Again, this information is on
VOIPSpeak.net
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ramSent: Friday, November 25, 2005 11:40 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] NewBie to Ast Server,help need for the
configurat
Hi guys,
I'm trying to forward a call from one * server to another using SIP.
Everything works when I use fromuser in the sip entry of the *
forwarding the call. The problem is that when the receiving * sends
the call to the UA, it puts the caller to be the value of fromuser
instead of the
All the instructions you need are in the handbook and
the articles at VOIPSpeak.net.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ramSent: Friday, November 25, 2005 11:39 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] NewBie
where can i find that
ram
On 11/26/05, Zafer Khodr <[EMAIL PROTECTED]> wrote:
I would recommend looking at some Virtaul machine ware software and installing it that way.
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of ramSent: Saturday, 26 November 2005 5:46 PM
To: Ast
Hi
yes i have downloaded now compiled
its went to successfully
i would like to suggest people
what are need to done later
to achice lan extentions
and out dialing using IAX provider
ram
On 11/26/05, Kerry Garrison <[EMAIL PROTECTED]> wrote:
If you had read the Wiki you would have seen the
I would recommend looking at some Virtaul
machine ware software and installing it that way.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ram
Sent: Saturday, 26 November 2005
5:46 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Aster
how much traffic do you have to philippines? please contact me offlist
On 11/19/05, Joseph <[EMAIL PROTECTED]> wrote:
> The problem are almost solved.
> With regards to Teialx it appears that I was registered to their server
> "co1" that worked in the past (but not now) as I'm suppose to be on
> "
If you had read the Wiki you would have seen there is a tar
version you can install onto an existing operating system.You should join the
forums there since this is not an [EMAIL PROTECTED] support list.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ramSent: Frida
Hi all
iam downloading the [EMAIL PROTECTED]
as recomended in this group
but when iam going through the docs
its need some dedicated server to install with fresh
but as of now i dont have any dedicated server allocated
until unless iam successfull of demo to my boss
so is there any way i ca
Phil.
You won't have much luck achieving this. The closest you will be able to
get is to set the intel voice modem up as an FXO port. Was this what you
meant?
If so, take a look on the voip-info.org site as there is a page there
that will point you in the right direction.
http://www.voip-info.or
On Thursday 24 November 2005 00:01, Rainer Maier wrote:
> Hi Matt,
>
> I did not move the whole asterisk directory I just put a link to it. (ln -s
> /usr/src/asterisk-1.2.0 /usr/src/asterisk)
> Then I tried to compile but the error stayed.
> I also tried with MySQL 4.1.15 and had the same error.
>
can somebody help or point in the right direction.
where i might be able to find info re: setting up intel voice modems as
fxs ports..
thanks in advance.
Phil
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing lis
if you have msn, add me in your list.. maybe i can help in your newbie
question ehehe
On 11/26/05, John Fraser <[EMAIL PROTECTED]> wrote:
>
>
> Im one! John, also a new user.
>
> cheers
>
> John
>
> [EMAIL PROTECTED]
>
> On Fri, 25 Nov 2005 18:54:02 +0100, Michael Kenjie Nukui wrote
>
> > Im one!
Stig Even Larsen wrote:
I'm having problems connecting my Sangoma cards to our PRI (E1)
interface. It seems that the card get connected (green led), but
Asterisk reports:
Status: Provisioned, Down, Active.
