On Wed, 2013-01-30 at 14:10 -0700, Carlos Alvarez wrote:
On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote:
Thanks - I was hoping there was some silver bullet to use out
there. Thanks
anyway.
There is. If you build a reliable network, the phones
2013/1/31 Ishfaq Malik i...@pack-net.co.uk
On Wed, 2013-01-30 at 14:10 -0700, Carlos Alvarez wrote:
On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote:
Thanks - I was hoping there was some silver bullet to use out
there. Thanks
anyway.
There
On Thu, Jan 31, 2013 at 3:08 AM, Leandro Dardini ldard...@gmail.com wrote:
2013/1/31 Ishfaq Malik i...@pack-net.co.uk
On Wed, 2013-01-30 at 14:10 -0700, Carlos Alvarez wrote:
On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote:
Thanks - I was hoping there was some
If you have no NAT or dynamic IP in your network, you can just
remove the registration process and assign to each peer its IP
address.
This is the answer. If 100% availability is critical, your IP
addresses shouldn't be changing anyway, so take the registration
process
On Thu, Jan 31, 2013 at 9:45 AM, joachim zoach...@securax.org wrote:
If you have no NAT or dynamic IP in your network, you can just remove
the registration process and assign to each peer its IP address.
This is the answer. If 100% availability is critical, your IP addresses
This is the answer. If 100% availability is critical, your IP
addresses shouldn't be changing anyway, so take the registration
process out entirely.
This advice is not valid for android / iphones though.
That's absurd. Why would you use a battery-powered smartphone if you
Hello. I am aware that 'sip show peers' will display my peers, and that 'sip
unregister ' (where is the peer name) will unregister a peer - however,
I want to force registration of a peer from the CLI.
Is there any way to force this? I have several user agents and I want to achieve
On Thu, Jan 31, 2013 at 9:26 AM, Adam Moffett adamli...@plexicomm.netwrote:
Maybe it's possible to send a NOTIFY to a peer on the last IP it was seen
at? I don't think I've seen anything that has a register command, but
lots of devices can get a check your config or reboot command via SIP
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of joachim
Sent: Thursday, January 31, 2013 11:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip register peer (the quest for near 100%
availability)
This is the answer. If 100
Is there any way to force this? I have several user agents and I want to
achieve
near 100% availability for all peers. I realise that the peer will be 'woken'
up
at my qualify intervals, but can I actually force registration from the CLI?
For those peers which are at known, fixed,
Hello. I am aware that 'sip show peers' will display my peers, and that 'sip
unregister ' (where is the peer name) will unregister a peer -
however,
I want to force registration of a peer from the CLI.
Is there any way to force this? I have several user agents and I want to
achieve
On 01/30/2013 11:26 AM, XBrian wrote:
Hello. I am aware that 'sip show peers' will display my peers, and that 'sip
unregister ' (where is the peer name) will unregister a peer -
however,
I want to force registration of a peer from the CLI.
Is there any way to force this? I have
I am aware that the direction is from peer to asterisk. Its
a valid question. If a solution did exist, guarantees near 100 per cent
availability. Especially if the device is actually there.
--
_
-- Bandwidth and
You can just shorten the time the phone device register on the asterisk
server. It is up to the peer to send the registration command. It cannot be
triggered or forced in any way.
Leandro
2013/1/30 XBrian bobo...@yahoo.co.uk
I am aware that the direction is from peer to asterisk. Its
a valid
Thanks - I was hoping there was some silver bullet to use out there. Thanks
anyway.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote:
Thanks - I was hoping there was some silver bullet to use out there. Thanks
anyway.
There is. If you build a reliable network, the phones will simply never
have a problem. We've got customers with phones that have never
Hi all,
question about register refresh time.
One of our supplier had a maintenance work on sat 4 Aug which was
replacing the production server for an Asterisk 1.4 running version.
We have few Asterisk connected to them (version 1.6, 1.8 and 1.10) with
register Username and Passwd. After
I'll check this option and see if it helps next time,
just to clarify, there were no actual calls in place, just DOS register
attack.
