On Sat, Jul 2, 2016 at 12:59 PM, Leandro Dardini wrote:
> Hello,
> I am moving from realtime chan_sip to pjsip and one of the problem I am
> facing is the lack of "sipregs". With chan_sip, when an extension
> registers, the server where it has registered to is stored in
t;
> FROM: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] ON BEHALF OF Leandro Dardini
> SENT: July 2, 2016 2:59 PM
> TO: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> SUBJECT: [asteri
, 2016 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] Registration server with PJSIP
Hello,
I am moving from realtime chan_sip to pjsip and one of the problem I am facing
is the lack of "sipregs". Wit
Hello,
I am moving from realtime chan_sip to pjsip and one of the problem I am
facing is the lack of "sipregs". With chan_sip, when an extension
registers, the server where it has registered to is stored in sipregs.
Is there something similar in pjsip? How can I find on which server the
pjsip
I have two Polycom phone configured with Asterisk server, both use
transport=tls
My provision server is FTPS, I have phone5006.cfg & phone5007.cfg
If I enable transport tls on both phones I get the following error:
[2016-14:03:52] WARNING[26566]: chan_sip.c:14967 check_auth:
username
Not that I know. You could monitor the log file and generate a UserEvent (call
file or AMI command).
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Is it possible to detect the failure of an agent to register with Asterisk via
the AMI ?
When I try to register with Asterisk 1.4 using an invalid password I don't see
any event in the AMI, but see this in the messages log:
[2013-10-05 22:05:03] NOTICE[24598] chan_sip.c: Registration from
Hi all ,
I have managed to install and configure the
1. asterisk-1.8-current
2. dahdi-linux-complete-current
I did not faced any issues during the installation. After that I installed
X-Lite soft phone in two different PCs and tested the setup. every thing was
success. I was able make
you don't need register = string here, it only need you want asterisk
register to another sip proxy as client.
just remove that line and you should fine.
for X-lite or any other sip phone the user AlphaUser is sufficient.
On Fri, May 24, 2013 at 12:32 PM, luke devon luke_de...@yahoo.com wrote:
We get the same error with this version.
On Sun, Jan 1, 2012 at 6:13 PM, Matt Hamilton mistral9...@hotmail.comwrote:
I have a very basic setup where a UAC registers with Asterisk 1.8.7.2 -
both on the same subnet, no nat.
The following is the flow of messages:
1. UAC sends the registration
I have a very basic setup where a UAC registers with Asterisk 1.8.7.2 - both on
the same subnet, no nat.
The following is the flow of messages:
1. UAC sends the registration request
2. Asterisk responds with 401 Unauthorized with a new nonce
3. UAC sends a new digest with the nonce received
Hi All-
I have successfully routed calls into our asterisk system from several DID
providers in the USA, but for some reason I'm having a problem getting Vitelity
to work.
We are using the IAX protocol, and the symptom is that only about 50% of the
calls terminate properly into my
Hello list,
I often see the following in my message log :
[Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00
sip:00@MY-IP' failed for '184.106.109.168' - No matching peer found
[Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00
sip:00@MY-IP'
On 2 April 2011 09:46, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello list,
I often see the following in my message log :
[Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00
sip:00@MY-IP' failed for '184.106.109.168' - No matching peer found
[Apr 2 08:15:01]
On 04/02/2011 02:08 PM, Steve Davies wrote:
On 2 April 2011 09:46, Jonas Kellensjonas.kell...@telenet.be wrote:
Hello list,
I often see the following in my message log :
[Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00
sip:00@MY-IP' failed for '184.106.109.168' -
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: Saturday, April 02, 2011 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Registration from
Hi,
Do these IMSI names / numbers match what your phone is trying to register
as? Are there actual at the end of the numbers, or are you
attempting to obfuscate?
yes xxx are numbers (not real letters x), it's just 'obfuscation' and
anyway it's easier to recognize them by the first few
On Fri, Feb 25, 2011 at 3:01 AM, Axelle aaforti...@gmail.com wrote:
yes xxx are numbers (not real letters x), it's just 'obfuscation' and
anyway it's easier to recognize them by the first few digits.
and yes, they match the phone.
