Updated asterisk Receptionist.
Shouldnt have problems with IAX calls causing an
error.
http://www.easyhomenetworks.com/AstRec/
Kyle
On Wed, Jun 02, 2004 at 02:01:49PM +1000, Shaun Ewing spake thusly:
exten = xx,1,Dial(IAX2/[EMAIL PROTECTED]/phoneSIP/phone|60|r)
By that example, you can see that I am dialing IAX2/[EMAIL PROTECTED]/phone and
SIP/phone at the same time with ring back with a timeout of 60 seconds.
Note
Thanks, Andy.
I have thus tried to use the other H323 driver written by Michael, I have
used the newest PWLIB and OPENH323 libraries and newest OH323 driver. After
installing, I was able to get two way audio and all. I have tried this
driver before but at the time, there was a false answer
On Tue, Jun 01, 2004 at 11:34:28PM +0200, Gunnar Schaller wrote:
...cut
When I try to make 3 simultaneous connections from SIP to ISDN the
first and second one works, but on the third connection this happens:
-- Executing Dial(SIP/gunnar-26ea, CAPI/7501:7986:bBYEXTENSION) in
new stack
Dear Michael
I tried using the newest version of your H323 driver, but somehow it seems
that it is not hanging up the channels and for some reasons, it is NOT
writing my cdr to the mysql database, it was writing properly before. As you
can see , the call finished at 2:40:12 but refused to hang up
There's a new version out with some bugs fixed
major ones fixed: deadlock on call end, iax thread getting locked out,
few contact group list bugs, one on exit crash bug fixed
I'd highly recommend upgrading to it
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
-Adam
Adam Hart
On Wed, 2 Jun 2004, Adam Hart wrote:
I'd highly recommend upgrading to it
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
Can I recommend you label files with version numbering - this must be
about the third ? fourth ? firefly-thirdparty you've released.
true, it's internally versioned though - look at the build number. But
yes, I'll start suffixing a buildnumber on the files.
i'm hoping this will be the last release before the magic feature called
conferencing, unless this sip registration issue is firefly related
-Adam
gARetH baBB wrote:
On
When I try to make 3 simultaneous connections from SIP to ISDN the
first and second one works, but on the third connection this happens:
-- Executing Dial(SIP/gunnar-26ea, CAPI/7501:7986:bBYEXTENSION) in
new stack
chan_capi.c:1147 capi_request: didn't find capi device with outgoing
msn =
Adam
The link doesn't seems to work. Get back the following:
Parse error: parse error, unexpected T_STRING in
/usr/virtual/www.virbiage.com/www/firefly/download/firefly-thirdparty.exe on line 121
Reto
There's a new version out with some bugs fixed
major ones fixed: deadlock on call end,
fixed
Reto Stauss wrote:
Adam
The link doesn't seems to work. Get back the following:
Parse error: parse error, unexpected T_STRING in
/usr/virtual/www.virbiage.com/www/firefly/download/firefly-thirdparty.exe on line 121
Reto
There's a new version out with some bugs fixed
major ones fixed:
Thanks, is working now.
Reto
Zitat von Adam Hart [EMAIL PROTECTED]:
fixed
Reto Stauss wrote:
Adam
The link doesn't seems to work. Get back the following:
Parse error: parse error, unexpected T_STRING in
Well here is my example.
I have a client, who has lots of work associates who call in from all over
the world to conference calls. For these calls, many of them use cell
phones because of local telco issues. This company then pays the cell
bills for these call ins. The bills are
I think no.
Oliver Vermeulen wrote:
Hi ,
Do anybody know how you can run multi proccess of * on a server ?
Thanks,
O
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Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
On Wed, Jun 02, 2004 at 09:27:14AM +0200, Gunnar Schaller wrote:
...cut
chan_capi.c:1147 capi_request: didn't find capi device with outgoing
msn = 7502. you should check your config!
app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI'
== Everyone is busy at this time
Do
Hi Gunnar,
here is our capi.conf for two controllers on two different ISDN lines
;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=041120
incomingmsn=*
controller=1
softdtmf=0
context=default
devices=2
msn=041682
Wim,
If ya don't need callerid then add the patch at
http://www.nodomain.org/asterisk to zaptel and asterisk directories.
I did this for UK callerid and the phone now rings on the first ring of the
CO.
Bit of a bodge but it works.