On Wed, Jun 1, 2011 at 12:22 PM, Ira i...@extrasensory.com wrote:
At 10:56 AM 6/1/2011, you wrote:
Do you have:
sip.conf
[general]
allowguest=no
So
Also you guys may need to use:
sip.conf
[general]
allowguest=no
*alwaysauthreject = yes*
On Thu, Jun 2, 2011 at 1:01 PM, Al lists asteris...@gmail.com wrote:
I'll check this option and see if it helps next time,
just to clarify, there were no actual calls in place, just DOS register
attack.
On 11-05-31 06:24 PM, Al lists wrote:
Hi List
Recently i have noticed this attack on couple of servers,
usually a foreign IP starts sending tons of register request without any
answer to authentication,
if you type sip show channels in cli you will see tons of these:
1.2.3.4 (None)
At 10:56 AM 6/1/2011, you wrote:
Do you have:
sip.conf
[general]
allowguest=no
So because of this I decided to type sip show channels into
my Asterisk and got this:
Peer
User/ANR Call
ID
Format Hold Last Message Expiry
Peer
216.xxx.69.xxx (None)
f2d8db55-0a7edd (nothing) No Rx:
OPTIONS
guest
Hi List
Recently i have noticed this attack on couple of servers,
usually a foreign IP starts sending tons of register request without any
answer to authentication,
if you type sip show channels in cli you will see tons of these:
1.2.3.4 (None) 2389603298 00101/1 0x0 (nothing)
Hello,
I define SIP registrations as follow in sip.conf :
register = number:passwd@sip-server
example :
register = 33:mypass@ip_sip_server
But apparently the SIP 'contact' header in the SIP REGISTER looks like
this :
/Contact: sip:s@ip_my_asterisk/
How come ? And how to change
On 4/4/11 5:13 PM, Jonas Kellens wrote:
I define SIP registrations as follow in sip.conf :
register = number:passwd@sip-server
example :
register = 33:mypass@ip_sip_server
But apparently the SIP 'contact' header in the SIP REGISTER looks like
this :
/Contact: sip:s@ip_my_asterisk/
The SIP trunking service that I am trying to set up keeps saying that my
registration from Asterisk is invalid.
Asterisk registration:
REGISTER sip:{registration_ip} SIP/2.0
Via: SIP/2.0/UDP {asterisk_ip}:5060;branch=z9hG4bK5c2eb10c;rport
Max-Forwards: 70
From:
Hi all,
I'm facing an issue with my asterisk server when an extension (X-Lite
softphone) tries to register on it...A huge amount of packets is exchanged
between endpoint and asterisk server while the X-Lite is online...Even when
I sign out from X-Lite, the asterisk server continues sending packets
These are requests where one endpoint pings the other to check if it
is still alive.
What is the problem?
michel freiha wrote:
Hi all,
I'm facing an issue with my asterisk server when an extension (X-Lite
softphone) tries to register on it...A huge amount of packets is
exchanged between
Dear Alex,
The problem is that the asterisk server is sending these packets
continuously with no stop and with a negligible duration between packets for
the same extension...My Asterisk server read the extensions from the
database and not from extensions.conf...There is a field in the sip buddies
By default, the interval at which the qualify pings are sent is, indeed
quite low.
There is no consequence to disabling it except for the obvious
implication that Asterisk then has no way way of knowing if the peer is
dead without first trying to reach it, every time and with every request.
You changed your default SIP (bindport) port to 5061 at the server, so
your client needs to look there.
Try like these
register = sipteszt:[EMAIL PROTECTED]:50/sipteszt
bindport=5061 ; UDP Port to bind to (SIP standard port
is 5060)
Adrià Vidal
Hi all,
We have two asterisk PBX. We would like to register it with SIP peer.
The client sends the register request. It gets back:
Jan 2 01:31:27 WARNING[186]: chan_sip.c:9760 handle_response_register:
Got 404 Not found on SIP register to service [EMAIL PROTECTED],
giving up
server:
I am trying to link an asterisk box up to a SIP server on the same
subnet. The SIP server does not have a password (and is locked down by
IP number 'allow'). How do I specify this on the register line?
Based on the documentation, the line looks like this:
register = user[:secret[:[EMAIL
I'm having trouble making calls over my VoIP provider. I do successfully
register, and when I try to establish a phone call Asterisk sends wrong
username and password. Instead of sending username and pass that I have
provided, he send username and pass of the SIP phone that is registered to *
-Commercial Discussion
Subject: [Asterisk-Users] SIP Register
I'm having trouble making calls over my VoIP provider. I do successfully
register, and when I try to establish a phone call Asterisk sends wrong
username and password. Instead of sending username and pass that I have
provided, he send username
Hi,
Two
questionsfor the gurus out here:
1)
I recently
asked, for a number of reasons, to have my provider changehis way of doing
SIP wth me: instead of registering with his server, I know simply send my
stuff to his IP without registration.
