On 9/24/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> My Asterisk book is on its way, so please bear with me.
> My experience so far is limited to sip.conf and extensions.conf, as I
> don't have a hardware board yet.
>
> First: It seems like an extension can be part of more than one context?
On Sun, 2005-09-25 at 08:58 +0300, Tzafrir Cohen wrote:
> On Sat, Sep 24, 2005 at 10:08:19PM -0700, trixter http://www.0xdecafbad.com
> wrote:
> > Has anyone built a game with the dialplan? I would think this would
> > most easily be managed by an AGI, but its possible with realtime
> > extension
On Sat, Sep 24, 2005 at 10:08:19PM -0700, trixter http://www.0xdecafbad.com
wrote:
> Has anyone built a game with the dialplan? I would think this would
> most easily be managed by an AGI, but its possible with realtime
> extensions.
>
> The game would be like 'adventure' that I first played o
Hi,
I was wondering if it is possible to setup with Asterisk a
Cisco 7960 to use extension mobility / roaming.
Meaning that a user logs into a phone and his profile moves
with him / her.
I have a network of ~75 Cisco 7960 phones, running SIP 7.5 distributed
across 2 asterisk servers i
thanks for the reply
Is Digium card compatible with EPABX standards
available in india , further how much does a card with
three FXS and one FXO interface cost,
Do u have any experience of implenting the same ,
I am in army what we lookin at is voice gateway to
interface our PBX with the data net
> > Can I ask how you are providing calls to us domestic numbers for free?
> >
>
> goiax.com is backed by TxLink [www.txlink.net]. We terminate a lot of
> minutes.
Matt:
That first logo ( companylogo / www.webaddresshere.com ) on the
website could use some work :) but the service works great!! T
Such hardware I believe incurs a stock standard duty of 35% plus some
other charges. All up, AFAIK it will cost you $2300USD to import the card
(based on the $1495 price for a 4 E1 card).
You can try guys like Drishti in Delhi, they can help out.
Regards,
Sahil Gupta
VoiceValley
On Sat, 24
Has anyone built a game with the dialplan? I would think this would
most easily be managed by an AGI, but its possible with realtime
extensions.
The game would be like 'adventure' that I first played on a prime in
1979. Or any of the infocom games (ie zork). Infact since the infocom
spec is k
where can i buy the digium or any other card to work
with asterisk in india and what is the cost like
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
_
Thanks for this. Interface works as it should now.
-Scott
- Original Message -
From: "Darren Wiebe" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, September 24, 2005 5:07 PM
Subject: Re: [Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.
IPSpeedDial creates speed
dial numbers for Asterisk.
Download from: http://ipsoftware.thorben.dk
Use this to create speed
dial numbers that can be used by all extensions on your Asterisk server. This
program will create entries in the asterisk database which you then can lookup
in y
On Sat, Sep 24, 2005 at 10:23:42PM -0400, Steve Gladden wrote:
> On the same P2 450Mhz box.
>
> I have tried both UHCI usb on a 2.4 kernel
> and enhanced RTC on a 2.6 kernel.
> Have not tried UHCI USB on a 2.6 kernel as of yet.
>
> Both seem to work GREAT.
>
> I have read in many places to b
On Sun, Sep 25, 2005 at 03:04:31AM +0200, Stefan de Konink wrote:
> On Sun, 25 Sep 2005, Marco Supino wrote:
>
> > I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my
> > problem is that the BIOS assigns the same IRQ to the SCSI controller,
> > and the TDM400P, i have tried severa
So the IP 601 is the 600 with a few extras? Looks like Polycom dropped the
ball again -- yet another pretty phone with NO BACK LIGHT. Does the design
team at Polycom have their brains unscrewed?
I've been playing with some Aastra phones lately, with limited success on
working properly. The
Could you post an example of you cdr output. The ASTPP question would
be better put on astpp-users. Visit
http://aleph.aleph-com.net/mailman/listinfo/astpp-users to subscribe.
Darren Wiebe
[EMAIL PROTECTED]
FaberK wrote:
Hi to All,
I've an Asterisk CVS Head working with Mysql.
My problem i
Hi to All,
I've an Asterisk CVS Head working with Mysql.
My problem is that instead of ANSWERED or something like, into the CDR
database records, I find only numbers.
