I have this Error Please Help me
loader.c: Error loading module 'cdr_radius.so': libradiusclient-ng.so.2:
cannot open shared object file: No such file or directory
--
_
-- Bandwidth
:31, Tue 11 Mar 08, NOC ph wrote:
>> Hi Mich,
>>
>> I added the following line for the RTP its still the same, I can hear
>> ring but no voice when answer from the other side. Any more ideas?
>
> Firewall rules look ok now.
>
> Like I said, did you set externip
ep state
pass in on $ext_if inet proto udp from any to any port 1:2 keep
state
pass in on bce0 proto tcp from $ipc to any port ssh flags S/SA keep state
pass in inet proto icmp all icmp-type echoreq keep state
pass in quick on bce1
Michiel van Baak wrote:
> On 07:00, Mon 10 Mar 08, NOC
Hi All,
I have an asterisk box on my DMZ, and I'm using a PF for my firewall, I
can make a call but some reasons I have a dead air.
Any Ideas? below are my rules...
ext_if = "bce0"
int_if = "bce1"
altitude = "172.16.1.0/24"
machines
vbox = "172.16.1.1"
uci = "172.16.1.4"
voices = "20
@lists.digium.com
Subject: Re: [asterisk-users] Newbie on VoIP
On Mon, Mar 03, 2008 at 10:14:02AM +0800, NOC Ph wrote:
> Hi Guys,
>
>
>
> I'm new in VoIP, I heard from a friend that asterisk is good in VoIP
service
> especially on SIP. I'm planning to replace o
Re: [asterisk-users] Newbie on VoIP
On Mon, Mar 03, 2008 at 10:14:02AM +0800, NOC Ph wrote:
> Hi Guys,
>
>
>
> I'm new in VoIP, I heard from a friend that asterisk is good in VoIP
service
> especially on SIP. I'm planning to replace our old PBX system (legacy of
&
n BSD discussion
Subject: Re: [Asterisk-bsd] Newbie on VoIP
On March 2, 2008 06:14:02 pm NOC Ph wrote:
> Hi Guys,
>
>
>
> I'm new in VoIP, I heard from a friend that asterisk is good in VoIP
> service especially on SIP. I'm planning to replace our old PBX system
> (legac
Hi Guys,
I'm new in VoIP, I heard from a friend that asterisk is good in VoIP service
especially on SIP. I'm planning to replace our old PBX system (legacy of
Panasonic) to VoIP so that even out of the country we can still communicate
cheaper than regular phone. But I have a few questions thoug
Hi,
I'm having problems installing codec g729 on my Asterisk that's running on
FreeBSD 6.0
codec_g729a.so module loads ok, but the register utility doesn't seem to
register the license key correctly, because when I issue "show g729" under
Asterisk's CLI it says that the command is invalid.
It doesn
Tobias Wolf a écrit :
Noc Phibee schrieb:
Hi
i read the list and see a lot of personn say T38 it's not possible
with asterisk and other says that he use T38 with asterisk ??
i don't understand ;=)
Well, if i understand it correctly then Asterisk currently only sup
Hi
i read the list and see a lot of personn say T38 it's not possible
with asterisk and other says that he use T38 with asterisk ??
i don't understand ;=)
He have a solution (commercial or free) to add T38 ?
I have :
Fax Machine --> Linksys PAPT --> Asterisk ===> IAX2 on Sdsl ===>
Asterisk
Hi
i have a big change or bproblems to update a asterisk 1.2.12 server to
asterisk 1.4.1 ?
Thanks bye
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digi
Hi
i have a big problems with my asterisk .. i use a Digium TDM400P for
connect a
analog line.
And not all time (i don't know why) he don't see the end of the call and
anyone can call me
(occuped)
For that's work, i am disconnect the phone cable and it's good
anyone have a idea ?
bye
___
Hi
anyone know if they have a solution in Cisco for:
1- Connect old PABX (with BRI or PRI) to a cisco router
2- Connect this cisco router in SIP to a Asterisk Server
I am search if cisco can this and what is the modele for this
Thanks ;=)
___
-
Hi
thanks for your answer,
for dtmfmode, all sip account have dtmfmode=rfc2833 ;=)
that's don't change
bye
Gordon Henderson a écrit :
On Fri, 9 Feb 2007, Noc Phibee wrote:
Hi
i have two problems with my Grandstream GXP2000 :
1- When i wan pickup a call, that's don&#x
Hi
i have two problems with my Grandstream GXP2000 :
1- When i wan pickup a call, that's don't work's (*8EXTEN)
and when i test whit Softphone, i have a error too, he say me
[EMAIL PROTECTED] not found ..
in features.conf, i have:
[general]
parkext => 700
Hi
it's possible to use a Digium TE110P Single T1 / E1 PCI Interface for supply
a E1 link to a old PABX ?
