Please don't top post.
On Sun, 7 Apr 2013, Thomas Perron wrote:
Got it...
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 954)
Asterisk*CLI> sip show registry
Host dnsmgr Username Refresh State
Reg.Time
s
Got it...
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 954)
Asterisk*CLI> sip show registry
Hostdnsmgr Username Refresh
State Reg.Time
sip3.voipvoip.com:5060 N 444222146 105
R
A better subject will yield better replies.
On Sat, 6 Apr 2013, Thomas Perron wrote:
Shouldnt I be able to at least ping the SIP provider IP?
Not if they don't allow it. They don't.
sip3.voipvoip.com registers fine for me with your credentials.
Did you put the registration statement in the
I have a very lite layout and attempting to get the SIP configuration set
up initially before proceeding into other areas.
VMware is running my Asterisk 11 on Ubuntu 12.
Shouldnt I be able to at least ping the SIP provider IP?
I run command "sip show registry" and do not see it set up.
I run sip
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Monday, October 08, 2012 12:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Sip re
Hello,
I have a local Asterisk server that keep loosing its registration to main
Asterisk server. The local asterisk server is on the local subnet, it acts
as a client with extension 808.
Local server
Sip.conf
register => 808:passw...@as2.x.com
registertimeout=20
registerattempts=10
Main A
On Thu, 26 Jan 2012, eherr wrote:
It is accessible from HTTP.
However, the access list only allows access from my home and the
password is strong.
Can you configure it to 'syslog' accesses where you can monitor it.
Maybe your access lists are invalid, misunderstood or not being honored.
--
:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Sip Registration Hijacking
On 20/01/12 01:36, eherr wrote:
>
> It is also register on an AudioCodes MP-118.
> Thanks,
>
> -E
>
Is the Audiocodes gateway accessible online? Have you set a strong
pas
On 20/01/12 01:36, eherr wrote:
It is also register on an AudioCodes MP-118.
Thanks,
-E
Is the Audiocodes gateway accessible online? Have you set a strong
password for it's web interface (and cli if it has one)? It is possible
someone is breaking into that and getting the SIP password o
Behalf Of Larry Moore
Sent: Saturday, January 21, 2012 1:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sip Registration Hijacking
On 20/01/2012 9:36 AM, eherr wrote:
I have a honey pot box with extensions that are not just numbers ie )
100
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mikhail Lischuk
Sent: Friday, January 20, 2012 7:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sip Registration Hijacking
Alejandro Imass wrote
: [asterisk-users] Sip Registration Hijacking
Rate limiting (google) via iptables FTW! Good luck!
- Original message -
>
>
> Alejandro Imass wrote 20.01.2012 18:09:
>
> > I would like to know how
> to block this MF because he makes calls at 1-2 AM
>
> I
On 20/01/2012 9:36 AM, eherr wrote:
I have a honey pot box with extensions that are not just numbers ie )
100-MySipUserName
And the passwords are from an openssl generated password ie)
Gq5VNIjDFWIQoUT6
Is the password stored in sip.conf in plain text or as an MD5?
If it is stored in plai
Rate limiting (google) via iptables FTW! Good luck!
- Original message -
>
>
> Alejandro Imass wrote 20.01.2012 18:09:
>
> > I would like to know how
> to block this MF because he makes calls at 1-2 AM
>
> I use this
> construction on my servers
>
> [users]
>
> exten =>
> _XXX,1
Alejandro Imass wrote 20.01.2012 18:09:
> I would like to know how
to block this MF because he makes calls at 1-2 AM
I use this
construction on my servers
[users]
exten =>
_XXX,1,GotoIfTime(1:00-2:00,*,*,*?block,1,1)
[block]
exten =>
_X.,1,HangUp(1)
--
With Best Regards
Mikhail Lischu
On Fri, Jan 20, 2012 at 11:17 AM, eherr wrote:
> I always thought Sip Vicious only does numbers ( 0 - 100 ) not
> Numberic-Alpha ( 100-MySipUserName ).
>
> To make my situation more interesting is that I also have fail2ban installed
> banning after 5 failed attempts.
I too have fail2ban an
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sip Registration Hijacking
On Thu, Jan 19, 2012 at 8:36 PM, eherr wrote:
> I have a honey pot box with extensions that are not just numbers ie )
>
>
>
> 100-MySipUserName
>
>
>
I have the same pr
On Thu, Jan 19, 2012 at 8:36 PM, eherr wrote:
> I have a honey pot box with extensions that are not just numbers ie )
>
>
>
> 100-MySipUserName
>
>
>
I have the same problem and I use contactpermit with specific ip blocks!