When I install another Sangoma card on the same system (pri_net), and
connect the two c
[EMAIL PROTECTED] is just
Asterisk with the AMP web GUI and a few other tools. Any size company that
Asterisk can handle would be indentical to what [EMAIL PROTECTED] can handle.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ramSent: Friday, November 25, 2005 9:0
Hi
iam downloading [EMAIL PROTECTED]
just want to know how many users it can handle
Lan Side
and how many IAX provider it support
i have made new plan.. planning to use 100users of my LAN 4 accounts to calling out side
may be in future 100+ also
is this a right solution
ram
On 11/
Hi
thanks, yes iam working on it
On 11/26/05, Michael West <[EMAIL PROTECTED]> wrote:
http://sourceforge.net/projects/asteriskathome/
From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of ramSent: Friday, November 25, 2005 11:33 PM
To: Asterisk Users Mailing List - Non-Commercial
http://asteriskathome.sourceforge.net
There are several setup guides at: http://voipspeak.net
The Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20at%20%20Home
The Handbook: http://www.voip-info.org/wiki/view/Asterisk%40home+Handbook+Wiki
Kerry GarrisonPublisher - GeekGaze
http://sourceforge.net/projects/asteriskathome/
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ramSent: Friday, November 25, 2005 11:33 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] NewBie to Ast Server,help need for the
configu
Hi
can some one point me where can i download this ??
ram
On 11/25/05, Juan Janczuk <[EMAIL PROTECTED]> wrote:
It seems like [EMAIL PROTECTED] could be your best solution.
It has a nice user interface (AMP, you can try to install it in your actual asterisk box),
that lets you do all you say.
Eric "ManxPower" Wieling wrote:
My point is that CALLERID(number) is ALWAYS the same as ${CALLERIDNUM}
so setting one to the other is pointless. It's like setting 2=2.
Same with the CallerIDName stuff.
___
Point taken.
Well, between our posts,
On Nov 25, 2005, at 7:00 PM, Manny A. Wise wrote:
-Original Message-
From: Tom Rymes [mailto:[EMAIL PROTECTED]
[snip]
On Nov 23, 2005, at 11:04 AM, Manny A. Wise wrote:
[snip]
Well, as the user stated on the original message, the asterisk
server is behind a NAT and the client is also
Adam Rybak wrote:
Hello,
i have succesfully ipgraded my system to asterisk 1.2 with OOH323C channel
driver, today i got hangup of my asterisk after this messages:
Nov 25 21:03:22 WARNING[24395] channel.c: Avoided initial deadlock for
'0x8198118', 10 retries!
This issue is currently being
Hi, everyone,
I have met a very strange problem. We use a Asterisk PBX connected with two rollover PSTN phone line provided by Verizon by Digium TDM cards. The incoming calls are always OK. But when I make outgoing calls, sometimes it works, sometimes it just get a busy tone, doesn't work at all.
> > I eventually switched to using a Astra 480i CT desk phone with a couple
> > of corless handsets. It's been great.
>
> My first thought was to use a cordless phone and a Sipura ATA. But this
> is a 100,000 sqft warehouse with a freezer section in the middle so the range
> isn't quite there. Nex
Hi,
Recently it seems that if someone places a caller on hold the system
will crash later that evening.Here is what the debug gave me from
the output:
Nov 25 02:59:53 DEBUG[20526] chan_sip.c: Auto destroying call
'[EMAIL PROTECTED]'
Nov 25 03:00:00 VERBOSE[20526] logger.c: -- Remote UNIX c
GSM 900MHz / 1800MHz GSM Gateway
Including antenna and power supply.
Limited Stock, please email gsm AT cyber-telecom.com for more info
Or visit www.cyber-telecom.net to purchase right away.
Sam
___
--Bandwidth and Colocation sponsored by Easynews.co
Hi Bob -
> I am now running sip 1.6.2 with a 2.6.1 bootrom.
> After moving from a 1.5 I now only see 2 softkeys
> at the main window: New Call and Forward.
>
> How do I get a Park softkey?
Just so you know, the park softkey will only show up when a call is active, and
so far it will only show
UK, London Based DID £1 per month
All number begin with 0208 0xx
If you are interested please email [EMAIL PROTECTED]
SIP based and support standard ulaw or alaw.
Unlimited incoming minutes.
For multi channels please email for pricing.