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://l
Hi
i use a lot of Grandstream GXP2000 with BLF
How to set up on the same key BLF blinking call interception?
So that someone is able to take a call that is destinated to another user
phone
Thanks bye
___
--Bandwidth and Colocation provided by Easyn
Hi
anyone have a sample of shorewall configuration for add a TC/QoS
on IAX2 traffic ?
Thanks for your help
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists
t; 100,3,Answer
exten => 100,4,Dial(SIP/220&SIP/221,30)
exten => 100,5,Hangup
exten => 200,1,Ringing
exten => 200,2,Wait,1
exten => 200,3,Answer
exten => 200,4,Dial(SIP/221,25,tm)
exten => 200,5,Hangup
;=)
Stefan Wintermeyer a écrit :
Hi,
Am 17.01.2007 um 15:07 schrie
Hi
I have two small question, if you can help me ;=)
Problems with Answer+Music
my extension:
[Cal-In]
exten => _81120,1,Goto(C-Internal,100,1)
exten => _81121,1,Goto(C-Internal,200,1)
[C-Phibee]
exten => 100,1,Ringing
exten => 100,2,Wait,1
exten => 100,3,Answer
exten
Hi
actually, i have only one Asterisk Server ;=)
Anyone know a how to for create a seconde asterisk in "Backup"
for hight availability ?
Thanks bye
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIB
Hi,
if i use System() or TrySystem() into my extensions.conf for execute a
external command, can i get and put the result of the command into a
variable ?
Thanks bye
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing li
Hi
actually, for call i use ZAP Channels on a E1 and SIP Account on a VoIP
provider ...
in Zap, we use group and we have:
exten => _1.,2,Dial(Zap/r1/${EXTEN:1},50,rt)
exten => _1.,3,Hangup
r1= he change of channels at all calls channel group 1
It's possible to create a
Hi
it's Colt-Telecom.
you have a TE405P ?
bye
pixiesfr a écrit :
Hi
what is your operator?
I have some pb on orange...
thx
Noc Phibee a écrit :
Hi
anyone have a idea for debug my digium TE405P card ?
My zaptel.conf:
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loa
Hi
anyone have a idea for debug my digium TE405P card ?
My zaptel.conf:
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone= fr
defaultzone = fr
My Zapata.conf:
[channels]
language=fr
context=from-E1
switchtype = euroisdn
pridialplan = unknown
signalling = pri_cpe
usecalleri
Hi
I don't see a answer to this question ;=) i am search this solution too ..
Thanks bye
Jea philippe a écrit :
Hi,
Actually on my setup all outgoing calls are going trhu a SIP unique
account
A have a second SIP account with another operator and I would like my
setup
to use alternatively
Hi
i use now iaxmodem for receive fax and that's work very good with
Hylafax ;=)
Do you know if we can sent fax using iaxmodem and Hylafax ?
when i test:
déc 13 13:47:21.12: [13725]: SESSION BEGIN 00014 330426690268
déc 13 13:47:21.12: [13725]: HylaFAX (tm) Version 4.3.0
déc 13 13:47:21.
Hi
i have a asterisk server with a Digium 4xE1 card connected to my local
operator.
I am search a How to for :
- Add a Mail to Fax server
- Add a Fax to Mail Server
thanks bye
___
--Bandwidth and Colocation provided by Easynews.com --
Hi
for put a "anonymous" clid on a out line sip, what is the config ?
thanks bye
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/
Hi,
i receive a call on my analog line but my asterisk don't answer ;=)
do you know if they hae a solution for know if the card see the call ?
for see if it's not my cable don't work ..
thanks bye
___
--Bandwidth and Colocation provided by Easynews.
Tzafrir Cohen a écrit :
* Use genzaptelconf from xpp/utils/genzaptelconf to save you from this
guesswork.
Hi,
thanks ;=) with genzaptelconf, now that's works ...
correct channel are put into zaptel.conf and zapata.conf
small question if you know the TDM400P: if the fxo module are
at the
Pranav Peshwe a écrit :
Hi,
Check your /etc/zaptel.conf and ensure that it has the right kind of
signalling set for the same channel number as that in you zapata.conf.
do : cat /proc/zaptel/1
and it should show channels and the effective signalling settings for
them.