Dave
-Original Message-
From: [EMAIL PROTECTED]
On Wed, Jun 02, 2004 at 08:14:44AM +0100, gARetH baBB wrote:
On Wed, 2 Jun 2004, Adam Hart wrote:
Can I recommend you label files with version numbering - this must be
about the third ? fourth ? firefly-thirdparty you've released.
.. but have firefly-thirdparty.exe be a symbolic link to
Hi,
I have two softphones connected to an Asterisk stable. I have two
extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be
completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on
extension 2000 will ring, but as soon as the call is picked up, extension
2000 will
In article [EMAIL PROTECTED],
Leo Ann Boon [EMAIL PROTECTED] wrote:
The new TDM400P with FXO doesn't take up any IRQ. I've 2 boards and both
are not using any IRQ.
Weird - does that mean they can't provide Zaptel timing like the X100P can?
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL
Resolved...
canreinvite=no
(i've put careinvite :-))
igor
On Tue, 2004-06-01 at 19:24, Igor Barsanti wrote:
I'v a sip client and a sip trunk to FWD:
[general]
port=5060
context=default
tos=reliability
disallow=all
allow=ulaw
careinvite=no
[freeworlddialup]
context=default
I was just wondering if someone has experiences to use Asterisk in an
existing Ericsson MD110 environment. Particulary I'd like to know if it is
possible to use the MD110's system phones directly connected to Asterisk.
I'm not very familiar with it but would it be possible to use ADSI with
these
If you are just doing VoIP (i.e. no FXO/FXS Cards involved) you should
be able to run up multiple virtual copies of Linux * in VMWare or
Virtual PC.
Though I guess you would need a pretty pokey machine
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
Has anyone managed to use a bluetooth headset or phone with their install
of Asterisk?
What I had in mind was either have a headset paired with the server and
use that to answer/make calls in some way, or forward the calls to my
mobile via bluetooth if that's possible.
--
-S
Hi.
Johnson-Perkins, Robert wrote:
If you are just doing VoIP (i.e. no FXO/FXS Cards involved) you should
be able to run up multiple virtual copies of Linux * in VMWare or
Virtual PC.
Though I guess you would need a pretty pokey machine
User Mode Linux is way better for that use, much more
The hangup of a channel depends on OpenH323. The driver just initiates
the call clearing and wait for a response from the library (through a
callback function). That response contains the call clearing reason and
the call duration. Of course there is a timeout that ensures that the
library will
I don't have any direct experience with the MD110's and Asterisk, but I
would envisage the MD110 digital phones are very much a proprietary
protocol, as with Nortel digital phones, you can't mix and match between
different vendors.
It may be possible to get Ericsson (as well as Nortel and others)
Hi,
Has anyone managed to use a bluetooth headset or phone with their install
of Asterisk?
What I had in mind was either have a headset paired with the server and
use that to answer/make calls in some way, or forward the calls to my
mobile via bluetooth if that's possible.
I can use DIAX
Hello,
Thanks for your capi.conf! It works great! I made the changes,
restarted Asterisk and made 3 calls with success.
Thanks again,
Gunnar Schaller
Hi Gunnar,
here is our capi.conf for two controllers on two different ISDN lines
;
; CAPI config
;
;
[general]
nationalprefix=0
Hi,
I hope this can help others, so this is it.
Use it at your own risk. I have test it on 3 separate systems without any
problem.
Take care to edit the following files taking into consideration your own
settings.
If you have all the CDR info in the Master.csv too, then delete all the data
from
On Wed, Jun 02, 2004 at 09:27:14AM +0200, Gunnar Schaller wrote:
...cut
chan_capi.c:1147 capi_request: didn't find capi device with outgoing
msn = 7502. you should check your config!
app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI'
== Everyone is busy at this time
Hi Kevin et al,
I have raised this as a flag with GV. They are on the case. Do you have a
Grandstream or similar? It does work on that when configured to use GV
directly so it is working at some level.
Mark
Kevin said:
Thanks for your suggestion. I will give it a try. The other issue I
Working perfektly over E1 link
I have MD110 with 13 E1 link and 3 link is on asterisk over digium card
Christopher Lee wrote:
I don't have any direct experience with the MD110's and Asterisk, but I
would envisage the MD110 digital phones are very much a proprietary
protocol, as with Nortel
Hello,
we have a PRI (E1) to a carrier and a second one to a legacy PBX:
DTAG ---pri * -- Hicmo
(PSTN) |
|
Sip
and
more
Many normal inbound calls are direcly routed to the hicom.