This is also a problem to let ASTPP works, infact I receive an error:
ERROR - ERROR - ERROR - ERROR - ERROR
DISPOSITION NOT MATCHED
On the same P2 450Mhz box.
I have tried both UHCI usb on a 2.4 kernel
and enhanced RTC on a 2.6 kernel.
Have not tried UHCI USB on a 2.6 kernel as of yet.
Both seem to work GREAT.
I have read in many places to be sure to use a digium card as a time source
and not to reply on the cheap soluti
That was pretty stupid of me... kept thinking I'd screwed something
up on the pickup portion, when the obvious issue was screaming in my
face. :( Works fine with your correction!
Thanks
> You have to tell it the extension you want to pick up, it's not psychic.
> Doing what you're doing now would
I have been looking all over for software to generate the keys needed
to have secure calls with my Sipura. The only one that I have found is
on-line and thus not so secure:
http://voxilla.com/certrequest.php
Any pointers?
Thx,
-RFH
___
--B
Hello:
My Asterisk book is on its way, so please bear with me.
Based on what I have read and my actual Asterisk experiences, I am not
too clear on the context-extension relationship. I am not sure if some
of the error messages (Not Found) are a result of a bug or a feature.
My experience so fa
Hi,
I tried setpci INTERRUPT_LEVEL (or something similar, cant remmeber
now), and also setpci seems like it changed the IRQ, lspci -v still
shows the old IRQ
Marco.
Stefan de Konink wrote:
On Sun, 25 Sep 2005, Marco Supino wrote:
I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI
Only one PCI slot can hold the full size card like the TDM400P , the
other slot has a smaller opening on the case.
Marco.
Alexander Lopez wrote:
Can you try a different slot on the PCI bus??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Marc
On Sun, 25 Sep 2005, Marco Supino wrote:
> I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my
> problem is that the BIOS assigns the same IRQ to the SCSI controller,
> and the TDM400P, i have tried several options of making the bios change
> the IRQ, but it will always move them
Can you try a different slot on the PCI bus??
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Marco Supino
> Sent: Saturday, September 24, 2005 8:51 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] IBM x306
>
> Hi,
>
> Thi
Hi,
This is a little off-topic,but if someone has any info, it could help me
a LOT!,
I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my
problem is that the BIOS assigns the same IRQ to the SCSI controller,
and the TDM400P, i have tried several options of making the bios cha
Anybody have these working with Asterisk?
I have an AT-320.
Brett
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSC
You have to tell it the extension you want to pick up, it's not psychic.
Doing what you're doing now would give the application no extension.
Exten => _*99.,1,Pickup(${EXTEN:3}) should work, with usage being
*99
Joshua Colp
On 9/24/05 7:28 PM, "Rich Adamson" <[EMAIL PROTECTED]> wrote:
>
> Wha
Okay, after spending 12 hours on it I checked the thing that has bit me
before. Turn SElinux off.
OUCH!! :-)
Darren Wiebe
[EMAIL PROTECTED]
Darren Wiebe wrote:
I fought with this one for hours last night. I have to get it yet but
I'm not sure what the problem is. The permissions are all f
Enjoy!
http://www.asterisk.org/vonfall2005
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
Would using an IVR that ends with "connecting .." do it - or do you have to
have the call answered by someone who will wait until the recording plays in
every call ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mir
Sent: Sunday, September 25, 2005 00:08
Try:
exten => *99,1,Pickup(${EXTEN:[EMAIL PROTECTED])
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Rich Adamson
> Sent: Saturday, September 24, 2005 6:29 PM
> To: Asterisk-users-list
> Subject: [Asterisk-Users] Directed pickup syntax?
>
>
Oops, I didn't cc the list. Julian suggested I should try the older
version of app_chanisavail.c and that worked out well. I can now use
the g(#) switch and that works very well.
On 9/24/05, Brian McEntire <[EMAIL PROTECTED]> wrote:
That fixes it! Thanks.
So I can run CVS HEAD but I need to chec
I have a clean install of rapid 1.1 installed. I have installed the Flash Operator Panel from the Install Other Software menu. I am able to log into the panel from another computer on my network but all I see is the Conference Room 300. There are no extensions or any other options on the panel.
What's the proper syntax for implementing directed call pickup?