I got quite a lot of those 0208 DID that I want to g
On Sat, 2005-11-26 at 01:39 +0100, Michiel van Baak wrote:
> On 16:25, Fri 25 Nov 05, trixter aka Bret McDanel wrote:
> > Due to a very poorly written webapp, it took about 3 days. I got it
> > from numberingplans.com and the way they get the database info is poorly
> > done. For each country the
I updated my Polycom 500 to bootrom 2.6.1 and sip 1.6.2 and things
_seem_ to be working. The biggest thing that is not working is any
change via the web browser is not saved. I have to make changes via the
sip.cfg, phone1.cfg, etc. type files. I can access the phone via the web
and click all th
Hello:
Looking for help ...
I need to setup an * box in order to swap my office from
an old pbx..., the only thing that I can't figure out on how
to do with * is to have something like this:
- User A and customer are in a bridged call
- User B needs go to that bridged call and participate
On 16:25, Fri 25 Nov 05, trixter aka Bret McDanel wrote:
> Due to a very poorly written webapp, it took about 3 days. I got it
> from numberingplans.com and the way they get the database info is poorly
> done. For each country there are 1-N pages, as the page number
> increases so does the time i
On Fri, 2005-11-25 at 16:11 -0800, snacktime wrote:
> Thanks Bret. How difficult was it to compile this list? I'm assuming
> it's a compilation of pubically available data? I'd be interested in
> how much work it would be to keep this up to date?
Due to a very poorly written webapp, it took abo
>
> I was wondering if you can use Asterisk from the command line to make it
> make an outgoing call and issue other commands whilst it's in the call?
>
> Sort of like when you use Minicom with a modem connected to a serial port
> and send it AT commands.
I would suggest call files or the manager
Thanks Bret. How difficult was it to compile this list? I'm assuming
it's a compilation of pubically available data? I'd be interested in
how much work it would be to keep this up to date?
Chris
___
--Bandwidth and Colocation sponsored by Easynews.com
Thank you Steve / Kevin.
I'll look for the jumper on the card when I go to our co-lo.
Andy
On 11/25/05, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
Andy Kuo wrote:> We have a Digium TE406P connected to 1 T1/PRI now.> Can I put in an E1 to one of the unused ports on the same card?
Yes, just cha
Andy Kuo wrote:
We have a Digium TE406P connected to 1 T1/PRI now.
Can I put in an E1 to one of the unused ports on the same card?
Yes, just change the selection jumpers on the card appropriately.
___
--Bandwidth and Colocation sponsored by Easynews.
I have compiled the last of the info that I had and I now have a 43MB
CSV file (3.5MB compressed) containing about 579,887 entries. The
information includes each country, whether its geographic (and where),
mobile, premium, etc. Even have short codes (like 911/999 for emergency
etc) for each coun
Hi.
I have a problem or require in my Asterisk, I need limit the out calls
from my outgoing context. I have configure a outgoing peer in
sip.conf [outgoing-xxx], but I need that in this peer out calls 4
only.
I configure my outgoing peer with registry parameters for out calls with
my SIP pr
Hi,
We have a Digium TE406P connected to 1 T1/PRI now.
Can I put in an E1 to one of the unused ports on the same card?
Thanks.
Andy
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.co
Yes, find the the mac address on the bottom of the phone and convert
the CAPs to lower case!!
It doesn't work otherwise...
Thanks for the help.
Hugh
On 11/25/05, Alvaro Parres <[EMAIL PROTECTED]> wrote:
> we have TFTP and also those files are created upgrade automatic.. And also
> we create man
For an example here is what I setup to call out, we have a job that runs on our mainframe, when the job completes it ftps flag1.txt to our asterisk server, the .bash program is run from the crontab at a certain time and notifiy staff if the job is not complete at that time. It will keep calling (u
I have a similar problem in Australia and I think it has to do with
chan_zap.c
Currently Digium are investigating it for me as it is in association
with one of their TDM400P cards.
Gonzalo Servat wrote:
Hi there.
I'm having a strange issue with the distinctive ring detection in
Asterisk (I
Hi there.