I have always had
two test numbers: one IAX
From the dump that I have attached It looks like the first attempt
at register does not work then followd by a second register which
then works.
This is happening on all the SIP phone attach to asterisk. The version
of asterisk here is 1.2.0b2.
Here is sip.conf for ext 204
[204]
username=204
Hello,
I have set up 2 different fwd.pulver.com accounts on my Asterisk. One will ring all my phones through one context, while the other account was set up to fool Nigerian scam artists, and will go directly to a special voicemail (after a few rings to give the impression of ringing a real
I did try this and did get it to register as this peer.
However inbound calls to that number are still coming into
the context defined in [general] sip.conf
I now have two numbers configured, the new peer as you sugested
and my original that just has the register line
without an associated peer
Yep thanks for the reply,
I figured out pretty quickly after one test that the /s did not work.
The issue remains that I have been unsuccessful in getting an incoming
call to come into any other context other than the one specified in
sip.conf [general] section
Anything I'm missing here?
I
Sorry this is a bit of a newbie question, I've been at this for a few
months and still have not quite figured this one out.
I've been able to setup one itsp (incoming calls) (sip account) with a
register line like this:
register = nnn:[EMAIL PROTECTED]
-or-
register = nnn:[EMAIL
Sorry this is a bit of a newbie question, I've been at this for a few
months and still have not quite figured this one out.
I've been able to setup one itsp (incoming calls) (sip account) with a
register line like this:
register = nnn:[EMAIL PROTECTED]
-or-
register =
Ok :)
--
From: Rich Adamson[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, October 10, 2005 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:Re: [Asterisk-Users] sip register incoming
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] sip register incoming call contexts?
Sorry this is a bit of a newbie question, I've been at this for a few
months and still have not quite figured this one out.
I've been able to setup one itsp (incoming
Steve Gladden wrote:
Sorry this is a bit of a newbie question, I've been at this for a few
months and still have not quite figured this one out.
I've been able to setup one itsp (incoming calls) (sip account) with a
register line like this:
register = nnn:[EMAIL PROTECTED]
-or-
Long list of questions to follow:
Short version:
Does the register line mate to a peer or is a register line totally
unrelated to a peer that is defined?
When using a register line does it have to refer to actual hostnames?
or can you refer to the peer name in the register line instead
of an
OK, I'm starting to get somwhere with this, I'm at least registering now..
however My inbound calls are still coming into the context defined
in [general] of sip.conf and not into the context I have defined
in my peer and extensions.conf
Here is what I have done:
IN sip.conf:
register =
OK, I'm starting to get somwhere with this, I'm at least registering now..
however My inbound calls are still coming into the context defined
in [general] of sip.conf and not into the context I have defined
in my peer and extensions.conf
Here is what I have done:
IN sip.conf:
Hi
I tested i can able to register 2 sip
phone by same user id and same phone number.
I need help to view there IP . i just
find one . not two of them, is there any command i can view both registration
IP.
Thanks.
Bashir
___
Asterisk-Users
Hi all * user
I did connected with * from 2 sip-softphone and i registered with asterisk
under same username and password and working both fine. but * shows only
one.
is there any way to find them both by using any tips.
Bashir
___
Asterisk-Users
@lists.digium.com
Subject: [Asterisk-Users] SIP register more then 1 with same username
Hi all * user
I did connected with * from 2 sip-softphone and i registered with asterisk
under same username and password and working both fine. but * shows only
one.
is there any way to find them both
Karl Brose wrote:
Is the SIPquest server sending the 401 Unauthorized message verbatim as
you printed it here?
I.e. is the WWW-Authentcate header broken up into several lines like that?
If so, how man spaces are actually at the beginning of each new line?
Continuation lines are allowed in SIP,
, November 16, 2004 8:29 PM
Subject: Re: [Asterisk-Users] SIP register problem
Just checked the RFC, and it does say that a tab is acceptable.
SIP header field values can be folded onto multiple lines if the
continuation line begins with a space or horizontal tab. All linear
white space
You're welcome.