If signalling does not app
Leo Ann Boon a écrit :
Noc Phibee wrote:
thanks for this information, but no change:
Nov 24 10:32:42 WARNING[6346] chan_zap.c: Unable to specify channel
4: No such device or address
Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to open channel 4: No
such device or address
here = 0, tmp
:42 ERROR[6346] chan_zap.c: Unable to register channel '4'
Nov 24 10:32:42 WARNING[6346] loader.c: chan_zap.so: load_module failed,
returning -1
Nov 24 10:32:42 WARNING[6346] loader.c: Loading module chan_zap.so failed!
Leo Ann Boon a écrit :
Noc Phibee wrote:
Thanks Giogio,
but no i do
Thanks Giogio,
but no i don't have this module
bye
Giorgio Incantalupo a écrit :
Hi Noc,
I had similar problem. Check If you have netjetpci module and try to
delete it...this solved my problem.
Giorgio Incantalupo
Noc Phibee wrote:
Hi
i have buy a Digium TDM400P with 1 fxo mo
Hi
i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect
my asterisk to a french analog line.
In my zaptel.conf, i have:
loadzone=fr
defaultzone=fr
fxols=3
when i load the module, i have in logs:
Nov 24 06:13:40 gw kernel: Freed a Wildcard
Nov 24 06:13:40 gw kernel: Zapata T
Hi
I have a small question on CDR Database:
It's used by billing software no ?
he have only one structure of data or they have multi structure with
more information
logged ? sample: cdr simple and cdr_extended
thanks bye
___
--Bandwidth and Coloc
Hi
thanks for your answer, no i don't have see this software because i
don't see
screenshot or demo ;)
Hermann Wecke a écrit :
Noc Phibee wrote:
after 2 mounth of search, i don't have see a billing solution
for my small business..
Not quite sure as I didn't rese
r new and used security items
http://www.bochterservices.com/?j=store&t=email_security
GOLD PLATING SERVICES
http://www.bochterservices.com/?j=plating&t=email
Noc Phibee wrote:
Hi
after 2 mounth of search, i don't have see a billing solution
for my small business..
i see only Adv
Hi
after 2 mounth of search, i don't have see a billing solution
for my small business..
i see only AdvancedVoIPBilling but i don't know if he can work's with
Asterisk.
I am search a billing software for the invoice of my custumers, no
Calling Card.
but i don't see a small and simple product
Hi
anyone know a list of external hardware supported by asterisk for
connect old Pbx to VoIP line ?
For supply Isdn BRI and PRI to my clients
thanks
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCR
Hi
actually, for out call, i use :
exten => _0.,1,Dial(SIP/out-l1/${EXTEN:1},50,rt)
exten => _0.,2,Dial(SIP/out-l2/${EXTEN:1},50,rt)
exten => _0.,3,Dial(SIP/out-l3/${EXTEN:1},50,rt)
exten => _0.,4,Hangup
can you say me with this config, if the first user call and use out-l1
th
Hi
For add a analog line to my asterisk, i want add a Dgium Fxo card.
but i want know a small information:
The quality of the call are good or not with this type of card ?
Thanks for your returns
___
--Bandwidth and Colocation provided by Easyn
Hi
anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ?
Actually my HandyTone 488 are connected to:
wan port to my lan
line FXO port are connected to my local analogic line
i want that when a call in by my analog line, it's sent to my asterisk
for other voip post can
Hi
do you know if they have "external Box" (not internal card) for
connect Analog Line and Pri/Isdn to asterisk for incomming and
outgoing calls ...
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSU
Thanks all for your answer ;=) i start test this week a2billing
Noc Phibee a écrit :
Hi
what is the best billing solution for Asterisk ?
With WWW manager interface for user can see the real invoice...
Thanks bye
Hi
what is the best billing solution for Asterisk ?
With WWW manager interface for user can see the real invoice...
Thanks bye
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options vis
Hi
a small question:
what is the best card for Asterisk for supply 2/4 BRI access to a old PABX ?
Thanks bye
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://list
anyone know this error ??
Noc Phibee a écrit :
Hi
today, i have a big problems with my asterisk ...
when i want call i have this error :
Sep 8 12:38:07 WARNING[28369]: chan_sip.c:1226 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 102 (Critical
Hi
today, i have a big problems with my asterisk ...
when i want call i have this error :
Sep 8 12:38:07 WARNING[28369]: chan_sip.c:1226 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 102 (Critical
Request)
Sep 8 12:38:07 WARNING[28369]: chan_sip.c:1243 r
yusuf a écrit :
Hi,
you dont have to/should'nt be using different SIP ports for each
phone. Its completely not needed. Also, you dont have/need to port
forward. Just open ports 5060 and 1000-2, on the box that
asterisk is running, and on your NAT router. Dont port forward.