I'm having a strange issue with the distinctive ring detection in
Asterisk (I have a FXO card).
It certainly seems to be enabled as I can see the Asterisk console
spitting out the cadences (same cadence every time: 0,0,0) but the
problem is that it is not waiting 2 seconds after "Startin
Gosh, hadn't kept up on the list lately. Sorry to ask a solved question!
Moj
Mojo with Horan & Company, LLC wrote:
But more importantly, what would you do with it if you found it? Has
anybody made this softkey interface with Asterisk's parking functionality?
Anyway, is this what you want?
Serach in the list, about 1 o 2 weeks ago.. there is a guide for how to setup the key with asterisk
On 11/25/05, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote:
But more importantly, what would you do with it if you found it? Hasanybody made this softkey interface with Asterisk's parking
If this is the problem you're having, you would simply replace something
like:
exten => s,1,Dial...
with
exten => s,1,Wait(1)
exten => s,2,Dial...
Gary MacKay wrote:
How do I make it wait? For how long? I watched the logs but did not see
anything that related to this.
Check your logs, ma
But more importantly, what would you do with it if you found it? Has
anybody made this softkey interface with Asterisk's parking functionality?
Anyway, is this what you want? in ipmid.cfg,
HTH!
Mojo
Bob Knight wrote:
I am now running sip 1.6.2 with a 2.6.1 bootrom.
After moving from a 1.
> Well I disagree on the untraedit, since vi does a far better job. :)
you're right, if you know the secrets of this operating system ;-)
Regards and nice weekend to all out there
Guido Hecken
___
--Bandwidth and Colocation sponsored by Easynews.com --
fixed by adding a w at the end of the dialstring:
"w" would normally indicate a "wait", but putting it at the end of your
dialstring will cause asterisk to assume the number before is "complete".(sic!)
~markster
e.g. Dial(Zap/G4/${EXTEN}w)
regards
chris
On Fri, 25 Nov 2005 20:31:41 +0100
Chr
hi,
i tried several days without success. I have installed Debian sarge with
Kernel 2.6.14, misdn (installed from "install-misdn.tar.gz") and two
hfc cards.
HFC1 = NT-Mode
HFC2 = TE-Mode
I want to do, what everybody has done ;-) .
Internal:
ISDN <-> ISDN
SIP <-> SIP
ISDN <-> SIP
External
Hi Roger,
There are quite a number of areas that could stop this working on the
MD110 PBX side. I'll show you how mine are configured - you may like to
try some of my paratemeters. My MD110 is running BC9 software - should
consult your Ericsson manual to understand the roles of each of these
I am now running sip 1.6.2 with a 2.6.1 bootrom.
After moving from a 1.5 I now only see 2 softkeys at the main window:
New Call and Forward.
How do I get a Park softkey?
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
___
--Bandwidth a
Hello,
i have succesfully ipgraded my system to asterisk 1.2 with OOH323C channel
driver, today i got hangup of my asterisk after this messages:
Nov 25 21:03:22 WARNING[24395] channel.c: Avoided initial deadlock for
'0x8198118', 10 retries!
Nov 25 21:03:25 WARNING[24395] channel.c: Avoided ini
I'm having problems connecting my Sangoma cards to our PRI (E1)
interface. It seems that the card get connected (green led), but
Asterisk reports:
Status: Provisioned, Down, Active.
When I install another Sangoma card on the same system (pri_net), and
connect the two cards with a PRI cross-ove
We have done this before but we had to do go back to 1-2. This
should have not caused a problem but, we had a lot of call quality problems
when we had it set to 1-10500 and only had 7 ext on the server.
Just my two cents
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMA
we have TFTP and also those files are created upgrade automatic.. And
also we create manually the file for the new phones so they have
the minimal addres book of the company.
On 11/25/05, Watkins, Bradley <[EMAIL PROTECTED]> wrote:
Hrmmm... I'm not sure how much more help I can be on this exactly
Hi list:
what are the steps to do to asterisk to be ready fro
callback system?
