1/25 David fire
>
> run
>> dahdi_cfg -
>> and send me the output.
>>
>> David
>>
>> 2009/1/25 broadband Voice
>>
>>> More Info
>>> [r...@newmark1 ~]# dahdi_cfg
>>> Notice: Configuration file is /etc/dahdi/system.conf
&
More Info
[r...@newmark1 ~]# dahdi_cfg
Notice: Configuration file is /etc/dahdi/system.conf
line 0: Unable to open master device '/dev/dahdi/ctl'
1 error(s) detected
[r...@newmark1 ~]#
On Sun, Jan 25, 2009 at 4:46 PM, broadband Voice
wrote:
> Yes, we have Digium Asterisk card in.
t load any modules.
> David
>
> 2009/1/25 broadband Voice
>
> More information
>>
>> service dahdi start
>> Loading DAHDI hardware modules:
>> FATAL: Module dahdi not found.
>> wct4xxp: [ OK ]
More information
service dahdi start
Loading DAHDI hardware modules:
FATAL: Module dahdi not found.
wct4xxp: [ OK ]
Error: missing /dev/dahdi!
[r...@newmark1 ~]#
On Sun, Jan 25, 2009 at 3:31 PM, broadband Voice
wrote:
> I had several pa
I had several panic attacks after upgrading to 1.4.22 but now we have no
dial tone on the T1. Urgent production system.
On Tue, Jan 13, 2009 at 5:13 AM, Benoit wrote:
>
> Personnaly, i had recently encountered a global machine check exception
> with
> two cards (TE220p and B410) and many kernel
lol
On Mon, Oct 20, 2008 at 3:34 PM, Sam Tam <[EMAIL PROTECTED]> wrote:
> VPN IP phone?
> Then firewall up the asterisk to disable any outside access and place the
> vpn server with the asterisk in a locked cabinet .
>
> Sure that will stop someone trying to physically listen to their call.
> Or
Is anyone seeing it in 1.4.22? I plan to upgrade to that version, my fear is
zaptel compatibility.
On Fri, Oct 17, 2008 at 1:08 AM, Yehavi Bourvine
<[EMAIL PROTECTED]>wrote:
> unfortunately I still see it in 1.6.0...
>
> __Yehavi:
>
> 2008/10/17 br
I am having a similar problem and I'm using Asterisk 1.4.19 and have that
problem on some calls through our calling card platforms. Someone suggested
we use 1.4.3 and have not tried it yet. Any comments from the group.
On Tue, Jul 29, 2008 at 1:19 AM, Yehavi Bourvine +972-8-9489444 <
[EMAIL PROTEC
You need to download a patch for zaptel, thats why your server is crushing.
Search through the forum, there is a known problem or reverse to a version
of Asterisk that is compatible with you zaptel.
On Tue, Oct 14, 2008 at 2:19 AM, Roberts Klotins <[EMAIL PROTECTED]> wrote:
> Hello there,
>
> Wit
Has anyone done a modification where you can Interrupt Asterisk's
SayDigits(). This will be helpful in order to be able to interrupt an
announce and dial digits without waiting to hear all the announcements.
___
-- Bandwidth and Colocation Provided by htt
Yes, this has already been answered. Search previous post for
implementation.
On Thu, Oct 9, 2008 at 3:34 AM, michel freiha <[EMAIL PROTECTED]> wrote:
> Dear all,
> Does asterisk supports H323?If yes how to enable it?
>
> Regards
>
> ___
> -- Bandwidth
Anyone using Tribox from Fonality. I understand its open source and free.
Can I use it for a call center functionality? Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
R
you are looking for a list of Call Center software packages that
> work with Asterisk then take a look here:
>
> http://www.voip-info.org/wiki/view/Predictive+dialer
>
> There are over 20 now I believe.
>
> MATT---
>
>
> On 10/1/08, broadband Voice <[EMAIL PROTECT
I stumbled upon this call center software that works with Asterisk calles
Aheeva. Does anyone else use it? Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now:
Can I used Aastra phones as agents instead of web-base on
astGUIclient-VICIDIAL suite: 2.0.4? Thanks. Our Asterisk is remote and call
center will be using Aastra phones or Linksys ATA.
On Mon, Dec 3, 2007 at 3:03 AM, Matt Florell <[EMAIL PROTECTED]> wrote:
> Hello,
>
> We've released another upda
Any responses please. I'm interested in this as well.