Outbound calls from the Hicom go
Hi all,
Anyone know how put my X101P cards to answer at different ring times ?
Like x101P(a) Answer at 3 rings
x101p(b) Answer at 4 rings
My * it's connected into a PBX thats when receive a call send to two lines at same times a
ring.
(So i must have a way to just put one channel to answer
Hi everybody,
I'm trying to develop a web application for controlling if SIP users
are registered in * or not, and show it in a web.
My problem is that I don't now if it's possible to do a Shell Script to
control this:
1. Connect to console.
2. Execute command.
3. Obtain users registered.
4.
On Wed, 2004-06-02 at 04:27, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Leo Ann Boon [EMAIL PROTECTED] wrote:
The new TDM400P with FXO doesn't take up any IRQ. I've 2 boards and both
are not using any IRQ.
Weird - does that mean they can't provide Zaptel timing like the X100P
On Wed, 2004-06-02 at 01:01, Tracy R Reed wrote:
Note however that this WILL NOT work if one of the devices you are calling
is on a Zap channel. I have a PRI and I would love to ring my cell phone
AND my desk phone (SIP) at the same time but if I try only the Zap
interface rings. I posted
Hi,
XISCOAIR wrote:
Hi everybody,
I'm trying to develop a web application for controlling if SIP users
are registered in * or not, and show it in a web.
My problem is that I don't now if it's possible to do a Shell Script to
control this:
1. Connect to console.
2. Execute command.
3. Obtain
It seem there are trouble with some sound file about checksum
calculation
By example I have a wav file = 99kb after converted in ul = 39 kb , but
makering give me checksum error !!
I trying a wav file recorded with voice recorder, work fine , just
chunk error message
checksum before = db8e
I have this two ip at the same machine, but I tried it using the both
address, the result is the same.
Kind regards,
Miguel
Date: Wed, 02 Jun 2004 13:50:05 +1000
From: Adam Hart [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Firefly version
Reply-To: [EMAIL PROTECTED]
On Wed, 2004-06-02 at 00:33, Wim Kerkhoff wrote:
Following problems have been observed, and are preventing us from
dumping our existing Nortel Merdian PBX:
1. echo at beginning of call for several seconds, even with various
combinations of echocancel and echotraining in zapata.conf
Echo is
My DNS gui(Cpanel/WHM) only allows the following options for entry type:
A6
CNAME
MX
NS
PTR
TXT
WRK
Does anyone know if any of these options are acceptable substitutes for an
SRV record, or do I need to put in a ticket to have a SRV record
specifically created for me?
-
Andrew Thompson
Hi all, I try to do next transfer:
A person contact with me, I would like transfer to other person
in next manner. I call to other person and when I say who wants talk
with him I hangup phones an call is redirect automatically to other
person:
1. call to me
2. Hold the
Eric Wieling wrote:
On Wed, 2004-06-02 at 01:01, Tracy R Reed wrote:
Note however that this WILL NOT work if one of the devices you are calling
is on a Zap channel. I have a PRI and I would love to ring my cell phone
AND my desk phone (SIP) at the same time but if I try only the Zap
interface
Hi,
I have installed Asterisk with sip clients and an ISDN card from Billion.
From an ISDN phone I can dial the Asterisk and hear the welcome message,
hear the echo test etc.
I want to use Asterisk as a gateway between PSTN and SIP so that callers
to my ISDN will be transferred to my fwd
Matteo Brancaleoni [EMAIL PROTECTED] wrote:
[...]
User Mode Linux is way better for that use, much more efficient.
VoIP-only Asterisk also works nicely under vservers (see
www.linux-vserver.org), which is even more efficient than UML.
--
PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full
Please, send to me.
Kind regards,
Miguel
Date: 2 Jun 2004 04:39:39 -
From: muralikrishnan lakshmanan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: RE: [Asterisk-Users] New Firefly version
Reply-To: [EMAIL PROTECTED]
This is a multipart mime message
I too have the same problem on a few units, but not on others. I also
have been having difficulty hooking up multiple lines from one Sipura
to the same multi-line phone system (seems to create a line cross) but
have no problems with either cisco or dlink boxes. In general they are
nice units,
On Wednesday 26 May 2004 19:42, Jon Lawrence wrote:
It looks like my missing digit problems are down to the dect phone I have
connected to my handytone ata-286. When i have my Binatone dect connected,
I only get the first 8 digits, if I connect my panasonic dect then I see
all the digits -
Stig Hess wrote:
Now if I could only get my GS phones to load the ring tone files. The
TFTP log shows all the requests for the usual boot files and the cfg
files but NO requests for the ring tones, not even file not found
responses. I can't believe that this is the tftp server. I have tried
it
Tony Mountifield wrote:
Make sure the ring tone files are no bigger than 65536 bytes.