I know for a fact I'm getting hijacked by sip vicious on extension 100
bu
I have a honey pot box with extensions that are not just numbers ie )
100-MySipUserName
And the passwords are from an openssl generated password ie)
Gq5VNIjDFWIQoUT6
However, this one extension keeps getting hacked and showing up on a different
IP address.
It is also register o
turday, November 19, 2011 8:43:22 PM
> Subject: [asterisk-users] SIP registration issues
> Hi,
>
> Having problems with a client trying to login to Asterisk 1.6.2 from
> behind a DSL router. The account can be accessed perfectly from other
> clients.
>
> Would appreciate if
Hi,
Having problems with a client trying to login to Asterisk 1.6.2 from
behind a DSL router. The account can be accessed perfectly from other
clients.
Would appreciate if you could look at the the attached log and see if
you spot any glaring issues. The user is very infrequently available
f
On 17/03/11 05:37, Patrick wrote:
Dear mailing list,
I've a Asterisk 1.4.21.2~dfsg-3+lenny1 package installed on my debian
and I've a strange behavior.
After some days running normally, my asterisk is under heavy attack,
however, there is nothing logged in the console (logging from debug ->
err
Dear mailing list,
I've a Asterisk 1.4.21.2~dfsg-3+lenny1 package installed on my debian
and I've a strange behavior.
After some days running normally, my asterisk is under heavy attack,
however, there is nothing logged in the console (logging from debug ->
error) or file (level from notice ->err
Hi,
Are sip.conf's defaultexpiry and maxexpiry global?
Or can they be used on a per-extension basis?
I'd like to "force" some extensions to re-register more frequently than others
(server-side).
Thanks,
Vieri
--
_
-
I have a problem that when one of my SIP providers has a problem the
rest of my SIP extensions and trunks stop working until either the SIP
provider fixes the problem or Asterisk stops trying to register to that
provider. Why does this happen? A single provider having problems
should not
Try:
core set verbose 4
>From the Asterisk CLI
-uzzi
PS: If you're not seeing any connection information, be sure to double-check
the IP address is correct. Learned that lesson the hard way =\
On Sun, Jan 31, 2010 at 5:51 PM, Jim Rosenberg wrote:
> Let's say I have two Asterisk boxes, A and
Let's say I have two Asterisk boxes, A and B. I am trying to get A to do
SIP registration on B, so an extension for A can dial SIP phones covered by
B. If I examine the logs on B, if the registration succeeds, I am seeing a
notice to that effect on B. But if the registration *fails*, i'm not see
SIP-registration errors are solved by restarting the Asterisk-server.
But I expect them to return in time...
I can make call now, but the other end does not hear me. So problem with
RTP-flow...
Can someone guide me to the solution ?
On the Asterisk-server I have this (iptables):
-A RH-Firewal
Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports
opened and 5060 forwarded to Asterisk (192.168.2.2)
Can someone see why SIP-registration fails ??
register => 092779077:x...@85.119.188.3
[3starsnet]
type=peer
host=85.119.188.3
username=092779077
secret=
fromuser=092779077
7 apr 2009 kl. 12.08 skrev Steve Davies:
> 2009/4/7 Olle E. Johansson :
>>
> [snip]
>>
>> The REGISTER request in the RFC was really written for a device.
>> The way providers use it for trunks with multiple DIDs is outside
>> of the
>> RFC and is discussed in relation to the SIPconnect specifi
2009/4/7 Olle E. Johansson :
>
[snip]
>
> The REGISTER request in the RFC was really written for a device.
> The way providers use it for trunks with multiple DIDs is outside of the
> RFC and is discussed in relation to the SIPconnect specification in
> the SIP forum.
>
> Some providers solve this
6 apr 2009 kl. 18.46 skrev Steve Davies:
> Thanks for the reply - Perhaps I was not clear.
>
> On the register=> line, if I set /extension to be /12345, then this
> just replaces 's' with 12345, and ALL calls, regardless of their
> destination number will be routed on the INVITE line to 12...@x.x
Thanks for the reply - Perhaps I was not clear.
On the register=> line, if I set /extension to be /12345, then this
just replaces 's' with 12345, and ALL calls, regardless of their
destination number will be routed on the INVITE line to 12...@x.x.x.x,
and the actual destination is specified in the
Have you looked at the syntax of register => keyword ?
register => [transport://]user[:secret[:authuse...@host[:port][/extension]
; If no extension is given, the 's' extension is used.