Show us. The error you're receiving specifically states that
Hi list,
Currently, one of my phones registers fine, and the other does not,
though for me they have the same config...
Can somebody help debug/understand why?
The logs in asterisk say:
[Feb 24 13:48:09] NOTICE[20626]: chan_sip.c:15642
handle_request_register: Registration from
On Thu, Feb 24, 2011 at 7:24 AM, Axelle aaforti...@gmail.com wrote:
Hi list,
snip
in /etc/asterisk/extensions.conf:
exten = 2102,1,Macro(dialSIP,IMSI2081) ; this one registers ok
exten = 2111,1,Macro(dialSIP,IMSI20830061) ; fails
These lines have nothing to do with endpoint
Of Zeeshan
Zakaria
Sent: Friday, September 17, 2010 5:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Registration attempts
It means that fail2ban is not configured correctly on your machine. For me
it works fine, and in fact lately these registration
I am getting several hundred registration attempts on my aserterisk per
minute. I have fail2ban installed but it's not stopping the attempts. Any
suggestions. Whatever they are using is changing the userid on each
attempt.
Latest IP: 209.172.57.219
Thanks,
Dave
--
I wrote a script to help with these here:
http://www.teamforrest.com/blog/171/asterisk-no-matching-peer-found-block
To each their own... there's 1000 ways of combatting this.
---fred
http://qxork.com
On Sep 17, 2010, at 5:18 PM, dave george wrote:
I am getting several hundred
It means that fail2ban is not configured correctly on your machine. For me
it works fine, and in fact lately these registration/hack attempts have gone
up significantly, thanks to cloud computing I guess.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-17 5:28 PM, dave george
jonas kellens wrote:
asterisk*CLI sip show domains
Our local SIP domains: Context Set
by
jocan.local (default)
[Configured]
192.168.1. (default)
[Configured]
asterisk*CLI sip show domains
Our local SIP domains: Context Set
by
jocan.local (default)
[Configured]
192.168.1. (default)
[Configured]
[Jun 26 17:49:03] NOTICE[5570]: chan_sip.c:15889
According to my IAX-provider, an account has been created for me on
their Asterisk-server...
But the Asterisk CLI tells me this :
asterisk*CLI iax2 reload
== Parsing '/etc/asterisk/iax.conf': Found
[Apr 30 20:51:30] NOTICE[6391]: chan_iax2.c:10124 set_config: Ignoring
bindport on reload
[Apr
Thanks for your help,
Unfortunatly neither the xx...@domain.com@domain.com nor the '
xx...@domain.com'@domain.com nor the xxx...@domain.com@domain.com worked
and when I try to do:
register = X:passw...@provider
[provider]
type=peer
host=domain.com
fromdomain=domain.com
2009/1/22 Laurent Bonny laurent.bo...@gmail.com
Hello,
I am trying to connect an asterisk 1.6 to a trunking plate forme. With
asterisk 1.4.x I added to sip.conf a line asking for registration in the
form of:
register =
Hello,
I am trying to connect an asterisk 1.6 to a trunking plate forme. With
asterisk 1.4.x I added to sip.conf a line asking for registration in the
form of:
register =
xx...@domain.com:Password:xx...@domain.comassword%3axx...@domain.com
@domain.com
Unfortunately, as you can see,
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Laurent Bonny
Sent: Thursday, January 22, 2009 9:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
sgueye@orange-ftgroup.com
Subject: [asterisk-users] registration problem using asterisk 1.6
Hello,
I am trying to connect
Maybe you can write your own patch that will allow this based on the
useragent somehow mapping it to 2nd peer based on the useragent? But
this feature is not there now.
What will happen when host=dynamic is the last registration will be
the one used, so if you have two SIP devices trying to
Hi,
Is there a way to limit only one registration for each user at a time?
meaning if a user tries to register, but that user is already
registered. i will deny?
or is it possible to for a single user at the same time, and when
someone calls that user, it will ring both phones?