Earlier versions of my program didn't check for this, but the latest
one does.
That's potentially important information, but even still, the tftp log
would show the phone requesting the file, even if it rejects
Hi Brian,
Brian Cuthie wrote:
BTW, anyone know how to get the SPA-2000 do drop loop current
momentarily when the other end hangs up?
-brian
There is a web configuration option to reverse the polarity in the
latest 2.0 firmware.
--
Nicolas Gudino
House Internet S.R.L.
Buenos Aires - Argentina
On Wed, 2004-06-02 at 07:41, Sergio Serrano wrote:
Hi all, I try to do next transfer:
A person contact with me, I would like transfer to other person
in next manner. I call to other person and when I say who wants talk
with him I hangup phones an call is redirect automatically to other
I have the same problem, I have a domain name but I do not want to pay
for DNS services... I kept on trying to find a place where I can get SRV
records from but none of the free DNS services provide them. I've tried
ZoneEdit, DNS Park, etc. I've seen one which there might be a
possibility,
Bartosz Jozwiak wrote:
Hello,
Did anybody successfully tried upgrade Adtran TSU 600 to
firmware which is working properly with T100P and asterisk ?
B.
Yes, but it was a while ago (last August). I currently have the TSU600
with 2 FXO/1 FXS cards running on a T100P with the only problem being
Andrew Thompson wrote:
Does anyone know if any of these options are acceptable substitutes for an
SRV record, or do I need to put in a ticket to have a SRV record
specifically created for me?
As with email you technically don't need MX records, an A record will
also work fine. I'm pretty sure
XISCOAIR wrote:
Hi everybody,
I'm trying to develop a web application for controlling if SIP users
are registered in * or not, and show it in a web.
My problem is that I don't now if it's possible to do a Shell Script to
control this:
1. Connect to console.
2. Execute command.
3. Obtain users
On Wed, 2004-06-02 at 13:40, Andrew Thompson wrote:
My DNS gui(Cpanel/WHM) only allows the following options for entry type:
A6
CNAME
MX
NS
PTR
TXT
WRK
Does anyone know if any of these options are acceptable substitutes for an
SRV record, or do I need to put in a ticket to have a
Sergio Serrano wrote:
Hi all, I try to do next transfer:
A person contact with me, I would like transfer to other person
in next manner. I call to other person and when I say who wants talk
with him I hangup phones an call is redirect automatically to other
person:
1. call to me
User Mode Linux is way better for that use, much more efficient.
Matteo.
I am using user mode Linux very successfully to run as many asterisks as
I need. Besides asterisk, UML is my other favourite open source
project with which I am involved developing complete turn key solutions
(including
Nicolas Gudino wrote:
Hi Brian,
Brian Cuthie wrote:
BTW, anyone know how to get the SPA-2000 do drop loop current
momentarily when the other end hangs up?
-brian
There is a web configuration option to reverse the polarity in the
latest 2.0 firmware.
Yeah, I saw that too. But it doesn't always
Hi,
I have Debian Linux with kernel 2.6.6. The all packages compiled
except ZAPTEL where I have the following error:
voipgw:/usr/src/zaptel# make
make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory `/usr/src/linux-2.6.6'
CC [M] /usr/src/zaptel/zaptel.o
Hi!
This is possible? There are any best way to implement this?
Yes, look at asterisk -rx command
That command then can be sip show peers or database show sip.
Here is an example of a related CRON job that I use for restart:
# Restart Asterisk PBX once a day to prevent any problems from
On Wed, Jun 02, 2004 at 03:14:58PM +0200, Stephen Rosebush said:
I have the same problem, I have a domain name but I do not want to pay
for DNS services... I kept on trying to find a place where I can get SRV
records from but none of the free DNS services provide them. I've tried
ZoneEdit,
www.xname.org =)
-Original Message-
From: Stephen Rosebush [mailto:[EMAIL PROTECTED]
Sent: 02 June 2004 2:15 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] DNS SRV records
I have the same problem, I have a domain name but I do not want to pay
for DNS services... I kept on
I just did it today successfully. :)
The one BIG problem I have is: there is no battery when I pick up a phone
connected to FXS port.