__
Yahoo! Music Unlimited
Access over 1 million songs. Try it free.
http://music.yahoo.com/unlimited/
___
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Hi,
I'm currently facing some issues regarding echo between the asterisk
box and the 3750, here is my scenario:
TELCO --> Asterisk --> HiPath 3750
(E1) (TE210P)
|
SIP PHONES
When I dial from a SIP p
Im one! John, also a new user.
cheers
John
[EMAIL PROTECTED]
On Fri, 25 Nov 2005 18:54:02 +0100, Michael Kenjie Nukui wrote
> Im one! kenjie pre, i am just a new user of asterisk.
>
> regards,
>
> kenjie nukui
> [EMAIL PROTECTED]
>
> On 11/25/05, Angelito Manansala <[EMAIL PROTECTED] >
Ethernet address != IP address
Ethernet address = MAC address
Look on the bottom of your phone
On Nov 25, 2005, at 10:55 AM, hugolivude wrote:
I cannot seem to get the "Local Directory" feature to work. I've
consulted section 3.1.17 of the Administrator Guide. It says to put a
file -directo
Diego Andrés Asenjo González wrote:
Hi!
I'm registering an asterisk server in a Sysmaster with a SIP account.
The registration succeeds and I can establish a call that come from the
Sysmaster.
After around 80 seconds the Sysmaster sends a BYE SIP message and the
call hang up. This does not occ
On 11/25/05, harry gaillac <[EMAIL PROTECTED]> wrote:
> Try to post your problem to asterisk-dev
Hmm that seems to be your solution for just about everything doesn't
it Harry? :)
I think the problem is that asterisk-addons got built out of order or
didn't get rebuilt at all, but I can't remember
Hello everyone..
I'm trying to lock down my asterisk install as much as possible and I
keep reading about people saying 'you can narrow the range of ports in
rtp.con' (by default it's from 1 to 2 I think).
My question is this - how much can I narrow it down? Can I narrow it to
10 ports,
Hello Group!
While parsing my cdrs of the last week, i realized that approx. 1 in
100 _successful_ outbound zap-calls are recorded with a truncated
destination number in the verbose logs and in the cdrs.
Several digits are simply missing. eg 0049 is recorded as
0049. I have received cdrs
Well I disagree on the untraedit, since vi does a far better job. :)
On 11/25/05, Guido Hecken <[EMAIL PROTECTED]> wrote:
> also add winscp and ultraedit to your windows system, it works great.
> http://winscp.net/eng/index.php
> http://www.ultraedit.com/
>
> Regards
>
> Guido Hecken
>
> > > Witho
HI List,
You'll have to pardon the newbieness of this question, I was editing
the sip.conf file on my asterisk server yesterday, and now none of my
asterisk trunks will connect. From my knowledge sip.conf does not
effect registration, but there have been no other changes at all. Below
is my sip.co
Steve Davies wrote:
Hi,
This is probably just me mis-reading the documentation, but I have
been led to believe that the '.' in extensions.conf means zero or more
digits, such that
exten => _X.,1,NoOp()
Would trigger for either a single digit, or for a longer number (as
long as it starts with a
have you added allow=speex & allow = ilbc in the sip & iax conf files ?
On 11/25/05, Alejandro Vargas <[EMAIL PROTECTED]> wrote:
I'm testing [EMAIL PROTECTED] 2.0 beta 6.I'm checking de different codecs but with speex and ilbc I don'treceive any sound. I tested xtensofphone and iaxComm. With both I
On 08:26, Fri 25 Nov 05, Kerry Garrison wrote:
>
> Don't you actually want to do a move instead of a copy? During a copy
> Asterisk might actually pull a partial file but a move will not be detected
> until the file is 100% in place. Probably not a problem unless you were
> writing a very busy ca
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Truth springs from argument amongst friends.
