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
-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 registration Asterisk
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 not looked at the log files, but often times DSL routers may use PPPoE
which has a little bit of overhead so you need to set the MTU below the default
of 1500. Some info about the issue can be found here:
http://www.ezlan.net/PPPOE.html and
http://www.cisco.com/en/US/tech/tk175/tk15/tech
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
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
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
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
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
I have seen this issue where there were internet connectivity issues. Asterisk
registers every so often with the ITS. For some reason or another (it can be
many reasons such as DNS, internet, ISP has issue etc). asterisk cant
re-register so it keeps trying.
As far as the so context if you have a
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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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
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
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
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
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.
Title: SIP registration fails
You may better look at example sip.conf files you will
be able to find on WIKI as there appears to be several incosnsistencies in your
sip.conf.
My suggestion is get rid off what you dont need and use
only those what is barely essential.
When you are using NAT
sterisk Users Mailing List - Non-Commercial Discussion'"
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
> u
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
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
disallow=all
allow=g729
dtmfmode=rfc2833
qualify=4
permit=0.0.0.0/0.0.0.0
[EMAIL PROTECTED]
-Original Message---
In the Grandstream setup, turn off "subscribe to message waiting
indication".
...or upgrade to CVS head, where I've fixed this problem with SUBSCRIBE.
Best regards,
/Olle
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I have this problem for 2 days and i dont understand
I am behind a nat
my sip.conf is:
[general]
port = 5060
bindaddr = 0.0.0.0
context = from-sip
disallow = all
allow= gsm
allow= ilbc
allow= ulaw
all
Alberto Martínez wrote:
Hello,
I am trying to register in asterisk with a softphone (x-lite) and I am
getting the following message:
Jan 19 13:27:01 NOTICE[3359]: chan_sip.c:7531 handle_request: Registration from 'tito
' failed for '192.168.1.5'
Just a guess, but the ip's don't match up.
[...]
I
I have tried uncommenting the section for xlite included in the sample
configuration file sip.conf and I can't register.
[xlite1]
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
regexten=1234
Scott Laird wrote:
First, what's in your extensions.conf? That controls the flow of calls
once they get into the system. There should be a context that has
extensions for 1001 and 1002, and sip.conf should direct calls into that
extension via a 'context =' line.
Indeed, I had not changed the e
On Nov 3, 2004, at 4:16 PM, Ben Greear wrote:
Hello!
I have a Grandstream and a Cisco SIP phone, and I'm trying to make
a call between them. I added this to my sip.conf:
; Grandstream
[1001]
type=friend
host=dynamic
; cisco phone
[1002]
type=friend
host=dynamic
First, what's in your extensions.con
I set up my own STUN server and turned reinvite
off.
Lyle
- Original Message -
From:
[EMAIL PROTECTED]
To: '[EMAIL PROTECTED]'
Sent: Tuesday, August 31, 2004 8:53
AM
Subject: [Asterisk-Users] SIP
registration with public dynamic ip address
Hi, I'm trying
On Tue, 20 Jul 2004 23:50:05 +0200, Andy Powell
<[EMAIL PROTECTED]> wrote:
> Hi,
>
> I've just (earlier today) updated from CVS so that I can apply the dtmf caller id
> patches. Unfortunately this has had an undesired effect.
I'm using * with an IX66 and no issues, with CVS head I suggest you
ha
>From the wiki...
http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone
"If you are having problems with the phone losing registration periodically,
make sure that "SUBSCRIBE for MWI" is set to "No" in the phone's
configuration. This applies to at least version 1.0.4.55, possibly othe
EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Richard Neese
Sent: Wednesday, June 09, 2004 7:07 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP Registration seems to timeout
try changing your codec to ilbc and make sure that his gs has the latest
flash
to suppo
try changing your codec to ilbc and make sure that his gs has the latest flash
to support it.
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Brian Rathman wrote:
I am using snom200 phones registering with Asterisk via SIP. I can see
where the phone registers without a problem, and then when you try and
make a call I get a proxy authentication required message on the phone
and failed to authenticate user error in the Asterisk messages
Title: Re: [Asterisk-Users] Wiki TOS - worrying for an open sourceproject?
To add
to the previous information I am receiving error messages on the phones that
say:
[5]28/5/2004 11:20:26: Match challenge for user=9991110035,
realm=asterisk,
valid=all:
[2]28/5/2004 11:20:26: Registrar
[EMAI
Karl Brose wrote:
This is also closely related to Asterisk SIP's lack of proper [user
section] authentication/recognition for incoming calls. We've seen a lot
of posts here where new users have problems with this, but the real
problem is usually not acknowledged.
So tell me what's wrong with th
No and Yes, Olle. But mostly NO.
What Asterisk is doing actually depends on how it is configured. If you
are, by design, accepting calls for a particular [user] through the
default context from the general section in sip.conf it will generate
the correct response, but this is not because aster
Karl Brose wrote:
If the response to an OPTIONS is generated by a proxy server, the
proxy returns a 200 (OK), listing the capabilities of the server.