There you have it ... Contact: wrote:
> I have an ITSP we are trying to work with that has an "Unusual" way of
I have an ITSP we are trying to work with that has an "Unusual" way of
working, but that said my understanding of their behaviour is that it
is fully RFC compliant. Can someone suggest how I might be able to
interoperate under these circumstances:
We register fine with them, and send the default a
Hi,
I have this weird problem i cant explain.
i have two asterisk, i'm using realtime table for my sip/user accounts.
my database is on a mysql cluster.
my prob is if i register on phone on asterisk 1 it is ok, but on second
asterisk it can't,
Registration from '"122144" ' failed for
'12.34
Hi,
I have this weird problem i cant explain.
i have two asterisk, i'm using realtime table for my sip/user accounts.
my database is on a mysql cluster.
my prob is if i register on phone on asterisk 1 it is ok, but on second
asterisk it can't,
Registration from '"122144" ' failed for
'12.34
you have this option on major phones also, try that.
2008/7/31 Vieri <[EMAIL PROTECTED]>
> Hi,
>
> If I set maxexpirey=60 in sip.conf and also set a "registration timeout=60"
> on client software, doesn't this mean that the SIP user (an ATA connected
> phone) should be "forced" to re-register eve
Hi,
If I set maxexpirey=60 in sip.conf and also set a "registration timeout=60" on
client software, doesn't this mean that the SIP user (an ATA connected phone)
should be "forced" to re-register every minute?
If I look at the CLI when the SIP user registers I do see a statement regarding
a 60
Hi,
I have been using asterisk-1.4.17 version. Have a SIP registration from bandtel
sip providers.
Use DID numbers for the incoming calls which works fine when i dont use any
peer setting in my sip.conf file. But when i use a peer and make calls thru the
DID number it doesn't reach asterisk at
ers@lists.digium.com
Sent: Saturday, May 05, 2007 4:08 PM
Subject: [asterisk-users] SIP registration problem
I've reposted with a more meaningful subject - hopefully someone will
replyWe have an Asterisk v1.2.16 box registering with an ITSP using SIP.
The registration succeeds, and is confir
I've reposted with a more meaningful subject - hopefully someone will
replyWe have an Asterisk v1.2.16 box registering with an ITSP using SIP.
The registration succeeds, and is confirmed with SIP SHOW REGISTER.
However, we frequently (every few minutes) see this on our console:
REGISTER attem
hi, to get it work i change under sip.conf
nat: route
Allow RTP reinvite:update
with that i can hear, without dmz... but... why?
2007/4/19, Manolet Gmail <[EMAIL PROTECTED]>:
Hi, now i can log in ok on my xlite, somebody calls me and everythink
its okey. i hear and the caller hear. (the pc wit
Hi, now i can log in ok on my xlite, somebody calls me and everythink
its okey. i hear and the caller hear. (the pc with the xlite have
DMZ).
But now i close xlite and put the same extension on a grandstream 286
(dont have DMZ). When somebody calls me the caller can hear me. but i
cant hear!
wha
hello,
I use both * 1.4 and *NOW... because the *gui is incomplete in *NOW, I
loaded 1.4 over *NOW because the gui regenerates files that, well, don't
seem to work very well. it seems to me the gui creates the users.conf
file, and then a script creates or uses the users.conf to create the
d
The quick way to check if a user is defined is to go to the asterisk console
and type "sip show users" which will list all the defined users and
passwords.
You say that it isn't a networking issue, but the fact that you are behind a
NAT (your local ip is 192.168.0.100) is causing the problem (i t
On Fri, 13 Apr 2007, Manolet Gmail said something to this effect:
mmm are you sure that asterisk-gui generate it on the sip.conf file?
cause i see a new file called users.conf, and i can see the sip users
on it. Anybody uses asterisk now and can check it please??
Hmm. I use 1.4.x here and
mmm are you sure that asterisk-gui generate it on the sip.conf file?
cause i see a new file called users.conf, and i can see the sip users
on it. Anybody uses asterisk now and can check it please??
2007/4/13, Alex Balashov <[EMAIL PROTECTED]>:
On Fri, 13 Apr 2007, Manolet Gmail said something to
On Fri, 13 Apr 2007, Manolet Gmail said something to this effect:
of course, download it from here:
http://contelecltda.com/sip.conf
but i dont edit the sip.conf, is the default make samples sip.conf file.
i just use the asterisk gui interface to add the user...