It's been submitted.
Cyrille Demaret wrote:
Hi,
Thank you, it's working now!
Do you think that this patch will be included in the next cvs versions?
Sincerely,
Cyrille Demaret
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi,
I'm trying to register Asterisk with my sip provider
but I've a problem. Here's the log file :
REGISTER sip:sip.aquanta.com SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2810ee43
From: sip:[EMAIL PROTECTED];tag=as2e43c573
To: sip:[EMAIL PROTECTED]
Call-ID:
Is the SIPquest server sending the 401 Unauthorized message verbatim as
you printed it here?
I.e. is the WWW-Authentcate header broken up into several lines like that?
If so, how man spaces are actually at the beginning of each new line?
Continuation lines are allowed in SIP, but I think it's
PROTECTED] De la part de Karl Brose
Envoyé : mardi 16 novembre 2004 19:05
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] SIP register problem
Is the SIPquest server sending the 401 Unauthorized message verbatim as
you printed it here?
I.e. is the WWW
Just checked the RFC, and it does say that a tab is acceptable.
SIP header field values can be folded onto multiple lines if the
continuation line begins with a space or horizontal tab. All linear
white space, including folding, has the same semantics as SP. A
recipient MAY replace any
Dear all,
I'm trying to perform a register = userid:[EMAIL PROTECTED] for incoming
SIP calls from a provider and is not going too well. Pls refer to the
attached debug log and hopefully someone can help me out. I believe
that they're using Huawei equipment and the same userid/password pair
is
On a newly built Asterisk 1.02 system I am getting a rather strange
SIP register message ...
REGISTER sip:ispvoip-.ocn.ne.jp SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK3d1d0887
and further down ...
Contact: sip:[EMAIL PROTECTED]
Event: registration
the register directive in
On Sun, 07 Nov 2004 09:47:33 +0200, Gilad Ben-Yossef
[EMAIL PROTECTED] wrote:
I don't claim to understand the code at all, but what little I think I
understand from it makes me believe this is not the change you're
looking for.
The differences between chan_sip.c of the version on the
Hello,
--- Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote:
The provider's support staff says that the userid in 'From:
sip:[EMAIL PROTECTED] ...' should be the phone number while the
userid in 'Authorization: Digest username=userid...' of the same
REGISTER message should be the
On Sat, 6 Nov 2004 06:33:15 -0800 (PST), Girish Gopinath
[EMAIL PROTECTED] wrote:
AFAIK, the 050 in the From header acts as a display name. It can be used to
determine the processing rules by other SIP entities.
[SNIP]
The Auth credentials in the Request can be different.
Thanks, this
The syntax for the register command is
register=username:secret:[EMAIL PROTECTED]:port/extension
Benjamin on Asterisk Mailing Lists wrote:
On Sat, 6 Nov 2004 06:33:15 -0800 (PST), Girish Gopinath
[EMAIL PROTECTED] wrote:
AFAIK, the 050 in the From header acts as a
On Sat, 06 Nov 2004 12:00:57 -0500, Karl Brose [EMAIL PROTECTED] wrote:
The syntax for the register command is
register=username:secret:[EMAIL PROTECTED]:port/extension
Trouble is though that this does not have any effect on the username
in the digest. Whatever it is intended for, it's
Benjamin on Asterisk Mailing Lists wrote:
On Sat, 06 Nov 2004 12:00:57 -0500, Karl Brose [EMAIL PROTECTED] wrote:
The syntax for the register command is
register=username:secret:[EMAIL PROTECTED]:port/extension
Trouble is though that this does not have any effect on the username
in the digest.
On Sat, 06 Nov 2004 20:32:05 +0200, Gilad Ben-Yossef
[EMAIL PROTECTED] wrote:
Well, it looks like the digest is being built with authname in
build_reply_digest().
This seems to indicate that it was indeed the intend of the
implementor to use authname in the digest's username field and
On Sun, 7 Nov 2004 10:00:25 +0900, Benjamin on Asterisk Mailing Lists
[EMAIL PROTECTED] wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=0002802
Mark has fixed this in the most recent CVS (Wow! that was fast!).
However, I will need this for a production system that cannot be
upgraded
Benjamin on Asterisk Mailing Lists wrote:
It would seem that there is only a single line which has changed in
respect of SIP reigstration ...