Just want
Dear All,
I'm using a2billing interface with asterisk in order to bill all calls
flowing through my PBX... I need to prevent my customers to use the same
extension from different IP addresses so I created a new extension under
extensions.conf as follow:
[michofr]
type=peer
username=michofr
Would this be a firewall problem?
chan_sip.c handle_request_register: Registration from sip failed for
ACL error (permit/deny)
___
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asterisk-users mailing list
To UNSUBSCRIBE or
At 23:45 1/3/2008, Doug wrote:
Would this be a firewall problem?
chan_sip.c handle_request_register: Registration from sip failed for
ACL error (permit/deny)
Nope. I just needed to reload the configuration
so that the phone could register.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Probably you have deny=something instead of disallow=all.
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php
Dear Support,
I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with
PSTN line.
I have 3 extensions:
250 - my extension
998 - I configured as Line 1 in SPA-3102
999 - I configured as PSTN Line 1 in SPA-3102
I have created 998 and 999 to the user extension list of the
Hi,
x-lite has extensive debug facility you can turn that on in the advanced
options, that probably will give better understanding as what is going on
from x-lite side. i also have experienced the same but that involved
firewall and NAT issues.
Thanks,
Vivek
On 11/30/07, Newbie [EMAIL
Subject: Re: [asterisk-users] Registration state: Failed
Hi,
x-lite has extensive debug facility you can turn that on in the advanced
options, that probably will give better understanding as what is going on from
x-lite side. i also have experienced the same but that involved firewall
*Sent:* Saturday, December 01, 2007 11:34 AM
*Subject:* Re: [asterisk-users] Registration state: Failed
Hi,
x-lite has extensive debug facility you can turn that on in the advanced
options, that probably will give better understanding as what is going on
from x-lite side. i also have
: [asterisk-users] Registration state: Failed
well, then i would recommend to see full log in debug mode that might give
some clue. if you have not done this before you can uncomment line starting
with full= in the logger.conf... the log will be the usual
/var/log/asterisk/ directory
- Original Message -
*From:* Vivek Shrivastava [EMAIL PROTECTED]
*To:* Newbie [EMAIL PROTECTED]
*Cc:* Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
*Sent:* Saturday, December 01, 2007 11:50 AM
*Subject:* Re: [asterisk-users] Registration state
:50 AM
*Subject:* Re: [asterisk-users] Registration state: Failed
well, then i would recommend to see full log in debug mode that might
give some clue. if you have not done this before you can uncomment line
starting with full= in the logger.conf... the log will be the usual
/var/log
Hi,
I am using the OS which bundled with AsteriskNow
- Original Message -
From: Vivek Shrivastava
To: Newbie
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Saturday, December 01, 2007 12:25 PM
Subject: Re: [asterisk-users] Registration state: Failed
Hi,
Yes it make sense to have multiple registrars and to have SER acting as
a transparent proxy that forwards also REGISTER messages.
The question is: why it does not work! :-(
Regards,
Stefano
Stefano,
It is not Asterisk, It is SER (dispatcher module ?).
Why Asterisk is acting as
Stefano,
if you have distributed Registrars, which will keep the user location
of registration ?
And you do not need OpenSER to fwd Register message.. Register / Proxy
/ Redirect could be totally separate entities.
By the way, if you post the SIP Register message likely someone could help you.
Stefano,
It is not Asterisk, It is SER (dispatcher module ?).
Why Asterisk is acting as Register ? make sense use openSER as
Register/Proxy and Asterisk only Proxy and MG
Regards,
Giovanni
2007/11/19, Stefano Capitanio [EMAIL PROTECTED]:
Hi,
we a have a SER (OpenSER) in front of 2
Hi,
we a have a SER (OpenSER) in front of 2 real-time Asterisk.
SER simply forward SIP messages to 1 of the Asterisks:
UA -- SER -- Asterisk
We have a problem with REGISTERs:
Asterisk answers with 200 OK, but changes the Contact header, inserting
the IP of SER instead of the original IP
I can confirm that this problem occurs in the latest svn version (revision
87498) as well.
What's the best way to work with the developers to have this tracked down and
addressed? I assume it needs to become a bug report.