Then in s
Hi
I am search a small information
- i use Asterisk on official IP without Nat
- My first VoIP phone are a Thomson 2030 on a NAT Network.
That's work very good.
But now, i want add a second phone, a Linksys SPA-941 on
the same network of the Thomson 2030 ...
My problems that i don't see a
Hi
anyone know where i can solve this problems ? :
Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping
extra frame of G.729 since we already have a VAD frame at the end
Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping
extra frame of G.729 since we alr
Hi
it's possible to create a group of outgoing dial ?
For exemple:
exten => _0.,1,Dial(SIP/voip1/${EXTEN:1},90,rt)
exten => _0.,2,Hangup
exten => _0.,1,Dial(SIP/voip2/${EXTEN:1},90,rt)
exten => _0.,2,Hangup
and when my user call, if voip1 are used, he use voip2
and use not the
Anyone have a idea ?
Noc Phibee a écrit :
Hi
when i want compile asterisk 1.2.11, i have this error :
make[1]: Leaving directory `/usr/src/asterisk-1.2.11/stdtime'
cd editline && unset CFLAGS LIBS && test -f config.h || CFLAGS="-O6"
./configure
loading ca
Hi
when i want compile asterisk 1.2.11, i have this error :
make[1]: Leaving directory `/usr/src/asterisk-1.2.11/stdtime'
cd editline && unset CFLAGS LIBS && test -f config.h || CFLAGS="-O6"
./configure
loading cache ./config.cache
checking for gcc... gcc
checking whether the C compiler (gcc
Hi
a small question:
I have one Asterisk Server with:
VoIP Provider gateway for incomming/outgoing call
5 VoIP Phone
(i name it "Master")
i want add a another Asterisk server but only connected to:
5 new VoIP Phone
To the master for incoming/outgoing call (in g729)
It's
Hi
i have a small problems with my asterisk connected to phonesystems :
Now i have this message:
<-- SIP read from 62.39.136.151:5060:
SIP/2.0 403 Cant accept register from myself
Via: SIP/2.0/UDP 84.14.xx.xx:5060;branch=z9hG4bK38f74bd7;rport=5060
From: ;tag=as42b95c05
To:
;tag=e3fe971527b049a
Hi
on a new Asterisk installation, i have a small problems
with Asterisk and the VoIP Operator PhoneSystems.
Anyone have connected Asterisk to Phonesystems ?
I have this when i want call:
chan_sip.c:9696 handle_response_invite: Forbidden - wrong password on
authentication for INVITE to '"Jero
Martin Joseph a écrit :
On Jun 8, 2006, at 10:26 PM, BJ Weschke wrote:
On 6/9/06, Noc Phibee <[EMAIL PROTECTED]> wrote:
anyone have a answer at this question ?
Noc Phibee a écrit :
> Hi,
>
> Is it possible de tell asterisk to increase the volume?
>
> When we pl
Tzafrir Cohen a écrit :
On Thu, Jun 08, 2006 at 02:12:48PM +0200, Noc Phibee wrote:
Hi,
Is it possible de tell asterisk to increase the volume?
When we place or recieve a call the volume is very low, using a smartphone
or a hardphone.
What phone is it, exactly?
Thomson
anyone have a answer at this question ?
Noc Phibee a écrit :
Hi,
Is it possible de tell asterisk to increase the volume?
When we place or recieve a call the volume is very low, using a
smartphone
or a hardphone.
Thanks for advance
Hi,
Is it possible de tell asterisk to increase the volume?
When we place or recieve a call the volume is very low, using a smartphone
or a hardphone.
Thanks for advance
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mai
Hi
it's possible that send and receive (receive in priority) a fax with
Asterisk without card ?
I am very interessed by a solution for receive the fax, convert in pdf
and sent to email
Thanks for your help
___
--Bandwidth and Colocation provided
Hi
i renew my question ;=)
i have 8 phone number provided by my VoIP supplier :
081037XX0
081037XX1
081037XX2
<...>
For each, i have a login/password
where in put the registrer into my config ?
it's a "Trunk" on incoming no ?
i have put one register=> per nu
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noc Phibee
Sent: Monday, November 28, 2005 10:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?
Thanks sergio for your answer.
But cisco france say me that i
s that the solution are buy
new voip phone and put the 7910 in "Dead"
If anyone know a solution for get the latest firmware, mail me
Bye
Sergio Chersovani a écrit :
Noc Phibee ha scritto:
it's possible to upgrade the firmware of a cisco 7910 with asterisk ?