On Fri, Feb 8, 2008 at 2:51 AM, satish patel <
[EMAIL PROTECTED]> wrote:
> Dear all
>
>I am going to setup Asterisk Call center solution and i have
> setup my queue and agent i have 2 SNOM ip phone but when i call to queue my
>
:00|*|*|*?default|4456,1)
> exten => ,n,Dial(SIP/80001&SIP/RegS-1/02001|45|m)
> ;exten => ,n,Goto(ext-local,80001,1)
> ;exten => 2222,n,Dial(SIP/Inter232/X|75|mL(60))
> exten => ,n,Hangup
>
>
> Thanks
>
>
>
> On Tue, Jun 17, 2008
Linksys SPA-8000 is an 8 port looks great, is there something similar which
serves as a wireless router as well.
On Mon, Sep 29, 2008 at 3:42 PM, Brian Webster <[EMAIL PROTECTED]>wrote:
> What is the best-recommended resource for searching archives of this
> mailing list?
>
> Thanks for your tim
You need a billing software or calling card module with an IVR. You can
install A2billing in addition to Asterisk.
On Sat, Sep 27, 2008 at 6:06 PM, Babcock, Michael Alex
<[EMAIL PROTECTED]>wrote:
> hi;
> I'm a new member to this list and have a question for you all. I'm
> sure it's something simp
Hi,
I searched throug the forum and could not find an answer. Sorry if already
posted.
I'm using Asterisk 1.4.19 a i686 running Linux Centos. Sometimes when I call
and the person picks up, it continues ringing and I can hear the person say
hello then it hangs. Has anyone experienced this problem
corporate image. Vitelity tested
> >> very well in a very limited time frame. VoicePulse was great too but
> >> they kept making changes that resulted in outages, if engineered
> >> properly, there should be no outage short of an act of God.
> >>
> >>
I'm using Net2phone termination and the experience has been horrible for the
past 2 weeks, I have put in several tickets and nothing has been done. I get
a lot of congestion, channel unavailable and calls not going through. Does
anyone use them? I have been using SIP debug to try to resolve it but
I used A2billing and its pretty good.
On Sat, Jul 5, 2008 at 6:07 PM, Daniel Varella <[EMAIL PROTECTED]> wrote:
> Hello Kashif,
>
> Do you have something already working ? Here in Brazil I've worked
> on some projects using Asterisk to make some passive call-centers
> receive calls from their r
I can provide you termination to both countries, premium traffic, contact me
off the list.
On Mon, Jul 21, 2008 at 11:18 AM, MFH <[EMAIL PROTECTED]> wrote:
> Can anyone recommend decent quality as close to pay-as-you-go SIP
> wholesale termination providers in both Singapore and Sydney,
> Austral
,GotoIfTime(8:00-18:00|mon-sun|*|*?day_menu,s,1)
exten => 1866x,n,Goto(night_menu,s,1)
On Tue, Jun 24, 2008 at 6:35 AM, broadband Voice <[EMAIL PROTECTED]>
wrote:
> I googled some information on voip.org. Its my fault though and
> implemented the sample implementation without cre
I already checked them out. If you read their fine prints well they have
minutes limitations then you have to buy licenses. From the responses that I
got, I can get one the pci gsm cards with the drivers and that will work for
us except that it does not scale very well.
On Tue, Jun 24, 2008 at 10:
important) misconception, maybe
> the docs could be reworded. Do you remember what caused you to think
> that context was created automatically?
>
> broadband Voice wrote:
> > fc7234153*CLI> dialplan show open
> > There is no existence of 'open' context
> >
ROTECTED]>
wrote:
> On Mon, 23 Jun 2008, broadband Voice wrote:
>
> > I am trying to use the GotoIfTime function and get a busy signal. What I
> am
> > trying to accomplish is to have the system tell callers that we are
> closed
> > after 5:00pm. Here is the code be
I am trying to use the GotoIfTime function and get a busy signal. What I am
trying to accomplish is to have the system tell callers that we are closed
after 5:00pm. Here is the code below.
; If we're open, then go to the open context
; We're open from 9am to 6pm Monday through Friday
exten => 3200
Thanks all for the good feedback.
On Mon, Jun 23, 2008 at 7:19 PM, Michael Graves <[EMAIL PROTECTED]> wrote:
> On Mon, 23 Jun 2008 10:09:21 -0400, Steve Totaro wrote:
>
> >On Mon, Jun 23, 2008 at 9:57 AM, <[EMAIL PROTECTED]> wrote:
> >> The quad-band model is around $250 USD.