Well, then my conjecture w
of course, download it from here:
http://contelecltda.com/sip.conf
but i dont edit the sip.conf, is the default make samples sip.conf
file. i just use the asterisk gui interface to add the user...
2007/4/13, Alex Balashov <[EMAIL PROTECTED]>:
Hi Manolet,
Can you provide your sip.conf?
Tha
Hi Manolet,
Can you provide your sip.conf?
Thanks!
-- Alex
--
Alex Balashov <[EMAIL PROTECTED]>
___
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hi!
First of all i want to tell i have a dedicated server on layeredtech
with direct internet connection and i currently dont use iptables, so
this is not about network configuration =).
well so, i install asterisk-1.4.2 on my server, and next install
asterisk-gui from the digium repository.
ne
The problem was on the polycom provisioning setup. In my dhcp settings I
wasn't giving it the correct domain-name-servers option. I changed that
and I changed the phones to use [EMAIL PROTECTED] instead of
[EMAIL PROTECTED] and that seems to have taken care of it.
Thanks for the help.
Nathan
That doesn't seem to make any difference. I still get the "Not a local
SIP domain" and I get this from the CLI:
ast*CLI> sip show peers
Name/username HostDyn Nat ACL Port Status
202(Unspecified)D 0Unmonitored
201
Hi Nathan -
I just saw this post about having trouble registering your phone ;-)
When my SIP phones try to register with my asterisk box, this is what I
get my log file:
Mar 26 14:46:41 NOTICE[3896] chan_sip.c: Registration from
'' failed for '192.168.3.2' - Not a local SIP domain
sip.conf
When my SIP phones try to register with my asterisk box, this is what I
get my log file:
Mar 26 14:46:41 NOTICE[3896] chan_sip.c: Registration from
'' failed for '192.168.3.2' - Not a local SIP domain
In sip.conf I have this for my global settings:
[general]
context=from-sip;
Thanks Andrew,
I see the resolved bug report. I'll get the patch fix.
Sorry for the unnecessary mail.
-Tom
On 1/20/07, Andrew Joakimsen <[EMAIL PROTECTED]> wrote:
http://www.google.com/search?q=423+%22Interval+Too+Brief%22&start=0&ie=utf-8&oe=utf-8&client=firefox-a&rls=org.mozilla:en-US:of
http://www.google.com/search?q=423+%22Interval+Too+Brief%22&start=0&ie=utf-8&oe=utf-8&client=firefox-a&rls=org.mozilla:en-US:official
Hint: Who develops Asterisk?
On 1/20/07, Thomas Madler <[EMAIL PROTECTED]> wrote:
Hi,
I'm trying to get my * server connected to a softswitch through an SBC. I
Hi,
I'm trying to get my * server connected to a softswitch through an SBC. I
get the following error when * trys to register.
Got SIP response 423 "Interval Too Brief" back from xxx.xxx.xxx.xxx
Jan 20 12:43:54 NOTICE[2138]: chan_sip.c:5473 sip_reg_timeout:--
Registration for '[EMAIL PROTEC
I have a customer on one of my Asterisk boxes that wants a small number
of DIDs in Hong Kong. Referring to voip-info.org, we found the provider
HKBN and their 2b service at www.2b.com.hk.
Following the information at
http://www.voip-info.org/wiki/index.php?page=asterisk+settings+HKBN+2b
we success
Can anyone give me any
direction as to why I'm getting a 404 during the registration process. Sip
Debug is:
<-- SIP read from
192.168.99.110:5060:
REGISTER
sip:asterisk1.rightsolve.com SIP/2.0
Via: SIP/2.0/UDP
192.168.99.110:5060;branch=123456789
To:
From:
;tag=12345
CSeq: 1 REGI
> Hi,
I'm a user of [EMAIL PROTECTED] for a couple of months now and been
playing around with it for a while. I'm facing a strange situation
which i am not able to solve.
I have my * server and a SIPURA 841 phone both behind my router at
home (No NAT between them). My * server is registered to 192
I am a newbie and am having trouble trying to register with a voip
provider using sip. I am able to connect using xlite softphone. in
xlite i use
domain/realm: providerdomain.com
sip proxy: host.providerdomain.com:9000
this difference in domain and sip proxy host is whats causing problem
I am a newbie and am having trouble trying to register with a voip
provider using sip. I am able to connect using xlite softphone. in
xlite i use
domain/realm: providerdomain.com
sip proxy: host.providerdomain.com:9000
this difference in domain and sip proxy host is whats causing problem
I'm runing [EMAIL PROTECTED] beta6 and I have a problem with registration of SIP phone.