My adtran tsu 600 has 24 fxs ports. Please could you tell me what kind of
configuration you
have in your adtran for fxs ports and what kind of configuration you
Is there a Linux tool that will splice several gsm sound clips together
into one clip?
In my agi script, I would like to use 'get_data' with one clip instead
of multiple 'stream_file' so the user doesn't have to listen to the
entire spiel before responding.
Thanks,
--
Michael Welter
I know that way, but some person ask for me for first way to do
transfers.
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Stephen R.
Besch
Enviado el: miércoles, 02 de junio de 2004 15:37
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Re:
I installed your wakeup agi script and it works well. There are some
typos on the wiki page--the printf format string seem to have been
corrupted. Also, I need a Linux tool to splice a series of gsm audio
clips together in order to use one 'get_data' instead of multiple
'stream_file'
In article [EMAIL PROTECTED],
vv [EMAIL PROTECTED] wrote:
It seem there are trouble with some sound file about checksum
calculation
By example I have a wav file = 99kb after converted in ul = 39 kb , but
makering give me checksum error !!
I trying a wav file recorded with voice recorder,
I've been watching to see if this problem comes up with anyone elses
firefly - but so far i'm the only one experiencing the problem.
When I connect to either my asterisk server or FWD all goes well on the
first call. I can hear and talk. But every call after the first one I
end up with no
Hi there,
I want to buy a IP Phone but i found it rather ro ask the asterisk
mailinglist...
Does anybody uses a Grandstream 1XX and have probles with the asterisk?
Wich phone would you me rate?
in a price range from 100 - 150$ ?
Best Regards,
Mark Nicolas
Stephen R. Besch wrote:
Not as far as I know, at least not exactly the way you have outlined it.
Try this:
1. call comes to you
2. You hold the call and call other person.
3. You say Someone wants to talk to you, OK, thanks
3a. Other person then hangs up.
3b. You flash back
Does anybody have any experience with the ZyXEL Prestige 2000W? I am
having problems with the line tear down when I call another extension.
If nobody picks up at the other end when I hangup the 2000W, the other
extension continues to ring. Is there any way to hangup a SIP call if
there is no
All,
I have been beating my head against a wall trying to figure out how I would
implement a separate moderator code and participant code for the same
conference using meetme, the deal is I dont want the participants to be able
to join until the moderator is in the conference.
Is it possible to
Anyone know how to place asterisk in initab so that it is loaded at boot and will respawn if the process goes down?
I
Duane wrote:
Andrew Thompson wrote:
Does anyone know if any of these options are acceptable substitutes
for an
SRV record, or do I need to put in a ticket to have a SRV record
specifically created for me?
As with email you technically don't need MX records, an A record will
also work fine. I'm
Michael Welter wrote:
Is there a Linux tool that will splice several gsm sound clips together
into one clip?
In my agi script, I would like to use 'get_data' with one clip instead
of multiple 'stream_file' so the user doesn't have to listen to the
entire spiel before responding.
Thanks,
cat
On Wed, 2004-06-02 at 09:22, Michael Welter wrote:
Is there a Linux tool that will splice several gsm sound clips together
into one clip?
In my agi script, I would like to use 'get_data' with one clip instead
of multiple 'stream_file' so the user doesn't have to listen to the
entire
I am having a similar problem. It is not frequent, perhaps once in
80-100 calls.
CVS-HEAD-05/08/04-21:57:50 using Cisco 7960 6.3 and X100P
--__--__--
Date: Tue, 1 Jun 2004 21:04:14 -0700 (PDT)
From: Bruce Komito [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] determining cause
Any one see these? Are they benign, or is some system tuning required
to remove them?
Can't seem to find a resolution in the archives. If you have a link, it
would be appreciated.
Jun 2 10:58:58 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP
codec 19 received
Jun 2 10:58:59
On Wed, 2004-06-02 at 14:56, Miroslav Nachev wrote:
I have Debian Linux with kernel 2.6.6. The all packages compiled
except ZAPTEL where I have the following error:
voipgw:/usr/src/zaptel# make
make linux26
F
___
Asterisk-Users mailing list
cat 1.gsm 2.gsm 3.gsm new.gsm
works fine
James
On Wed, 2 Jun 2004, Michael Welter wrote:
Is there a Linux tool that will splice several gsm sound clips together
into one clip?