*** static int transmit_register(struct sip_
*** 4054,4059
--- 4055,4061
if (!ast_strlen_zero(r-username)) {
I am trying to get Asterisk to register with a SIP provider who
officially only supports ATAs of the incumbent telephone monopolist
over here.
I have so far been lucky enough to get them to ***respond*** to my
requests for information on what parameters need changing in the
REGISTER messages in
And I would very much like feedback on those - are they useful?
If they are, I'll backport to chan_sip.
I find them useful (for the HTML based block LED field display of DeStar).
Today I even thought about writing a patch that sends the
available/unavailable messages from the quality=... code
And I would very much like feedback on those - are they useful?
Oh, I just found out by looking at the source code that there are database
entries SIP/Registry. I think the used database entries is something that
is currently under-documented ... :-)
At 9:41 AM +0200 on 7/23/04, Holger Schurig wrote:
And I would very much like feedback on those - are they useful?
If they are, I'll backport to chan_sip.
I find them useful (for the HTML based block LED field display of DeStar).
Today I even thought about writing a patch that sends the
Holger Schurig wrote:
And I would very much like feedback on those - are they useful?
Oh, I just found out by looking at the source code that there are database
entries SIP/Registry. I think the used database entries is something that
is currently under-documented ... :-)
If you simply type
The patch tries to send the time as well, but it fails. There are some
problems currently:
I applied the path from bug 2117. After this I got some events:
Event: SIPPeerStatus
Peer: weckhardt
Status: reachable
Time: 55
Event: SIPPeerRegistration
Peername: dnarotam
Status: Offline
I think we
Okay, I have finished my patch. With qualify=yes in sip.conf it looks
like this:
-
# telnet 127.0.0.1 5038
Trying 127.0.0.1...
Connected to localhost.localdomain (127.0.0.1).
Escape character is '^]'.
Asterisk Call Manager/1.0
Action: Login
Holger Schurig wrote:
The patch tries to send the time as well, but it fails. There are some
problems currently:
Seems like you are getting Time - how does it fail? Please explain...
/O
I applied the path from bug 2117. After this I got some events:
Event: SIPPeerStatus
Peer: weckhardt
Status:
Holger Schurig wrote:
I think we have several problems here. Once it's Peer:, the other time
it's Peername.
That's clearly a bug.
Also, I don't like the name of the event. It should just
be PeerStatus and PeerRegistration, because we might add something to
IAX2 as well. So I'd suggest to do
Holger Schurig wrote:
And while I was at this patch, I also changed the
Event: SIPRegistry
Domain: ...
Status: ...
to
Event: Register
Channel: SIP
Domain: ...
Status: ...
I still believe it would be better to call this Registry since that's a common
term across IAX and SIP for outbound
At 10:58 AM +0200 on 7/23/04, Holger Schurig wrote:
Okay, I have finished my patch. With qualify=yes in sip.conf it looks
like this:
-
# telnet 127.0.0.1 5038
Trying 127.0.0.1...
Connected to localhost.localdomain (127.0.0.1).
Escape character is
Hi John,
John Todd wrote:
At 10:58 AM +0200 on 7/23/04, Holger Schurig wrote:
Okay, I have finished my patch. With qualify=yes in sip.conf it looks
like this:
output snip
Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: Reachable
Time: 81
some more snip
Nicolas Gudino wrote:
On Fri, 2004-07-16 at 18:28, Matthias Endler wrote:
is it possible to receive SIP/IAX register and unregister
events via the
manager API (like in CLI)? I do receive all kinds of call events
(Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink).
Matthias Endler wrote:
As promised yesterday:
Anybody interrested can download the patch for Asterisk 0.9.1 at
http://matthiasendler.net/asterisk/patch/.
Great!
Please add it to the bugtracker in a .txt file created with
cvs diff -u channels/chan_sip.c
The diff has to be for CVS HEAD, that is
Hi Olle,
Nicolas Gudino wrote:
On Fri, 2004-07-16 at 18:28, Matthias Endler wrote:
is it possible to receive SIP/IAX register and unregister events via the
manager API (like in CLI)? I do receive all kinds of call events
(Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink).
Nicolas Gudino wrote:
On Fri, 2004-07-16 at 18:28, Matthias Endler wrote:
is it possible to receive SIP/IAX register and unregister events via the
manager API (like in CLI)? I do receive all kinds of call events
(Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink).
chan_sip2
Hi all,
is it possible to receive SIP/IAX register and unregister events via the
manager API (like in CLI)? I do receive all kinds of call events
(Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink).