[EMAIL PROTECTED] wrote:
>Hello friends,
> I have a strange problem. I am using asterisk 1.2 and asterisk addons 1.2. I
have three SIP phones and one H323 ph
Hrmmm... I'm not sure how much more help I can be on this exactly. For all
of my users, we use FTP and the files get created and updated automatically.
I should note that this is with all IP600s/601s but this should be the same
even for the 501s.
- Brad
-Original Message-
From: [EMAIL PR
I have installed bristuff 0.3.0 for Asterisk 1.2 with kernel 2.4, but when I doing : insmod qozap.o the shell show this messagge: qozap.o: qozap.o: unresolved symbol free_irq_Rsmp_f20dabd8 qozap.o: qozap.o: unresolved symbol pci_find_device_Rsmp_c584f4e3 qozap.o: qozap.o: unresolved symbol __requ
After playing around with the CALLERID(number) and
CALLERID(name) variables and things, I find that asterisk is sending
the "name" to my phone and the name is "unknown". I added a line
exten => _X.,Set(CALLERID(name)=${CALLERIDNUM}) and now it shows
the number. Is this the right way to do thi
Please move _all_ discussion regarding A2Billing (and the mechanics of
using any billing system unless the discussion is explicitly
Asterisk-related) to some other mailing list; this mailing list not a
support forum for add-on packages.
Thanks!
___
-
what hardware is involved? Are you using a hard or soft phone? if
hard, an ip phone or an ATA adapter? What country are you in?
Finally, please run
/usr/src/zaptel/zttest -v
and watch it for a while. If the average result is less than 99.98%,
consult http://www.voip-info.org/wiki/view/Ast
Im one! kenjie pre, i am just a new user of asterisk.
regards,
kenjie nukui
[EMAIL PROTECTED]On 11/25/05, Angelito Manansala <[EMAIL PROTECTED]
> wrote:--Best Regards,Angelito ManansalaMobile: +639175425807
DID: (+63) 44 7906770msn: [EMAIL PROTECTED]skype: bulcrack
Anyone has experiences with sending faxes using Asterisk and a TE405P
Digium card (or similar PRI) with a PRI connection?
Using HylaFax and a PRI card such as the Patton 2977, Eicon Diva sending
faxes works very well. There is a new project out there called
IAXModem (written by Lee Howar
On Fri, November 25, 2005 9:29, Kristof Hardy said:
> Francesco Peeters wrote:
>> I compiled 1.2 and bristuff 0.3.0 Pre1 yesterday late and that now seems
>> to work! * is up and running *with* 2nd card in NT mode...
>
> Nice to hear *1.2 and bristuff 0.3pre1 makes a difference..
>
>
Just switched
I tried changing the name to the MAC address format, but still no
luck. No contacts appear after re-boot and I still can't add them
manually either.
No particular reason for using TFTP over FTP. I'm a hack so I just
followed the instructions at:
http://www.voip-info.org/wiki/index.php?page=Poly
Asterisk 1.2
We tried today to send a number of sms messages at the same time.
the smsq application seems to send 7 messages at a time, and then stops.
If I send another sms message, then another 7 messages are sent.
Has anyone else seen this ?
Julian.
___
How do I make it wait? For how long? I watched the logs but did not see anything that related to this.
Check your logs, make sure you are waiting long enough before sending
the call to the polycom.
Uf asterisk sees the CID, it should send it and it should show up on the
polycom.
Greg
-
Don't you love clients that keep asking for features after an install?
I have a client that is asking about doing distinctive rings for external vs
internal calls. They are using Grandstream GXP-2000 phones which (although a
pain to configure) have 4 ring types. I am guessing that I would need to
The filename needs to be -directory.xml, not -directory.xml. Grab the MAC off the back of the phone.
This is the same as for the provisioning files if you are using your TFTP
server to do that. Also, is there a reason that you aren't using FTP? It's
much more robust, and does not require that y
Hi
edit your a2billing.conf and set debug level to 3, try again and send us your debug to see what happend.
rafael
On 11/25/05, Jose M. Ramirez <[EMAIL PROTECTED]> wrote:
Hi list, all. Please, I need help. Although
already I installed a2billing, simply I cannot initiate its execution.