I can't find/replicate when exactly its happends but sometimes after server restart or phone restart one of the phone can't register and I get this in the server:
Transmitting (no NAT) to 10.1.1.152:5060:SIP/
I have a client who is unable to register her SJPhone on my Asterisk
server. She is using a Westell DSL router connected to Verizon. Others
in her group, using cable modems, are able to register. The group is
located in the Dallas area.
Is Verizon still blocking SIP registrations?
Is there
Hi List,
I am very new to asterisk. I downloaded asterisk from CVS head yesterday and compiled it in Redhat linux 9.
I created a sip account for testing and configured it in the Firefly.
While Firefly try to connect to the asterisk server i am getting an
error
as below and failing the registrati
Hello.
Is there any know issue with Asterisk 1.0.9 concerning intermittent SIP
registration issues.
My SIP hard phone (aastra 9133i) and soft phone (xlite) keep losing
registration so calls to them go direct to VM although calling to other
phones from them works fine.
The logs show 'Tran
Hi,
We get this problem with some of our overseas SIP clients
that they get un-registered from our Asterisk server, but still able to make
calls. After sometime they get registered again. What can be causing this
problem, any ideas?
Zeeshan
Hi list,
I'm in central-europe and signed yesterday a broadvoice account. My
Asterisk box is CVS 2005-08-25.
Problem I face is:
"Failed to authenticate on REGISTER to '[EMAIL PROTECTED]'
(Tries 2)" then
"Registration for '[EMAIL PROTECTED]' timed out" and finaly
"Giving up forever to regist
I updated 2 weeks ago and am due to update again...
So Yes I will update
It seems that the giving up forever feature is by design,
As I had seen a post about it awhile back...
But I would rather not have asterisk give up (forever) if it can't
see a sip server.
I feel retries should certainly
On Wed, 24 Aug 2005, Steve Gladden wrote:
> I'm looking for some help in how to keep asterisk from doing this.
> If we loose Internet or routing to our upstream provider even for only a
> few short minutes asterisk quickly gives up & never tries again.
> I have to do a manual reload to get it to
Steve Gladden wrote:
You also want to look at the "registertimeout" and "registerattempts"
>
>
> Yes!!!, thank you VERY much this is what I needed.
> Where are these options documented at?
> I'm guessing the source code?
> Or is there a better place to find this stuff?
>
> A search on the wi
>>>You also want to look at the "registertimeout" and "registerattempts"
Yes!!!, thank you VERY much this is what I needed.
Where are these options documented at?
I'm guessing the source code?
Or is there a better place to find this stuff?
A search on the wiki for "registertimeout" or "registerat
On 8/24/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote:
> Try using IP addresses instead of hostnames in sip.conf. Asterisk's DNS
> support is supposed to be improved in CVS-HEAD, but you should still try it.
>
> However, using an IP address instread of a hostname in your host= line
> c
Steve Gladden wrote:
I'm looking for some help in how to keep asterisk from doing this.
If we loose Internet or routing to our upstream provider even for only a
few short minutes asterisk quickly gives up & never tries again.
I have to do a manual reload to get it to register with my
sip provider
I'm looking for some help in how to keep asterisk from doing this.
If we loose Internet or routing to our upstream provider even for only a
few short minutes asterisk quickly gives up & never tries again.
I have to do a manual reload to get it to register with my
sip provider(s) again before incomi
On 17:01, Sat 16 Jul 05, [EMAIL PROTECTED] wrote:
> Hi,
>
> Quoting Michiel van Baak <[EMAIL PROTECTED]>:
>
> > On 16:32, Sat 16 Jul 05, [EMAIL PROTECTED] wrote:
> > >
> > > The error on the console is:
> > > Jul 16 11:29:20 NOTICE[3361]:-- Registration for
> '[EMAIL PROTECTED]'
> > > timed o
Hi,
Quoting Michiel van Baak <[EMAIL PROTECTED]>:
> On 16:32, Sat 16 Jul 05, [EMAIL PROTECTED] wrote:
> >
> > The error on the console is:
> > Jul 16 11:29:20 NOTICE[3361]:-- Registration for
'[EMAIL PROTECTED]'
> > timed out, trying again
> > Jul 16 11:29:21 WARNING[3361]: Forbidden - wrong
On 16:32, Sat 16 Jul 05, [EMAIL PROTECTED] wrote:
>
> Hi everyone,
>
> I have a number of SIP registrations going fine, but am trying to get a new
> provider going, and they have no sample Asterisk SIP config. They have been
> helpful, but keep falling back to the way they "think" packets should
Hi everyone,
I have a number of SIP registrations going fine, but am trying to get a new
provider going, and they have no sample Asterisk SIP config. They have been
helpful, but keep falling back to the way they "think" packets should be
flowing,
and I've been trying to figure out how the Asteris
I have set up realtime for Asterisk just as the instruction provide. Everything works, except it apearer that SIP devices do not regisert correctly. I can place a call from a SIP device, but not place a call to a SIP device.