___
--Bandwidth and Colocation
@lists.digium.com
Betreff: Re: [asterisk-users] Registration of Snom 320 phone withAsterisk 1.4.13
Here are more details:
The phone and the Asterisk box are behind the same router (the Asterisk machine
is 192.168.0.2 and the phone is 192.168.0.4).
A ping command works:
[EMAIL PROTECTED]:~$ ping -c 10
On Mon, Oct 29, 2007 at 08:17:20AM +0100, Christian Stredicke wrote:
I guess the problem is that * sends the response to port 5060, while the
phone listens on port 2xxx for an answer.
That could be the problem.
The phone specifies port 2048 in its contact field. Is there a way to
configure
(not a response) to the phone.
CS
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Jason White
Gesendet: Montag, 29. Oktober 2007 09:16
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Registration of Snom
On Mon, Oct 29, 2007 at 09:22:21AM +0100, Christian Stredicke wrote:
Well, the response should go to the port number provided in the Via header.
If there is a rport set, then to that port. Everything looks good in the
log, the only problem is that the response is sent to the wrong port.
I
10:01
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Registration of Snom 320phonewithAsterisk 1.4.13
On Mon, Oct 29, 2007 at 09:22:21AM +0100, Christian Stredicke wrote:
Well, the response should go to the port number provided in the Via header
] Im Auftrag von Jason White
Gesendet: Montag, 29. Oktober 2007 10:01
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Registration of Snom 320phonewithAsterisk 1.4.13
On Mon, Oct 29, 2007 at 09:22:21AM +0100, Christian Stredicke wrote:
Well
On Mon, Oct 29, 2007 at 10:19:57AM +0100, Christian Stredicke wrote:
What you can still to is setting the port on the phone to port 5060 - just
as a little dirty workaround until there is a better solution available.
Would that be the sip_port settings entry? It is documented as for internal
On Mon, Oct 29, 2007 at 10:19:49AM +, Steve Davies wrote:
snom phones have been using ports in the 2000+ range since the dawn of
asterisk without any problems, so I suspect that this will be an
Asterisk configuration error, or a change to the asterisk SIP stack
that is causing problems.
Here are more details:
The phone and the Asterisk box are behind the same router (the Asterisk
machine is 192.168.0.2 and the phone is 192.168.0.4).
A ping command works:
[EMAIL PROTECTED]:~$ ping -c 10 192.168.0.4
PING 192.168.0.4 (192.168.0.4) 56(84) bytes of data.
64 bytes from 192.168.0.4:
Hello,
I am experiencing difficulty registering my Snom 320 phone with Asterisk
1.4.13, and have been receiving the same transport error messages on the
phone as described in this forum post:
http://forums.digium.com/viewtopic.php?p=40554highlight=sid=b6d7fd216103dcdafb0b995aff03f07f
Are there
Hello,
I have installed the latest beta of AsteriskNow on my machine.
Everything works fine except for one thing - my registration with
terminating peer times-out after some period of time.
I called my provider and they told that changing registration interval
should help.
I have once before
Hi,
I have configured a sip provider account, with register = user
:[EMAIL PROTECTED]/user, with Asterisk 1.4.2.
Then I start Asterisk, which register successfully to the sip provider: sip
show registry show me the provider and status Registered.
I do a sip reload in the CLI, and now
every few days my ADSL connection gets dropped for a few seconds. When
it does I find my SIP connection to one of my providers does not timeout
and retry. Does the following give some clues?
Asterisk 1.2.13, Copyright (C) 1999 - 2006 Digium, Inc. and others.
(note this is the debian etch/testing
What is the variable like $peerip to get the registered ip address for a
peer
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without express written
Am Dienstag, den 28.11.2006, 06:46 -0800 schrieb Khaled:
What is the variable like $peerip to get the registered ip address
for a peer
You can use ${DB(SIP/Registry/sip507)} where sip507 is the section
name as well as username from my sip.conf- no idea which of both to use,
try it out.
