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
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
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
>
> *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>
, 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 Serve
t;
>>>
>>>
>>>
>>> With best regards
>>>
>>> Florian Floimair
>>> Innovation - Software-Development - VoIP & DevOps
>>>
>>> COMMEND INTERNATIONAL GMBH
>>> A-5020 Salzburg, Saalachstraße 51
>>> Tel:
ONAL 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
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 Vi
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})
: 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 th
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
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,
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
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
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
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
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
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
Is directmedia set to no?
On 15 January 2014 23:11, Leandro Dardini ldard...@gmail.com wrote:
Hello,
I have an asterisk box with a peer configured with
nat=force_rport,comedia, but asterisk keeps sending the audio to the
private IP address and ignoring the client peer nat settings.
If I
Yes, thank you. Maybe I have found the problem. The asterisk server is
behind a nat and the RTP port range was not redirected to the asterisk box,
so the Symmetric RTP cannot work because the asterisk is not receiving any
RTP packet from the remote phone.
Leandro
2014/1/16 Ishfaq Malik
- Original Message -
From: Andres and...@telesip.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 15, 2014 7:51:28 PM
Subject: Re: [asterisk-users] Asterisk ignoring nat settings
Why don't you try with nat
On 1/16/14, 2:23 PM, Michael L. Young wrote:
- Original Message -
From: Andres and...@telesip.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 15, 2014 7:51:28 PM
Subject: Re: [asterisk-users] Asterisk ignoring nat
- Original Message -
From: Andres and...@telesip.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, January 16, 2014 4:17:53 PM
Subject: Re: [asterisk-users] Asterisk ignoring nat settings
I am curious why you would say
15, 2014 7:51:28 PM
Subject: Re: [asterisk-users] Asterisk ignoring nat settings
Why don't you try with nat=yes. It should be equivalent to what you
have but who knows. It might just work.
I am curious why you would say that nat=yes might work over
nat=force_rport,comedia? As you stated
Hello,
I have an asterisk box with a peer configured with nat=force_rport,comedia,
but asterisk keeps sending the audio to the private IP address and ignoring
the client peer nat settings.
If I check the sip show peer extension, I see both symmetric RTP and
Force Rport are set to yes, but
On 1/15/14, 6:11 PM, Leandro Dardini wrote:
Hello,
I have an asterisk box with a peer configured with
nat=force_rport,comedia, but asterisk keeps sending the audio to the
private IP address and ignoring the client peer nat settings.
Why don't you try with nat=yes. It should be equivalent to
Jeremy Kister wrote:
using asterisk 11.6.0-rc1 i just converted my nat=yes to
nat=auto_force_rport,auto_comedia
I have my asterisk box on the same subnet as a cisco 1760 (vgw1).
a few times per day, Asterisk thinks vgw1 is dead (by qualify/options).
A 'sip reload' always fixes the
using asterisk 11.6.0-rc1 i just converted my nat=yes to
nat=auto_force_rport,auto_comedia
I have my asterisk box on the same subnet as a cisco 1760 (vgw1).
a few times per day, Asterisk thinks vgw1 is dead (by qualify/options).
A 'sip reload' always fixes the problem.
i left 'sip set
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
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.
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
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
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 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
--
Jeremy
Hello may situation is the next:
Asterisk -- NAT1 (router)--- internet -- NAT2 (router) -- x-lite
^
|
ip phone (cisco)
Asterisk and de cisco phone are in the same LAN. I want to make a
call between the x-lite and the
On Thu, Dec 18, 2008 at 12:46 AM, Silvia Menendez silvia.menen...@gmail.com
wrote:
Hello may situation is the next:
Asterisk -- NAT1 (router)--- internet -- NAT2 (router) -- x-lite
^
|
ip phone (cisco)
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
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
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
Hi All;
My Asterisk has a public IP address, how can we let
two IP Phones in different site and both are behind
NAT (each one has a private IP address) to call each
other?
In other words,
Assuming Asterisk IP Address is 193.111.194.111
IP Phone (A): 192.168.0.1 and its default gateway is:
On Tue, 2007-12-11 at 00:14 -0800, bilal ghayyad wrote:
Hi All;
My Asterisk has a public IP address, how can we let
two IP Phones in different site and both are behind
NAT (each one has a private IP address) to call each
other?
In other words,
Assuming Asterisk IP Address is
On Tue, 11 Dec 2007 11:05:24 -0600, Carlos Chavez
[EMAIL PROTECTED] wrote:
The only thing you need to do is set nat=yes when you configure the
phones in Asterisk. You may need to use a STUN server in case the
phones do not properly see the outside address. Once the phones
register they
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
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
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
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
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
Does the Linksys know it should be using port 5070? It would seem to me
that port forwarding would be required as the phones are behind a NAT'd
firewall. How would asterisk know how to get there since it's not on
the same subnet (outside the firewall).
If the asterisk box has physical access to
Hi
I am search a small information
- i use Asterisk on official IP without Nat
- My first VoIP phone are a Thomson 2030 on a NAT Network.
That's work very good.
But now, i want add a second phone, a Linksys SPA-941 on
the same network of the Thomson 2030 ...
My problems that i don't see
Noc Phibee wrote:
Hi
I am search a small information
- i use Asterisk on official IP without Nat
- My first VoIP phone are a Thomson 2030 on a NAT Network.
That's work very good.
But now, i want add a second phone, a Linksys SPA-941 on
the same network of the Thomson 2030 ...
My
yusuf a écrit :
Hi,
you dont have to/should'nt be using different SIP ports for each
phone. Its completely not needed. Also, you dont have/need to port
forward. Just open ports 5060 and 1000-2, on the box that
asterisk is running, and on your NAT router. Dont port forward.