> >>
> >> See Ebay
I am thinking about using an existing asterisk box and turning it into a gsm
gateway. Has anyone tried this before, adding sonme gsm cards and an
antenna. Any ideas.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008
> > Syed Nasruddin
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
> > McGowan
> > Sent: Tuesday, June 17, 2008 5:45 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discus
Is anyone using Asterisk as a call center. I want to be able to set it up
for my office line, when calls come in after 7:00pm Est want a recording to
says the office is closed and have about 5 phones that I want to use as an
agent. Can anyone share their implementation? Thanks.
read the fine prints with them. Sometimes they have a thresh hold. Where you
cannot exceed 1500 minutes per month on the trunk. I use a T1 which is
cheaper if you compare it.
On Fri, Jun 13, 2008 at 4:52 PM, Jonn R Taylor <[EMAIL PROTECTED]>
wrote:
> I use bandwidth.com, works very well. 5 trunks
I am in Philadelphia, keep me updated and will try to make time to attend.
On Fri, May 23, 2008 at 10:25 AM, Dean Collins <[EMAIL PROTECTED]> wrote:
> Hey Adam,
>
>
>
> Yes I was thinking NYC - basically I was surprised at the lack of response
> about Ming from Voiceroute wanting to organize a p
Does anyone has termination to Togo and Cameroon? Thanks. You can send me a
private message if you do.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http:/
We have two servers but looks like G729 issues. Works fine on the old server
and not sure if it is T1 related. See SIP Debug. Any experiences to share.
Thanks
---
Newark1*CLI>
<--- SIP read from 194.xx.Xx.Xx:5060 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 76.xx.xx.xx:5060;branch=K784d
o a kernel panic.
>
> This is most likely the same problem so take a look at the forum
> postings and try disabling/enabling acpi in your grub startup.
>
> Of course it could be something else entirely, but this problem does
> seem to be common with Dell 2950, and this did fix the
We're using PAE Kernel.
On Thu, Apr 10, 2008 at 4:30 PM, Michael L. Young <[EMAIL PROTECTED]> wrote:
> > BUG: soft lockup detected on CPU#1!
> > [] softlockup_tick+0x96/0xa4
> > [] update_process_times+0x39/0x5c
> > [] smp_apic_timer_interrupt+0x5b/0x6c
> > [] apic_timer_interrupt+0x1f/0x24
> > .
I installed the Digium T1 card on Dell Poweredge 2950 and the system crashed
several times, we got a Kernel Panic and first though it was the OS so I
switched from Fedora 7 to Centos 5.1.
Our server was alarming in our monitoring system, when our Infrastructure
department investigated the issue the
rd 'signalling'
line 247: Unknown keyword 'group'
line 248: Syntax error. Start of range '> 1-23' should be a number from 1
to 1023
line 250: Unknown keyword 'relaxdtmf'
line 251: Unknown keyword 'dtmfmode'
14 error(s) detected
On 3/16
Cavalier did not leave me any paperwork to sign. I check with the colo as
well and they did not receive anything. I have a feeling I have not defined
my trunk groups well in zaptel.conf.
On 3/15/08, Ron Joffe <[EMAIL PROTECTED]> wrote:
>
> On Saturday 15 March 2008 21:22, Darren Wright wrote:
> >
)
Channel 21: Clear channel (Default) (Slaves: 21)
Channel 22: Clear channel (Default) (Slaves: 22)
Channel 23: Clear channel (Default) (Slaves: 23)
Channel 24: D-channel (Default) (Slaves: 24)
24 channels to configure.
On 3/15/08, broadband Voice <[EMAIL PROTECTED]> wrote:
>
> Can you s
27;ve setup quite a few Cavtel PRI's with
> *.the paperwork they asked you to setup?
>
> Typically, they only send 4 digits.
>
> Do you have the questionnare they asked you to fill out?
>
> dwright at d2 - tech dot com.
>
> ________
>
>
Thanks. I am in Philly. I may have to configure the extensions.conf well to
pass the incoming channels.
On 3/14/08, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
> On Fri, Mar 14, 2008 at 8:13 PM, broadband Voice
> <[EMAIL PROTECTED]> wrote:
> > I had Cavalier turn up a T
I had Cavalier turn up a T1 PRI. How can I put in the DIDs to direct to
Asterisk. Here is a log
Zaptel Tool (C)2002 Linux Support Services, Inc.
⤠T2XXP (PCI) Card 0 Span 1
ââ[3;10Hâterfaces â
â[3;37Hâ â â
Which Area codes are you looking for? Is this a multiple Exchange. I use
Link2voip but they don't scale well unless you're running a hobby. The best
solution is to get your PRI T1 from a telco.
On 2/23/08, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote:
>
> I posted the same question on asterisk-biz ma
problems with
> them. USLEC is another option.