This
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sippeer
O`e`o`ed'oth Anselm Martin Hoffmeister [EMAIL PROTECTED]:
Am Dienstag, den 28.11.2006, 06:46 -0800 schrieb Khaled:
What is the variable like $peerip to get the registered ip address
for a peer
You can use
Sergio R. D'Ippolito wrote:
Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to
register a linksys 922 phone thru internet and when I make sip debug
command i see this debug information:
*/SIP/2.0 401 Unauthorized/*
/Via: SIP/2.0/UDP
2006/10/31, Jon Farmer [EMAIL PROTECTED]:
Sergio R. D'Ippolito wrote: Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information:
*/SIP/2.0 401 Unauthorized/* /Via: SIP/2.0/UDP
firewall? i dont think so because sometimes the phone can register ok
and sudendly the appears unregistered
Leonardo Silva [EMAIL PROTECTED] ha escrito:
2006/10/31, Jon Farmer [EMAIL PROTECTED]:
Sergio R. D'Ippolito wrote:
Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I
Hi all, i have an * version: Asterisk
SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and
when I make sip debug command i see this debug information:
-- SIP read from x.x.x.x:1024:
REGISTER sip:mysipserver.com
SIP/2.0
Via: SIP/2.0/UDP
Sir,
I installed asterix in some system say with ip 172.16.7.63.From some other windows system say 172.16.7.50 i am running an xlite and configured a user say 200 with proxy as 172.16.7.63.I
modified the sip.conf file with user [200].When i run xlite in asterix cli i am able to see the mesage
17 aug 2006 kl. 18.38 skrev Ferguson, Michael:
G'Day List;
I hoping for some direction here:
The following message is scrolling without end on my asterisk box,
continuously: (NOTE: date and time changes accordingly and IP
addresses are not real)
Aug 17 11:49:53 NOTICE[1034]:
Olle,
Thanks
,preciate it.
Best Wishes
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Friday, August 18, 2006 3:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Registration Error
G'Day
List;
I hoping for some
direction here:
The following messageisscrolling
without end on my asterisk box, continuously:
(NOTE: date and time changes accordingly and IP addresses are not
real)
Aug 17 11:49:53
NOTICE[1034]: chan_sip.c:8038 handle_request: Registration from
Hi all,
I wonder if there are 2 UAs having the same sip account and
password. If they both register to the same server in same time.
Both of them can register successfully and make calls. Am I right?
How can I prevent the above case, say only one UA can register to the
server? Please advice.
- Original Message -
From: unplug
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Fri, 28 Jul 2006 07:35:34 -0300
Subject: [asterisk-users] registration
process
Hi all,
I wonder if there are 2 UAs having the same
:34 -0300
Subject: [asterisk-users] registration
process
Hi all,
I wonder if there are 2 UAs having the same sip account and
password. If they both register to the same server in same time.
Both of them can register successfully and make calls. Am I right?
They can register to the same
:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Fri, 28 Jul 2006 14:14:56 -0300
Subject: Re: [asterisk-users] registration
process
That's mean, if I have your sip account and password, I can
connect/register to the sip server as you do
- Original Message -
From: unplug
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Sat, 29 Jul 2006 00:51:38 -0300
Subject: Re: [asterisk-users] registration
process
Thanks!
I think I can't restrict the access by IP
What about limiting the simultaneous SIP sessions for each subscriber to one or two?On 7/28/06, Joshua Colp [EMAIL PROTECTED]
wrote:- Original Message -From: unplug[mailto:
[EMAIL PROTECTED]]To: Asterisk Users Mailing List -Non-Commercial Discussion
- Original Message -
From: Christopher Aloi
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Sat, 29 Jul 2006 01:25:27 -0300
Subject: Re: [asterisk-users] registration
process
What about limiting the simultaneous SIP
Hi:I 've a question:I'm using [EMAIL PROTECTED];
I've seen the dialparties.agi , I want to do this;
I've one softphone and I want register it in 2 different Proxy;only X-lite permitted this, all others no;I want have more proxy with others softphone;I run asterisk - R and I've seen when a
-- From: "Steve Totaro"
[EMAIL PROTECTED]
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
asterisk-users@lists.digium.com
Sent: Monday, May 29, 2006 4:39 PM
Subject: Re: [Asterisk-Users] registration at Voipbuster times out
If the d
Hi, I am new here on
thislist, and have a problem of which I hope that somebody here can help
me with it. I have a Voipbuster account, with which I would like to make
phone calls via my Asterisk PBX. If I let X-Lite register directly at
voipbuster.com, everything is OK, but if I let
Remko Muis wrote:
Hi,
I am new here on this list, and have a problem of which I hope that
somebody here can help me with it.