In my agi script, I would like to use 'get_data' with one clip instead
of multiple 'stream_file' so the user
On 1-Jun-04, at 6:57 PM, Leo Ann Boon wrote:
The new TDM400P with FXO doesn't take up any IRQ. I've 2 boards and
both are not using any IRQ.
Can you please double-check? I have 2 servers, each with a tdm400p +
quad-fxo ... with both of these installs, the card is assigned an
interrupt:
# more
On Wed, 02 Jun 2004 08:22:23 -0600, Michael Welter [EMAIL PROTECTED]
wrote:
Is there a Linux tool that will splice several gsm sound clips together
into one clip?
In my agi script, I would like to use 'get_data' with one clip instead
of multiple 'stream_file' so the user doesn't have to
On Wed, 2004-06-02 at 09:44, Tony Hoyle wrote:
Ugh. So Asterisk doesn't handle transfer?
Every company phone system I've ever used has not required 3a-3d. It
looks like a real hack to do so.
It anyone working on implementing this?
As far as I can tell it's a limitation of the phone,
I am learning about SIP phones little by little. I've been working
with * for about 2 weeks now with FXS and FXO ports and analog phones,
but we want to evaluate the utility of going to SIP phones directly
from here rather than investing in analog phones first.
One of the questions I have is
ax:2:respawn:/usr/sbin/asterisk -vvvcgf
Nate Turnbow
Systems Engineer
CHG Companies
On Wed, 02 Jun 2004 10:01:34 -0500
Terry Goodwin [EMAIL PROTECTED] wrote:
Anyone know how to place asterisk in initab so that it is loaded at boot
and will respawn if the process goes down?
I
--
Hi Bruce,
I am doing smth similar in one appli.
just using MeetMecount inside an AGI script, btw I know
when there is someone in the conference and I can let the user go in or
not!
Thr trick is to save the result in a variable and then use GET
VARIABLE to get the nb user!
/
snprintf(
Hi,
I work for an ISP/CLEC, and we have developed our own OSS (Operations
Support System), which handles all billing, sales, provisioning, and
support issues. When it was originally being designed, the idea was to
integrate it with Asterisk.
Other than Caller-ID information (so that past trouble
Hi,
I have really googled and read the wiki but I still no idea, how to supress
the fax recognizion.
Our users are not able to fax and that is bad... Could you give me an hint,
please?
Thanks
Felix
Hello,
we have a PRI (E1) to a carrier and a second one to a legacy PBX:
DTAG
On Wed, Jun 02, 2004 at 11:25:26AM +0200, Tor Houghton wrote:
Hi,
I have two softphones connected to an Asterisk stable. I have two
extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be
completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on
extension 2000 will
I put an update on the website. It fixes the IAX calls crashing the
program. ANd added Voice Mail checking. It will tell you how many new
and old voice mails are in your box.
http://www.easyhomenetworks.com/AstRec/
Kyle
John Fraizer wrote:
Kyle Hagan wrote:
Ok I have a testing version
Hi,
-Original Message-
I have been beating my head against a wall trying to figure
out how I would implement a separate moderator code and
participant code for the same conference using meetme, the
deal is I dont want the participants to be able to join until
the moderator is
John Fraizer wrote:
Spoken like a true n00b13.
If the current SIP bug isn't annoying enough to push people away from
asterisk you just have to chip in your 2 cents worth to push things that
little bit more...
You can *sometimes* get away with not having MX records. You can
*sometimes* get
Hi all,
I just installed chan_h323 to interface to a H.323/ISDN gateway.
It works really well after two days learning and testing except one thing
somebody of you may have an answer to:
If I call in from PSTN to a busy asterisk SIP extension I can see a SIP status
486 BUSY, but don't get it
Hello Fran,
I try with make linux26 but the result is the same:
voipgw:/usr/src/zaptel# make linux26
make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory `/usr/src/linux-2.6.6'
CC [M] /usr/src/zaptel/zaptel.o
/usr/src/zaptel/zaptel.c: In function
Hi,
I've just tried it in my setup and it does not occur anymore.
I did see this problem when I first got the phone, but since then I've
updated everything, and it appears to have gone away :-)
asterisk CVS-04/10/04-15:32:35
ZyXel P 2000
Software version WJ.00.0a bootrom version B.00.13 release
Either put the channel that your fax machine is on in a context without
exten = fax or remove the exten = fax from the context the fax
machine is in. The exten = fax is ONLY needed if you want to share an
inbound line between fax and voice.
On Wed, 2004-06-02 at 11:01, ePyron Felix Deierlein
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