You need the le
Hi
anyone know if a Trunk SIP howto are created ?
I have 8 VoIP account with for all 1 login/pass per number.
i want add into my asterisk but not know where ;=)
Other questions:
my supplierhave a dns:sip.phonesystems.net
this name have 2 IP address
it's not a problems for Asterisk that he
Hi
it's possible to upgrade the firmware of a cisco 7910 with asterisk ?
he have a other solution for upgrade it without callmanager ?
thansk for your help
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNS
Hi
i have buy a used Cisco Phone 7910 for use with my asterisk.
The firmware version are 3.2(2.8), it's good for connect to asterisk ?
For update the fiormware, where i can get a new firmware ?
thanks bye
___
--Bandwidth and Colocation provided by Ea
Dear All,
The home page already move to the top, you can try again.
Cary LEUNG
Network Operator
Hong Kong VOIP exchange Network
引用 Glynn Condez <[EMAIL PROTECTED]>:
> What happened to your website. I am trying to open it but its empty.
>
> regards
>
>
> - Original Message -
> From: <[E
Dear All,
Thank for your visit our site, I found some users can not read our home page
from some browser, I will move all the pgae to the top directory later.
I had some idea, do you agree?
I want to setup a voip provider group to share the local PSTN connection, every
member must provide at l
Dear All,
I had setup a server to be Hong Kong VOIP Exchange gateway, do you want to join
us, you can find the detail at
http://www.voiphk.net
Thank You.
Cary LEUNG
Network Operator
Hong Kong VOIP Exchange Network
___
Asterisk-Users mailing list
[EM
I just checked on Cisco software center ... There is no 2.17 version ...
The latest one, which was released on Oct 22,2003 is 2.16.1, file name :
ata18x-v2-16-1-030709a-2.zip
Do you need one ?
Regards,
Alexander
http://asterisk.xv
Patrick,
We have started unofficial Asterisk Forums : http://asterisk.xvoip.com
I think such web-presence will help all parties to participate and exchange
information on all cases, starting from SS7 compatibility, ending with
business solutions, proposals, RFQ, etc.
I had this morning quick chat
David,
We has successfully tested last week interoperability between Asterisk and
our SIP softswitch.
We can definately help you with your project. Our company is New York
based, so it will be very easy to get interconnection with you. We are mostly in
wholesale business.
Also you can ge
Robert,
If we can try, it will be great.
If your Asterisk has Public IP, I think we can make registration between our
systems.
I need interconnect via SIP or H323. Our NExtone softswitch is very
powerfull solution.
If you have time to try , please tell me how to contact you .
My ICQ: 2851311
MSN
We just made interconnection from NAT-ed Asteriks to our softswitch. And
everything worked well... maybe because of our softswitch ..which is really
powerfull. We can test with you to see if it is a case.
Let me know.
Alexander
- Original Message -
From: "rnc Info Lists" <[EMAIL PROTECTE
I think it can help you.
http://www.zebraroaming.com/stuff/X-Lite-and-Asterisk.pdf
Let me know how it will go.
Thanks,
Alexander
ICQ: 2851311
- Original Message -
From: "Phillip Jackson, Director of IT" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, October 24, 2003 2:29 P
Hello All,
We are looking to test interoperability between
Asterisk and Nextone softswitch.
Please let me know who is wishing to participate.
We will open free US Long distance service for testing.
Please email me for more details and to be added to
testing participants.
To qulaify
Hi,
I'm considering giving the Grandstream BudgeTone-102 phones a try. I've been using
Cisco 7960's to date, but the low cost of the Grandstream phones are hard to ignore.
I have two questions:
1) Does the message waiting indicator on the BudgeTone's work with Asterisk?
2) The one line 12-di
-- Original Message --
From: Tilghman Lesher <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
Date: Thu, 18 Sep 2003 15:20:54 -0500
>On Thursday 18 September 2003 14:37, noc wrote:
>> 2) When listening to messages with VoicemailMain2, th
-- Original Message --
From: Steven Critchfield <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
Date: Thu, 18 Sep 2003 14:55:04 -0500
>On Thu, 2003-09-18 at 14:37, noc wrote:
>
>I don't use VM2 yet, but lets see if I can answer a
Hello,
I recently started playing with voicemail2. I'm having two minor problems that I
can't seem to find discussed in the archives.
1) New message 0 in mailbox 7606. New voice mail message count seems to start with 0
for the first new message instead of 1. Any tricks to fix this?
2) When
85 matches
Mail list logo