Running cvs-head from today (9/24/05 including Mark's fixes), and
tried:
exten => *99,1,Pickup(${EXTEN:3})
but that does not seem to work, and there isn't an example in the
configs directory. 'show application pickup' suggests the
On Saturday 24 September 2005 21:21, Mohammed Salim wrote:
> Hello everyone.
>
> Let me first begin by explaining what I'm trying to do...
>
> I have a calling card that has an access number and requires a PIN to be
> entered and then the number you want to dial, like normal calling cards. So
> w
Thanks for your answer.
This is not what the customer wants, they answer +500 calls a day, and
dont want to say "Welcome to BigCorp" every time.
They want a personal welcome file to be played to the caller every
time they pick up the ringing phone.
Michael
2005/9/24, Mathew McKernan <[EMAIL PROTE
hi:
no , i dont think it an authentication problem
,because once i have experienced this problem with
another voip provider and when i told him the problem
he fix the problem at his side ,so i think it an
invitation problem at his side.
--- Rich Adamson <[EMAIL PROTECTED]> wrote:
> > i have an as
> i have an asterisk box (195.112.214.99) with this
> configuration:
>
> sip.conf
> [callshop]
> type=peer
> host=sip.callshopcompany.com
> username=XXX
> secret=XX
> allow=all
>
> extensions.conf
>
> [call]
> exten => _00.,1,Dial,SIP/callshop/$
Hmm. Thanks for the heads up, but I'm not sure that's it.
It's jumping to 208 rather than 209, so it looks more like an off-by-one error.
I tried changing to priorityjumping=yes in
/etc/asterisk/extensions.conf and reinstalled the CVS-HEAD version, but
it still jumps to 208 whereas it used to jum
Hello everyone.
Let me first begin by explaining what I'm trying to do...
I have a calling card that has an access number and requires a PIN to be
entered and then the number you want to dial, like normal calling cards. So
what I have done is assign a local DID which when called, initiates a Dial
Adam Moffett wrote:
Hello.
I'm having trouble with callerid on outgoing calls. The recipient of
the call only sees "unknown" rather than the number I'm specifying.
If I set callerid info when calling an internal extension then I see
the callerid name and number when I call that extension.
I fought with this one for hours last night. I have to get it yet but
I'm not sure what the problem is. The permissions are all fine.
Any comments anyone?
Darren Wiebe
[EMAIL PROTECTED]
Scott Wolfe wrote:
I just installed the CVS 9-22 and am trying to get ASTCC up and
running. I was able t
On Fri, 23 Sep 2005 [EMAIL PROTECTED] wrote:
> Hello Armin,
> I tried your new version of chan capi and it works well.
>
> I did have one question about capi.conf. I have a bri with 2 spids, but
> I want to have the second go to a zap fax channel.
>
> Right now I can direct it, but the echo canc
I just installed the CVS 9-22 and am trying to get
ASTCC up and running. I was able to get the web interface config running and it
made the database but when I go to the brands page it says there is a problem
with the table. Also when I save the config file through the intraface it wont
save
On Sat, 2005-09-24 at 09:10 -0400, Alex Vishnev wrote:
> I briefly looked thru the code and I don't believe there is a way to
> separate the context or really make them independent. I know exactly what
> you want to accomplish. I think it could be done with a little trick. For
> example, every cust
Can someone share its working files experience on areskicc with me.
I got it installed but my sip user and iax could not get registered talkless
of making call and all the include directives instructed in the idiot guide
were followed.
Can someone share its experience with me on this?
Aruna
---
I have set up a * box with two hfc ISDN pci cards
using mISDN both in TE mode with PmP mode.
(using $MODPROBE hfcpci protocol=0x2,0x2
layermask=0xf,0xf)
I have no problem dialing out by explicitly naming the
mISDN port, ex: Dial(mISND/1/${EXTEN},60)
or Dial(mISDN/2/${EXTEN},60)
But it does NOT wo
Under 1.2 the +101 jumping is not enabled by default. There is a
variable returned showing the status of the application. You need to add
a "j" flag or put priorityjumping=yes in extensions.conf
Julian.
Brian McEntire wrote:
Hmmm...
I checked out CVS-HEAD, built and installed it this morning
Hmmm...
I checked out CVS-HEAD, built and installed it this morning. Most
testing was going well, but then I found out the behavior of
ChanIsAvail has changed (is broken?)