I cannot seem to get the "Local Directory" feature to work. I've
consulted section 3.1.17 of the Administrator Guide. It says to put a
file -directory.xml (where is the
IP address of the phone) into the TFTP directory. Polycom provides a
template.
The IP address for one of my phones is 192.168
Hi,
This is probably just me mis-reading the documentation, but I have
been led to believe that the '.' in extensions.conf means zero or more
digits, such that
exten => _X.,1,NoOp()
Would trigger for either a single digit, or for a longer number (as
long as it starts with a digit)
In practice (
Hi list, all. Please, I need help. Although
already I installed a2billing, simply I cannot initiate its execution.
Only appears this:
-- Executing Answer("SIP/20-456d", "") in new
stack
-- Executing Wait("SIP/20-456d", "2") in new
stack
-- Executing DeadAGI("SIP/20-456d", "a2billing.ph
Do you have accountcodes in the database? If you do, you could use
astpp quite easily. We could cut out most of the functionality for
you. Right now I don't have a way to search by date but that would be
failry easy to add and I will be working on it soon anyways. Drop me a
line if you want
Don't you actually want to do a move instead of a copy? During a copy
Asterisk might actually pull a partial file but a move will not be detected
until the file is 100% in place. Probably not a problem unless you were
writing a very busy call center app.
Kerry Garrison
Publisher - GeekGazette.co
Hi!
I'm registering an asterisk server in a Sysmaster with a SIP account.
The registration succeeds and I can establish a call that come from the
Sysmaster.
After around 80 seconds the Sysmaster sends a BYE SIP message and the
call hang up. This does not occur to the hard/soft SIP phones register
Look under phone home:
http://mundy.org/blog/index.php?p=63
Hope this helps
Kib Eki wrote:
Hi,
we have a html based telephonelist on our intranet site.
Does there exist any solution to initiate a call from a link ?
We use Polycom SIP IP phones.
thanks and regards,
bk
___
ram wrote:
> Hi
>
> why my posting are not accepting in this list
Don't know.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineap
Obelix wrote:
> I am quite familiar with Asterisk AGI, but I am looking for forums or groups
> that discuss more techniques, like the Manager API etc.
>
> Asterisk Users only delves into Asterisk dial plans, configuration etc and
> Asterisk Dev deals with the main Asterisk itself.
>
> I am looki
Kib Eki wrote:
> Hi,
>
> we have a html based telephonelist on our intranet site.
> Does there exist any solution to initiate a call from a link ?
> We use Polycom SIP IP phones.
If you know how to code, have a look at the sample.call file in the
/usr/src/asterisk directory. This file can be fil
Nitin Joshi wrote:
> Hi All,
> I am using Asterisk 1.0.7 with an X101P analog card which is connected
to an
> Alcatel pbx. My problem is that when I place outbound calls on the zap
> channel, Asterisk returns a connect event as soon as the phone start
> ringing. This means that Asterisk is not bein
Hi everyone,
just a question: is there a way to remove this
message on the CLI ?
== Manager 'root' logged on from 127.0.0.1
== Manager 'root' logged off from 127.0.0.1
Thanks
Giordano
___
--Bandwidth and Colocation spons
I have the newest SpanDSP setup with asterisk 1.2.
Generally, 99% of received faxes are OK, but only about 20% of faxes
sent are delivered properly.
In zapata.conf I have set faxdetect=both, but it doesn't seem to disable
echo cancellation (I looked into asterisk logs and it says "Enabled ech
Erwan DESVERGNES wrote:
> Did someone use a Te411p with 4 T2 in France ? I’ve got some problem
Not exactly France, but I do have TE405P and TE110P running fine in
French Polynesia (should technically be the same network as France). The
only problem I had was the operator not configuring EuroISDN
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