Then in
Does the phone have stun settings? If so, try using stun.fwdnet.net and take out the port forwards and see if it works.
bp
On 9/7/06, Noc Phibee [EMAIL PROTECTED] wrote:
yusuf a écrit : Hi, you dont have to/should'nt be using different SIP ports for each
phone.Its completely not needed.Also, you
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
-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
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
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
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
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
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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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 *
Hey guys.
I have to put my * behind a Firewall through nat on the firewall.
The asterisk is running, but for example a register to an outside PSTN
provider won't work.
I enabled nat for the register but i only get Code 120 Send request.
The other problem is, when i try to register with a sip
René Enskat [Teamware GmbH] wrote:
Hey guys.
I have to put my * behind a Firewall through nat on the firewall.
The asterisk is running, but for example a register to an outside PSTN
provider won't work.
I enabled nat for the register but i only get Code 120 Send request.
The other problem
You've not said much about your firewall setup. I presume you've opened
up 5060 and RTP ports?
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Asterisk-Users mailing list
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Alex Lake
Gesendet: Dienstag, 4. Oktober 2005 11:47
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [Asterisk-Users] Asterisk and NAT
You've not said much about your firewall setup. I presume
you've opened up 5060 and RTP ports
I guess you could post your config files here and hope that someone
feels inclined to look them over! ;-)
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As well as entered externip=xxx.xxx.xxx.xxx in your SIP.conf file?
Alex Lake wrote:
You've not said much about your firewall setup. I presume you've opened
up 5060 and RTP ports?
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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 mailing list
Not all firewalls handle sip correctly.
What is the firewall that u have?
Basically I would set everything like this:
extenip = static ip or DDNS ip
localnet = you local network the same the asterisk is on
On the firewall forward ports 5060 and ports in rtp.conf file to
you asterisk box, if
what is the upload speed on B?
Looks to me as you have bandwidth problem!
Martin Kronstad wrote:
Hi!
Problem:
I can’t hear what the people at Location B i saying, they hear me but I
do not hear them. They can call, I can call. Just no sound.
My current setup is:
Hi!
The bandwith is not the problem, uploadspeed is about
400 kbits.
I think I found the solution, I need to have a Proxy in
the middle, or set up a IAX2 client and server at each end
I will be testng this next week.
BR Martin Kronstad
What is the
upload speed on B?
Looks
Switch to IAXCOMM and use an IAX extension. Problem
solved.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin
KronstadSent: Friday, August 05, 2005 7:03 AMTo: 'Asterisk
Users Mailing List - Non-Commercial Discussion'Subject:
[Asterisk-Users] Asterisk - Firewall
On Aug 4, 2005, at 10:37 PM, Martin Kronstad wrote: Hi!Problem:I can’t hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound.My current setup is:Softphones/Hardphones(Location A) - Asterisk - Firewall/Nat - Internet -
Hi!
Problem:
I cant hear what the people at Location B i
saying, they hear me but I do not hear them. They can call, I can call. Just no
sound.
My current setup is:
Softphones/Hardphones(Location A) - Asterisk -
Firewall/Nat - Internet - Firewall/Nat -
Dear All,
I am new to this mailing list , I have bought some digium cards to play with , Installed it and configured asterisk . I was able to test voicemail IVR , I succeeded also to use xlite from a windows machine to call another phone through a PSTN line. and call the xlite client from a
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
H.323 will not traverse NAT.
Sorry... I know, I was a big proponent of it when H.323 was the
only standard VoIP protocol out there. Probably because
when it came out NAT wasn't even thought of.
The problem is that the control channel in H.323 discloses the internal
IP address, and the various
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
- 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
-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
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
-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
: 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
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
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
Hello,
I've been working a lot with asterisk lately. I've
had a LOT of positive experience with various SIP
clients (grandstream hardware phones ATAs, X-Lite,
SJPhone, etc...), and I've had no trouble getting
asterisk behind a NAT to talk SIP to clients across
the internet behind another 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
:[EMAIL PROTECTED] On Behalf Of Paul Fielding
Sent: Martes, 29 de Marzo de 2005 12:52 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate
- Original Message -
From: Anton Krall [EMAIL PROTECTED]
would
-
From: Anton Krall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, March 29, 2005 5:28 AM
Subject: RE: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate
Thank you for your story Paul, nice work
PROTECTED] On Behalf Of Paul Fielding
Sent: Martes, 29 de Marzo de 2005 08:27 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate
Basically, I'm forwarding the standard Asterisk ports:
tcp 5060
udp 5060
udp
Anton Krall wrote:
Any problems with RTP or voice just on one side?
So as long as you use some STUN server, the RTP packets have the right IP.
Did you install your own stund or are you using a public one?
You didn't have to use SER at all right?
Setting nat=yes does pretty much the same as a STUN
ManxPower
Sent: Martes, 29 de Marzo de 2005 10:28 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate
Anton Krall wrote:
Any problems with RTP or voice just on one side?
So as long as you use some STUN
Guys.
Im reading a lot about ser, nat, stun, etc. And I noticed there are a lot of
ways to get around nat but I would like to hear some success stories about
handling nat users with multiple voip phones behind nat.
I have my asterisk box behind but ports are forwarded (5060 5004 1-2
for
- Original Message -
From: Anton Krall [EMAIL PROTECTED]
would like to hear some actual setups and how people are solving the nat
issue within scenarios like:
Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones
I've been playing with this with my friends for awhile
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
: 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
Stuart Ford wrote:
Seriously, this has to be the simplest NAT problem there is with
Asterisk. What's the secret? How do I learn the dark art? What am I
missing?
I'm guessing here, but the NAT'd grandstream does not have the correct
external IP configured.
The phones are trying to establish a
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
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
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
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