EMAIL PROTECTED]

[8703405315]
type = friend
reinvite = no
canreinvite = no
nat = yes
username = 8703405315
secret = 
context = from-sip-provider
-

With 'sip debug' on, I can see it sending the REGISTER requests and getting
back a response with STUN headers like so (also edited):

-
SIP/2.0 407 Proxy Authorization Required
X-Stun-Server: w.x.y.z:3478
X-Observed-Adr: a.b.c.d
...
-

However, when Asterisk sends the auth it doesn't sends the REGISTER again to
the same address without seeming to take into account the STUN details, a
la:

-
REGISTER sip:sip-provider.not SIP/2.0
Via: SIP/2.0/UDP 10.20.15.4:5060;branch=z9hG4bK43e3ead5
...
Contact: sip:[EMAIL PROTECTED]
...
-

This results in me getting a 406 Bad Contact (NAT) response.

My questions:

 a) Does Asterisk support what I want to do (please don't tell me to use
IAX instead - I am already talking to the provider about that, but they
are in the early stages of playing with Asterisk)?

 b) What have I done wrong in my sip.conf? I've been hacking it around for a
while this afternoon so it's a bit of a mess of mangled attempts to make
it work.

Any help gratefully appreciated.

Jonathan

-- 
Jonathan Hogg
Director, Technology

Seventh Wave Systems Ltd.
4-14 Tabernacle Street
London EC2A 4LU
Telephone: +44 20 7074 0423

http://www.seventh-wave-systems.com/

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Re: [Asterisk-Users] Asterisk behind NAT to SIP provider

[Olle E. Johansson]

Jonathan Hogg wrote:

OK. I've tried trawling the archives, but I'm not getting very far. I've got
an Asterisk box behind a NAT which I want to register with a SIP provider.
If you've travelled around the archives, you should now that this is a FAQ.

At this moment, Asterisk behind a NAT can't connect to an outside SIP
provider. If you put asterisk outside your NAT, your inside clients
can connect to Asterisk and Asterisk will be able to connect to your providers.
There are bug reports, web pages and mail in the archive that document this.
Start at http://www.voip-info.org - click on Asterisk.
/O

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