I have a Voipbuster account, with which I would like to make phone
calls via my Asterisk PBX. If I let X-Lite register directly at
voipbuster.com, everything is OK,
Maybe a silly question but can you ping sip.voipbuster.com from your
asterisk box?
Second question and probably the answer, what is your dial statement in
extensions.conf?
Contact:sip:[EMAIL PROTECTED] EXTERN IP]
One way to test is to create a dial statement like this exten =
Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 29, 2006 3:43 PM
Subject: Re: [Asterisk-Users] registration at Voipbuster times out
Maybe a silly question but can you ping sip.voipbuster.com from your
asterisk box?
Second question and probably the answer, what is your dial statement
On Mon, May 29, 2006 16:20, Remko Muis said:
Hi Steve Attilla,
Thanks for the quick replies!!
Attilla: your suggestion sounds promising, since I know my system clock is
not too accurate. But that is the reason I use the network time protocol
daemon. Time and date settings are now correct.
Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 29, 2006 3:43 PM
Subject: Re: [Asterisk-Users] registration at Voipbuster times out
Maybe a silly question but can you ping sip.voipbuster.com from your
asterisk box?
Second question and probably the answer, what is your dial statement
,
Remko
- Original Message -
From: Francesco Peeters (Asterisk) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 29, 2006 4:32 PM
Subject: Re: [Asterisk-Users] registration at Voipbuster times out
On Mon, May 29
, May 29, 2006 4:39 PM
Subject: Re: [Asterisk-Users] registration at Voipbuster times out
If the domain resolves you are probably OK, they just dont reply to pings.
Type asterisk -r then type sip debug and even set verbose 15 and try
to dial. Post the relevant console output. Also, disable
: Steve Totaro
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 29, 2006 4:39 PM
Subject: Re: [Asterisk-Users] registration at Voipbuster times out
If the domain resolves you are probably OK, they just dont reply
.
Best, Remko
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 29, 2006 5:19 PM
Subject: Re: [Asterisk-Users] registration at Voipbuster times out
No. If you can
Remko Muis wrote:
Steve,
I will try that, but now I am at my office. Can I dial some number
from the command line ;-) ?
Thanks,
Remko
Not from the command line, but you *can* from the manager API...
(not that it matters now, as I'm sure you're home now, just like me G)
Have a look at
- Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] registration at Voipbuster times out
Steve,
I will try that, but now I am at my office. Can I dial some number from the
command line ;-) ?
Thanks,
Remko
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users
Josep Aguilar wrote:
Is it possible that voipbuster refuses to connect to asterisk?, perhaps
asterisk agent is blacklistet by them
Josep
Unlikely, as mine connects just fine...
--
Francesco
___
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do you have any problems with audio connection
mine connection is just fine but audio from voipbuster.com is poor
(breaking)
with any other SIP client audio is OK
fpeeters pravi:
Josep Aguilar wrote:
Is it possible that voipbuster refuses to connect to asterisk?, perhaps
asterisk agent is
Matic wrote:
do you have any problems with audio connection
mine connection is just fine but audio from voipbuster.com is poor
(breaking)
with any other SIP client audio is OK
fpeeters pravi:
Josep Aguilar wrote:
Is it possible that voipbuster refuses to connect to asterisk?, perhaps
, 2006 8:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] registration with different username
Well, I did, but the reason is still the same, if the username is
different from the phone number, asterisk rejects the registration :-(
Dovid Bender
Hello,
I am trying to register to the asterisk with different phone number,
login and password. This is my setting in the sip.conf:
[246079011]
type=friend
context=cisco
secret=XXX
host=dynamic
username=tomas
allow=alaw
nat=yes
canreinvite=no
mailbox=246079011
but I get this reply:
Mar 27
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