My manager.conf looks like this:
[general]
enabled = yes
port = 5038
bindaddr =
Hi Matthias,
On Fri, 2004-07-16 at 18:28, Matthias Endler wrote:
Hi all,
is it possible to receive SIP/IAX register and unregister events via the
manager API (like in CLI)? I do receive all kinds of call events
(Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink).
chan_sip2
As I understand it, if I understand you correctly, the register
parameter is for the client side. The nat=yes parameter is for the
server side, so it has nothing to do with your register statement. The
sip debug displays no nat because sip.broadvoice.com is not behind
the nat, it's in front of
This may be a newbie SIP/NAT question. If so I am sorry. But any help
would be appreciated. My Asterisk server is behind an ipchains box and I am
trying to connect to Broadvoice. All works fine without the NAT. I have a
global nat=yes prior to my register, but the sip debug allows shows no
]
[mailto:[EMAIL PROTECTED] Behalf Of Micke
Andersson
Sent: Tuesday, February 17, 2004 9:05 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP REGISTER
Hiyas..
I have a little problem ..
I try to register my Asterisk at a sip provider.. but it just wont work.
It works fine with eg xlite
I put an externip=xxx.xxx.xxx.xxx in my sip.conf so I can register with FWD
from behind a NAT
With this entry my PSTN calls have a problem in that the other party cannot
hear me - I can hear them.
It does not matter whether I make the call or the other party does.
Any ideas ?
TIA
Simon
I am having an issue with registering SIP client w/ NAT. I have set
this up before on other boxes... But for some reason this one is not
acting the same... I have attached a sip debug from the registration...
For what ever reason it does not appear to be setting up the nat session
correctly
I was able to resolve the issue... Me being stupid...
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AstGrp
Posted At: Sunday, February 22, 2004 7:45 PM
Posted To: Asterisk User Group
Conversation: Sip Register Fail - NAT
Subject: [Asterisk-Users
John Fraizer wrote on the Tuesday, February 17, 2004 7:14 PM
Try sending us the registry line and context information from your
sip.conf. It is much easier to figure out what you're doing wrong
from there. I register my Asterisk server with 6 different SIP
providers with no problems at
On Wed, 18 Feb 2004, Micke Andersson wrote:
John Fraizer wrote on the Tuesday, February 17, 2004 7:14 PM
Try sending us the registry line and context information from your
sip.conf. It is much easier to figure out what you're doing wrong
from there. I register my Asterisk server
John Fraizer wrote on the Wednesday, February 18, 2004 6:16 PM
You've got a syntax problem. It SHOULD be:
register = pstn-number:[EMAIL PROTECTED]/pstn-number
Tried that too, no go..
I thought the syntax were:
register = username:passwd:[EMAIL PROTECTED]/local number
/Mike
You've got a syntax problem. It SHOULD be:
register = pstn-number:[EMAIL PROTECTED]/pstn-number
Tried that too, no go..
I thought the syntax were:
register = username:passwd:[EMAIL PROTECTED]/local number
Don't know about all the possible variations, but I'm using
Hiyas..
I have a little problem ..
I try to register my Asterisk at a sip provider.. but it just wont work.
It works fine with eg xlite or Grandstream.. .but not with Asterisk.
I think it is in the Register process:
This is the difference I cen tell in the sip headers between Xlite and
Try sending us the registry line and context information from your sip.conf.
It is much easier to figure out what you're doing wrong from there. I
register my Asterisk server with 6 different SIP providers with no problems
at all.
John
Micke Andersson wrote:
Hiyas..
I have a little problem
Hi,Key Aavoja,
Have you successfully registed to * with secret specificated?
Regards.
bfrac
- Original Message -
From: Key Aavoja [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 2:00 AM
Subject: [Asterisk-Users] SIP register/auth with Grandstream BudgeTone-100
Hello,
I have a problem with asterisk and Grandstream BudgeTone-100.
With default configuration everything works (in anonymous mode and fixed
IP), but if Im trying to enable registering, it dos not work.
I used 'sip debug' and verbose level 10, nothing happens if I switch
telephone on (no
1 - 100 of 111 matches
Mail list logo