The response does not contain a message body.
Allow, Accept, Accept-Encoding, Accept-Language, and Supported header
fields SHOULD be presen
for those who want to patch their SIP, here is a quck fix to make
Asterisk do a little better:
--- chan_sip.c 2004-05-16 01:33:06.0 -0400
+++ chan_sip.c_OPTIONS 2004-05-17 14:30:36.0 -0400
@@ -5916,6 +5916,7 @@
/* Initialize the context if it hasn't been already */
t: Re: [Asterisk-Users] Sip Registration Problem
Karl Brose wrote:
Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or
not, Asterisk doesn't do it correctly either.
The host should respond with 200/OK if the call >could< succeed
theoretically if it were an INVITE or else
>>>I removed the qualify lines and sip reload [ed]. The extension still
>>>showed up as "UNREACHABLE" instead of "UNMONITORED". I had to do a
>>>full restart to get it to stop sending the OPTIONS messages.
>>>What did I do wrong here? How can I make a change to qualify without
>>>restarting?
> If a
D] On Behalf Of Olle E.
Johansson
Sent: Tuesday, May 25, 2004 1:13 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Sip Registration Problem
Karl Brose wrote:
> Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or
> not, Asterisk doesn't do it correctly either.
&
Karl Brose wrote:
Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or
not, Asterisk doesn't do it correctly either.
The host should respond with 200/OK if the call >could< succeed
theoretically if it were an INVITE or else it should send a
404 or maybe a 487(? hmm, have to look)
It's a bug in Asterisk.
I believe it's still open also on the bugtracker. There are a few
reported senarios with these kind of problems.
Some of them where solved with the recent 'ast_gethostbyname' fix. Are
you running a recent version?
Btw, Ignoring OPTIONS is not a valid option (:-) whether s
Hi!
Registration only works if you have set "host=dynamic" for the client! In
case of a static host registration makes no sense, anyway! The only
purpose of registration is to tell the server at which IP address the
phone can be found.
Cheers, Philipp
Larry Keyes wrote:
Hi...I've got two Grandstream phones attached to my Asterisk on the same
subnet. The phones have fixed IP addresses. Asterisk is generated an error
for one of them only, even though both appear to be registered correctly.
The current state of the sip.conf is included below. Anyo
Larry Keyes wrote:
Hi...I've got two Grandstream phones attached to my Asterisk on the same
subnet. The phones have fixed IP addresses. Asterisk is generated an error
for one of them only, even though both appear to be registered correctly.
The current state of the sip.conf is included below. Any
- Original Message -
From: "SW" <[EMAIL PROTECTED]>
To: "[EMAIL PROTECTED] Digium. Com" <[EMAIL PROTECTED]>
Sent: Tuesday, December 16, 2003 1:47 PM
Subject: [Asterisk-Users] sip registration send out by asterisk
> Hi friends,
>
> I've noticed that first register message sent by * always
On Tue, 2003-09-30 at 20:21, Brian Capouch wrote:
> Does this imply that it will work even in a NAT environment?
>
> I have watched the list like a hawk for evidence of FWD working for
> machines placed behind NAT, but so far haven't seen that anyone could
> actually get it going.
>
> If so, t
Dave Cotton wrote:
I have SIP registrations working correctly for FWD and Sipphone, but it
is impossible to connect to Sipcall or Nikotel, I saw that someone on
the list has problems with ICH.
Does this imply that it will work even in a NAT environment?
I have watched the list like a hawk for ev
MAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, September 19, 2003 12:12 PM
Subject: Re: [Asterisk-Users] SIP registration between *'s
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
th another SIP server.
>
> That's the matter.
> - Original Message -
> From: "Jamie Carl" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, September 19, 2003 12:12 PM
> Subject: Re: [Asterisk-Users] SIP registration between *&
>
Sent: Friday, September 19, 2003 12:12 PM
Subject: Re: [Asterisk-Users] SIP registration between *'s
> Why?
>
> Use IAX2, it is s much better...
>
> J
>
> On Fri, 19 Sep 2003 11:54:23 +0200
> "Xisco" <[EMAIL PROTECTED]> wrote:
> >Hi ever
Why?
Use IAX2, it is s much better...
J
On Fri, 19 Sep 2003 11:54:23 +0200
"Xisco" <[EMAIL PROTECTED]> wrote:
Hi everybody,
I'm trying to SIP register between two asterisk, each one
have a Public IP. Asterisk told me that Unathorizae
In * one sip.conf
register =>usuario1:pass1@
In
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 18 September 2003 19:04, Hielke Christian Braun wrote:
> try to change [siptestphone] to [atrg613test] in sip.conf. Maybe
> that helps.
It didn't. And now something else is weird. Asterisk fails sending audio to my
SIP phone. Found this
>
> -Mensaje original-
> De: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] En nombre de Jan Janak
> Enviado el: viernes, 19 de septiembre de 2003 8:59
> Para: [EMAIL PROTECTED]
> Asunto: Re: [Asterisk-Users] SIP registration
>
>
> Hello,
>
> I don&#
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