Sounds like a good solution to me!
Could potentially use externpass then use flock(),
fcntl() et al. every once in a while to write all
passwords back to voicemail.conf.
Thanks for the insight.
--- Michiel van Baak <[EMAIL PROTECTED]> wrote:
> On 17:05, Wed 29 Nov 06, Scott Keagy wrote:
> > If
And as I wrote before, Asterisk <-> mySQl connection is already up and
runnig (for CDR). So it just would have been quick and easy if Asterisk
could have used the same path for audio data.
O.K., lets invest some time in installing ODBC.
NOrbert
Norbert, mate, I don't know why you're havi
On 11/29/06, Brian Capouch <[EMAIL PROTECTED]> wrote:
Complaints are always considered, but calling the developers childish
and repeating that complaint over and over in an email isn't likely to
do much to advance the cause you've taken on.
Sorry about the rant. I apologize for making the ch
AFAIK, ODBC helps you have any DB underneath, be it MySQL, PGSQL, etc.,
so why not go ahead with it?
cheerz
- Ben.
Norbert Zawodsky wrote:
Hi Peder,
I asked the same question some time ago.
Never got any answer... :-(
Norbert
Peder @ NetworkOblivion schrieb:
Is the storage of actual
Just curious if anyone knows of any hacks to enable announce entry/exit
in MeetMe conferences with SIP (as opposed to ZAP) channels since the |i
option will not work with SIP.
Thanks,
Mike
___
--Bandwidth and Colocation provided by Easynews.com --
a
At voip-info.org they show the following example
exten => s,1,Set(foo=${STAT(s,/var/t3)})
which I guess is suppose to work and make foo = size of t3
I did the following
exten => 542,1,Set(s1=${STAT(e,"/var/lib/asterisk/t1")})
which should set s1 = 1 if the file exists and 0 if not.
b
I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card
connected to a POTS line and a phone set (physical extension). I've got
all incoming calls launching directly into an AGI script. I'd like to do
the same for the physical extension. In other words, when picking up the
hand set
Brad Templeton wrote:
My understanding was that the "port=" field on a particular SIP
channel defines the port used at the remote end, ie. The
user's phone will be talking on port X of their IP address, it
does not alter what SIP port Asterisk is listening on on the
Asterisk box.
The host a
Lacy Moore - Aspendora wrote:
I think that's the problem with the Asterisk community right now.
Anytime something is suggested, the response is either write it yourself
or deal with what is there.
Do you have experience with other big, complex Open Source projects? Do
you know of any whe
Agree. When I meant to say is that the step-by-step guide is for hooking
up the PAP2 to the A/C line, network, and telephone set.
This should give you a pretty good idea of the educational level of these
people.
- Daniel
-Original Message-
From: "Zeeshan Zakaria" <[EMAIL PROTECTED]>
Sent
That's how it's setup. However, setting up a VPN on the client's side is
not done thru the provisioning :)
- Daniel
-Original Message-
From: "Andrew Joakimsen" <[EMAIL PROTECTED]>
Sent: Wed, November 29, 2006 8:39 pm
To: [EMAIL PROTECTED], "Asterisk Users Mailing List - Non-Commercial
Dis
Either write what you want, or learn to use what we have and hope
that SLA when it appears is better. Parking is not the best solution,
I think that's the problem with the Asterisk community right now. Anytime
something is suggested, the response is either write it yourself or deal
with what
On Wed, Nov 29, 2006 at 04:49:38PM -0700, Joseph wrote:
> What I have is that each device is listening on different port ex.
>
> [pstn-5665] ; incoming/outgoing calls on FXO port
> type=friend
> ...
> port=5066 ; port on Pstn line
> ...
>
> [318] ; incoming/outgoing calls on FXS Sipura-2002
> ty
Hi all,
I noticed that when an user register asterisk, asterisk will update
the corresponding record in db user table. However, if the user
register failed, maybe wrong password. There is no record in
database. How can I log those register records, including successful
and failed login to the
On Wed, Nov 29, 2006 at 06:05:31PM -0500, Steve Sobol wrote:
> On Mon, 27 Nov 2006, Brad Templeton wrote:
>
> > On Mon, Nov 27, 2006 at 11:20:27PM -0500, Andrew Joakimsen wrote:
> > > Can you explain how ValetParking and twenty minutes worth of "dialplan
> > > creativitiy" can't do the same EXACT
Andrew - I have been told they have no plans to introduce US
distribution or availability on these products in the foreseeable
future. I was told this by one of the channel managers from Siemens. I
received some eval units of some of the Siemens SIP products from a
reseller in EU, and they are qu
Hey Vincent -
1. What are the settings (in sip.conf?) to tell Asterisk to use specific
ports for RTP?
I guess you didn't see my reply earlier today - that setting is in rtp.conf
2. With this kind of setup, does Asterisk stay in the loop to forward RTP
packets, or do X-Lite at home and the V
Inital setup for testing will be 2-4 channels in order to prove the concept.
When successful we may include some PBX systems that do not have available T-1
slots.
On Wed, 29 Nov 2006 19:18:50 -0800, "Tom Lynn" <[EMAIL PROTECTED]> wrote:
> How many channels do you require? I'd favor T1 for a
At 03:49 PM 11/29/2006, you wrote:
That data is no easier to parse than the output generated from:
Action: Command
Command: sip show peer
Note the colons used both as a field delimiter AND stuck in the regcontact.
Actually, finding the data between a CR and a colon get's you the
data label
How many channels do you require? I'd favor T1 for a few reasons. Higher
port density means fewer cards per system, which will mean fewer
interrupts. T1s won't require you to tune analog levels. Echo probability
will be lower.