If a I use sip.conf everything seems to work. I have not posted all
1:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SIP registration behind Linksys WRT54G - URI?
I think I'm getting closer to figuring this out ...
I just tried Linksys PAP2 and it registered just fine. I looked at the SIP
packets c
o: 'Pedro'; 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SIP registration behind Linksys WRT54G
Yes that's the first thing I tried ... I'm able to make it work (using
different routers than Linksys) in the following ways:
- S
re.
Tomas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pedro
Sent: Saturday, April 23, 2005 10:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
Have you tried to enable NAT tran
ECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo
> Sent: Saturday, April 23, 2005 8:48 PM
> To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
>
> Oh yeah, duh.. Forgot.. I also have an S
on
Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running
behind my Linksys WTR43GS with no issues. This is at home registering to an
external * box and to vonage.
- Original Message -
From: &quo
List - Non-Commercial Discussion"
Sent: Saturday, April 23, 2005 9:41 PM
Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
The WRT54G work fine...
I have a Sipura 1000 and a Grandstream 286, both nated through a
WRT54G on a single public IP. Worked "out of the box&qu
rom: "Rich Adamson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, April 23, 2005 10:24 PM
Subject: RE: [Asterisk-Users] SIP registration behind Linksys WRT54G
I've got a 7960 behind a Linksys wireless box and
The WRT54G work fine...
I have a Sipura 1000 and a Grandstream 286, both nated through a
WRT54G on a single public IP. Worked "out of the box" -- no special
settings needed. I was even surprised that I did not need to turn on
the NAT handling in the Sipura ATA.
Then I have a WRT54G running as a w
me other Linksys routers so I'm curious.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Scott
> Henderson
> Sent: Saturday, April 23, 2005 7:20 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject:
EMAIL PROTECTED]] On Behalf Of Scott
Henderson
Sent: Saturday, April 23, 2005 7:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
Please make sure you post any solution you find to this issue to the
list I have bee
risk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
Please make sure you post any solution you find to this issue to the
list I have been frustrated by this as well.
Scott
Please make sure you post any solution you find to this issue to the
list I have been frustrated by this as well.
Scott Henderson
Finite Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Phone: 907.339.
Hello,
I'm having some major problems getting SIP phones to register whenever I put
them behind a Linksys router. The same phones will register behind any other
NAT (I've tried 3 others without problems)
I've been debugging using Ethereal and these are the differences that I
found between Linksys
Title: AW: [Asterisk-Users] SIP registration fails
Hi Seshu,
that's where I started off. But most of them are not working (at least not for me).
My desired setup (for now) is very simple: SIP provider(web.de) <--> * <--> 2 SIP phones
But none of the examples explains
NAT Ip you need to have entries
like:
host=dynamic
Seshu
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William
MarksSent: Wednesday, April 13, 2005 10:57 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP
registration fails
Hello List ;)
I'm quite a
Title: SIP registration fails
Hello List ;)
I'm quite amazed by the features, asterisk offers but as I'm quite new to this stuff, I've got a few questions.
First of all the relevant part of my sip.conf:
cut sip.conf --
[general]
port = 5060
: [Asterisk-Users] Sip registration Problems With Zyxel
P2000W
Hi
I have a Zyxel P2002 (ATA) with this config.
Registration works but i cant call inn. Outgoing works fine.
Any clue?
Thore
- Original Message -
From: "Paul Dracevich" <[EMAIL PROTECTED]>
To: "'A
Sent: Sunday, April 03, 2005 6:51 AM
Subject: RE: [Asterisk-Users] Sip registration Problems With Zyxel P2000W
Hi ya I have also three of these phone, here is my entry in my sip.conf
[4701721]
type=friend
username=4701721
secret=password721
host=dynamic
canreinvite=no
context=internal
disa
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