In my Dial Plan, if a call comes in on the PSTN line, and is not
answered by the extension (or if the extension is busy), Ch
> exten => _06.,1,Dial(IAX2/X/${EXTEN},30,r&SIP/[EMAIL PROTECTED])
Even that's incorrect. It should be:
exten => _06.,1,Dial(IAX2/X/${EXTEN}&SIP/[EMAIL PROTECTED],30,r)
See:
[Description]
Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][|URL]):
Also, see the Wiki or thi
I'm new to asterisk and need some help with getting a SIP connection
working.
I am trying to establish a termination point/DID number in another
country. I am currently running Asterisk CVS-HEAD. My foreign provider
uses SIP and authenticates via IP address. I am not required to
register my S
Hi gang,
I've been trying to use asterisk with an MTA device can any one offer some help as to how asterisk can work with the thing.
thanks a mil
Calvin
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asteris
My BT100 won't register with my Asterisk server, it always comes
back with a 403.
I've included my sip_additional (only one to to have the username 2201)
and a portion of the sniffer trace (packets 27 & 28). This has me puzzled
as I have my SPA-3K working (incoming and outgoing). On my BT100 I ge
Hi,
You stated that Digium is discontinuing the Wildcard series - that would
be there whole product line! In particular I am looking at the Wildcard
TDM 400P series of cards..
Thanks
Matt Roth wrote:
Don't bank on it. We were going to use a Wildcard as a timing source
on our Dell PowerEdg
I briefly looked thru the code and I don't believe there is a way to
separate the context or really make them independent. I know exactly what
you want to accomplish. I think it could be done with a little trick. For
example, every customer on hosted pbx would be given some kind of unique
identifie
Hi all,
Just a couple of quick questions. I have a HP DL360 G4 (dual 3.0Ghz
processors). The processors are EM64T. I am using a TE411P in the
system.
1. Should I run the a x86_64 Linux (CentOS) or just go with the plain old 32 bit version?
2. This being a dual processor system, should I turn on
i have an asterisk box (195.112.214.99) with this
configuration:
sip.conf
[callshop]
type=peer
host=sip.callshopcompany.com
username=XXX
secret=XX
allow=all
extensions.conf
[call]
exten => _00.,1,Dial,SIP/callshop/${EXTEN}
and when i try to sen
Hi all. Is it possible to get * to send calls to
different sip trunks depending on what codec the incoming call use? This to
avoid transcoding
Anders
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users ma
On Sat, 24 Sep 2005, Usman wrote:
> Is there any digium card that support E1 with SS7 and does Asterisk
> support SS7 ???
>
> any 1 who has done this ?
Maybe google has?
http://www.google.nl/search?q=Asterisk+SS7&start=0&start=0&ie=utf-8&oe=utf-8&client=firefox&rls=org.mozilla:en-US:unoffic
Is there any digium card that support E1 with SS7 and does Asterisk
support SS7 ???
any 1 who has done this ?
Usman
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://l
> i tried to send calls through an asterisk box to a
> voip provider the calls failed and here what i got :
>
> *CLI> Sep 24 11:09:19 WARNING[23356]: chan_sip.c:6890
> handle_response: Forbidden - wrong password on
> authentication for INVITE to '"asterisk"
> ;tag=as667cb0ae'
> -- SIP/call-3f
Hi list:
i tried to send calls through an asterisk box to a
voip provider the calls failed and here what i got :
*CLI> Sep 24 11:09:19 WARNING[23356]: chan_sip.c:6890
handle_response: Forbidden - wrong password on
authentication for INVITE to '"asterisk"
;tag=as667cb0ae'
-- SIP/call-3f73 is c
The patch is in cvs-head, which has been very stable for me. :)
> Hi Richard,
> I am experiencing the same problem. I'd like to test your patch. Thing, is,
> I don't know which
CVS it's in :)
>
> ... I checked out 1.2-beta on Tuesday (9/21) and compiled it. When I t
You should either use Agents (standard or callback) or disable voicemail
on the second server, with a straight dial instead of the dial+voicemail
macro you'll likely be using.
bye
l.
In data Fri, 23 Sep 2005 17:15:38 +0200, <[EMAIL PROTECTED]> ha scritto:
I all.
I have configured a pair
Hi,
I am looking for what sifira use to provide its services like
Callrecorder, Family voicemail..etc
Does it uses Asterisk, if yes, for what specific services?
Thanks
Gurminder
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asteris
65 matches
Mail list logo