>
> I have direct contact with Cavtel agents.
>
> Feel free to ping me off list to discuss.
>
> -Darren
> D2 Technology, INC.
>
> dwright at d2 - tech dot com
>
>
>
>
> ____
>
Steve,
Can you recommend a T1 provider for me? I tried Cavalier but have no
response and the other provider I am waiting for quotes is Broadviewnet.
Thanks.
On 1/17/08, broadband Voice <[EMAIL PROTECTED]> wrote:
>
> Steve,
>
> That is very helpful, How much are we talking abo
I have been running FC5 for 3 years with no issues, I also started using for
FC7 the last 2 months. I have not had any issues.
On 2/3/08, Goke Aruna <[EMAIL PROTECTED]> wrote:
>
> Tzafrir Cohen wrote:
> > On Sat, Feb 02, 2008 at 05:13:39PM -0600, [EMAIL PROTECTED] wrote:
> >> Ubuntu server for me
eally isn't needed.
>
> I've run the digium cards on all manner of Dell hardware (from
> old-school desktops all the way to the high end servers) and have
> never had issues.
>
>
>
> On 1/31/08, broadband Voice <[EMAIL PROTECTED]> wrote:
> > Digium has a c
Digium has a compatibility list of servers, however, it has not been updated
since 2006. One of the servers on the list has since been taken out of
production by Dell. Here are the remaining servers on the list: HP Proliant
DL360IBM x206IBM x346
Does anyone has a most recent list and I will be ad
, 5x10 NBD Onsite U3OS
[960-8162][960-8192][970-4070][984-1399][984-1417] 29*Installation
Services*:
No Installation Assessment NOINSTL [900-9997] 32
Print
On 1/29/08, Massimo Nuvoli <[EMAIL PROTECTED]> wrote:
>
> broadband Voice ha scritto:
> > Does anyone hav
Abdul,
Can you explain your request more in details? From your mobile phone or
landline into your Asterisk is a a different network and will have to pay
origination to into it. For the toll free the receiver pays for it. So if
you set up the toll free that means you pay for the call. Its never fre
Does anyone have any compatibilty issues with Dell *PowerEdgeTM 2950 III
2-Socket, Quad-Core 2U*? I plan on using this with the Digium T1 cards.
Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing lis
How did this workout? I am considering getting the Dell PowerEdge 2950.
On 11/5/07, Steven <[EMAIL PROTECTED]> wrote:
>
> 2950s work fine.
>
> I have had the parity error for over a year with no noticable problems.
> It is working fine.
>
> I did have to make some IRQ changes to clean up the sys
Steve,
That is very helpful, How much are we talking about in terms of the loop and
minute charges. If you want it offline I can send you a private my with my
phone number.
On 1/17/08, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
>
>
> On Jan 17, 2008 5:23 AM, broadband Voice
Can anyone share their experience with me? I am looking for a provider that
delivers Dialtone over T1 to terminate to my asterisk box and also provide
DIDs. Does the DIDs come with the T1 services or those are purchased/charged
seperately. Any help greatly appreciated. My target markets are Philade
Sent it off the list. Looks like it could be a problem on my side, it has
eth1 instead of eth0. but will take the response offline.
On 1/5/08, Terry Wilson <[EMAIL PROTECTED]> wrote:
>
> > > When I run the utility register of asking me for the license
> > number. I
> > > get the following.
> > >
>
Thanks Kevin for the prompt response.
On 1/5/08, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
>
> broadband Voice wrote:
>
> > When I run the utility register of asking me for the license number. I
> > get the following.
> >
> >
> > [EMAIL PROTECTED] mo
I have downloaded and unpacked tarballs
codec_g729a_v33_nocona.tar.gz.1
/pub/register/x86-64/register
/pub/register/x86-64/asthostid
When I run the utility register of asking me for the license number. I get
the following.
[EMAIL PROTECTED] modules]# /root/register
Digium Product Registration
What you need is SIP Trunking from a CLEC to your Asterisk. I know Level
provides this service but they deal only large volumes(3,000,000 minutes per
month). You can check with them.
On 1/2/08, Senad Jordanovic <[EMAIL PROTECTED]> wrote:
>
> Dovid B wrote:
> > Senad,
> > You can get unlimited as i
n Fri, 28 Dec 2007, Steve Totaro wrote:
>
> > Gordon Henderson wrote:
> >> On Fri, 28 Dec 2007, Steve Totaro wrote:
> >>
> >>
> >>> broadband Voice wrote:
> >>>
> >>>> On 12/27/07, *broadband Voice* <[EMAIL PROTECTED]
>
I figured it out from asteriskguru.org. If you are using kernel 2.6 enter
the following command '#make linux26', before doing '#make install'.
and also do ./configure. Hope it will help someone else.
On 12/29/07, broadband Voice <[EMAIL PROTECTED]> wrote:
>
> I su
Additional information
Linux 2.6.23.8-34.fc7 #1 SMP Thu Nov 22 20:39:56 EST 2007 x86_64 x86_64
x86_64 GNU/Linux is check the version of the kernel I am using if this helps
in anyway.
On 12/29/07, broadband Voice <[EMAIL PROTECTED]> wrote:
>
> I successfully obtained the Aster
I successfully obtained the Asterisk code and extracted them into /usr/src.
When I make and install asterisk, zaptel, libpri etc. Are they supposed to
move automatically into their respective directories?
I cannot find:
/etc/asterisk/
/usr/lib/asterisk/modules/
/var/lib/asterisk
Do I have t
frir Cohen <[EMAIL PROTECTED]> wrote:
>
> On Fri, Dec 28, 2007 at 07:56:39PM -0500, broadband Voice wrote:
> > I figured it out. The ftp site was not named well and corrected. The
> other
> > problem I have it after the extraction and make; it was suppose to go
> under
>
I figured it out. The ftp site was not named well and corrected. The other
problem I have it after the extraction and make; it was suppose to go under
/etc but that did not happen. I am trying to figure out why.
On 12/28/07, broadband Voice <[EMAIL PROTECTED]> wrote:
>
> I successfull
On 12/27/07, broadband Voice <[EMAIL PROTECTED]> wrote:
>
> I am using Asterisk and A2billing Calling Card Platform and after the 6th
> call the quality starts to degrade. The way it set up is the user calls into
> the system then dial out so I have 12 channels being used up bu
I am using Asterisk and A2billing Calling Card Platform and after the 6th
call the quality starts to degrade. The way it set up is the user calls into
the system then dial out so I have 12 channels being used up but 6 active
calls. Here are my specs Asterisk SVN-branch-1.4-r79142 on a i686 running
I use Gafachi.com and have good quality with no minimum requirements. Try
them at www.gafachi.com
On 12/16/07, Benjamin Jacob <[EMAIL PROTECTED]> wrote:
>
> Hello ppl,
>
> Am looking at some PSTN termination providers in US. If this question
> has been repeated, please point me to the correct link
Take a look at agent.conf and queue.conf.
On 12/12/07, satish patel <[EMAIL PROTECTED]> wrote:
>
> Dear all
>
> I need call center setup on asterisk so i need do
> doucment and book .is it available on net
>
>
>
>
> PGP Signature--
>
> Satish Patel
> mobile:- +91-98188
Does anyone have information on how to provide your own DIDs? I am currently
using Link2voip and even though they do a good job, its costing me a fortune
and wanted to be able to provide my own DIDs in the US. Any help will be
greatly appreciated. Thanks.
___
when I do a show G729 i get a 0/0 even though I believe it is working for a
carrier that accepts only g729. My feeling is becuase it is installed on 32
bus instead 64 bus thats why it is showing the wrong status.
On 12/6/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
> OkWhat is the issue?
I am thinking of putting a GSM and CDMA gateway in a country for termination
to the US. Has anyone experiance with this and any recommended vendors that
work well with Asterisk.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
aste
: yes
flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov
pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe constant_tsc pni
monitor ds_cpl cid xtpr
bogomips: 5985.45
On 11/28/07, broadband Voice <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> Ca
Hi,
Can anyone assist me in resolving this problem? I installed the G729 on a 32
and just found out that the server is 64. Thanks.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or upda
Is Asterisk capable of sending text messages to a cell phone or is there an
application for that?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.
Just a follow up, I have my server with Cari.net in San Diego. How do you go
about getting a block of DIDs and performing my own origination? Anyone has
any experience in this field? Thanks.
On 11/19/07, Eric Chamberlain <[EMAIL PROTECTED]> wrote:
>
> We use VoicePulse Connect. They now have a P
I looked through /etc/asterisk and could not find the folder sampl.call.
On 11/18/07, Tilghman Lesher <[EMAIL PROTECTED]> wrote:
>
> On Sunday 18 November 2007 10:20:18 broadband Voice wrote:
> > I have created a conference call solution for a client and works fine.
> The
>
I have created a conference call solution for a client and works fine. The
next challenge is to let the conference dial out the participant instead.
Has anyone done this before or know the function to achieve this? Thanks.
___
--Bandwidth and Colocation P
81 matches
Mail list logo