On 11/29/06, asterisk-robert <[EMAIL PROTECTED]> wrote:
We are
Asterisk 1.2.7
Redhat 9
I have DiDs from two different ITSP both set up as IAX2. Each one
works when it's the only one in my iax.conf, but when I have them both
defined in iax.conf at the same time, only one will work. My iax.conf
is provided below.
Any ideas how to fix? I'd like to use both
g beeping sounds at
regular intervals no matter which phone we use. Does anyone know why?
We are using a diqium tdm card.
Thanks
Kim
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Hi All,
I have a problem in configuring in asterisk.
I configure asterisk meetme.conf and extension.conf, but when i transfer
call to conference it give me this message and asterisk kill it self.
Ouch ... error while writing audio data: : Broken pipe
If any one knows abou
I didn't have a problem writing a client in C.
There were a couple things to worry about
I have teh code in svn on sourceforge and havn't done much w/ it anymore
since i've moved jobs. Anyone want it?
e-mail me directly. i'l freely hook it up.
On 11/29/06, Richard Lyman <[EMAIL PROTECTED]>
http or tftp provisioning is what you do in this type of business. I can't
imagine how can one do this business without using remote provisioning. PAP2
devices have excellent support for http, https and tftp. Using remote
provisioning, you can easily control their ports, DIDs and all configuration
What if you used a better defined dialplan?
On 11/28/06, Crazy Boy <[EMAIL PROTECTED]> wrote:
Hi Friends,
I am facing a strange problem with DISA. I have installed and configured
Trixbox. I've created a secret extension i.e., 555 and called this
extension in Digital Receptionist using custom e
Does anyone know where to source the Siemens Gigaset phones in North
America? I called 1-800-SIEMENS and was told the Gigaset range is no longer
marketed here since a few years ago. How far from being FCC compliant is the
DECT standard?
On 11/29/06, Eugen Leitl <[EMAIL PROTECTED]> wrote:
I've
Why don't you use the provisioning? it runs through http server...
On 11/29/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
That's not possible. These are residential people who hardly know enough
to hook up their PAP2 with detailed step-by-step instructions on hand and
support on the phone :)
At 03:08 PM 11/29/2006, you wrote:
It'd actually be great, because the behavior would be almost
identical to that of our old PBX.
Either write what you want, or learn to use what we have and hope
that SLA when it appears is better. Parking is not the best solution,
but it works. If you have A
TP'n to follow flow..
yes i understand the out is similar/same.
i was attempting to get james to note that there already are manager
commands.
as a note for parsing the manager interface..
you will probably find it easier if you parse as a block, meaning *till*
a blank line.
then parse the
Tony Mountifield wrote:
In article <[EMAIL PROTECTED]>,
Richard Lyman <[EMAIL PROTECTED]> wrote:
just wait till you get a 'hiccup' that causes a line to get cut off,
drop a char, and continue on next line.
(examples below)
I've made heavy use of the Manager interface for over 2 years
Hello
When I make calls from home to the PSTN by going through the Net ->
Asterisk -> the Net -> VoIP provider -> PSTN, I get no sound either way. I
assume it's because I must tell Asterisk to use fixed ranges of UDP ports
and map ports accordingly on the NAT firewall under which it is located
That data is no easier to parse than the output generated from:
Action: Command
Command: sip show peer
Note the colons used both as a field delimiter AND stuck in the regcontact.
CodecOrder: ulaw,g729,gsm
Status: UNKNOWN
SIP-Useragent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098
Reg-Contact : si
What I have is that each device is listening on different port ex.
[pstn-5665] ; incoming/outgoing calls on FXO port
type=friend
...
port=5066 ; port on Pstn line
...
[318] ; incoming/outgoing calls on FXS Sipura-2002
type=friend
...
port=5069 ; port on FXS line
...
etc.
Though I'm not sure if
On Fri, 24 Nov 2006, Brad Templeton wrote:
> As I was noting in an earlier message, the parking lot concept is to my
> view not a thrilling interface at best, and I can't see many times one
> would want it in a SOHO environment.It seems best for a large PBX
> where people are moving to random
Can you please tell me what am I missing? If I don't make a choice before the
last prompt starts playing the call is falling through. I want the menu to be
replayed if a choice is not made, but the call gets disconnected and I see the
following message in the Asterisk BE colsole.
== Pars
> -Original Message-
> From: James Texter [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, November 29, 2006 3:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] What's up with the Manager Interface?!?!
>
>
> Doug,
> Your issue isn't with t
On Mon, 27 Nov 2006, Brad Templeton wrote:
> On Mon, Nov 27, 2006 at 11:20:27PM -0500, Andrew Joakimsen wrote:
> > Can you explain how ValetParking and twenty minutes worth of "dialplan
> > creativitiy" can't do the same EXACT thing you are describing? Sometimes the
> > simplest answer is never th
On Wed, Nov 29, 2006 at 05:05:13PM -0500, Scott Keagy wrote:
> If you have enough users where this comes up as a real issue, I'd
> recommend migrating to Asterisk Realtime voicemail,
[ snip ]
If you have expected a different type of reply, consider discussion this
in asterisk-dev .
Yes. The c
ecuting Set("SIP/1656-b7d10740",
"AGENTFILENAME=1656-20061129-183350-") in new stack
-- Executing Monitor("SIP/1656-b7d10740", "wav|1656-20061129-183350-|m") in
new stack
-- Executing Queue("SIP/1656-b7d10740", "NOC") in new stack
Ed Nuñez
James Texter wrote:
Doug,
Your issue isn't with the manager. It's with the CLI output you are
trying to hijack via manager :D If you run "sip show peer 2944093" in the
CLI, you'll see a blank line, followed by a line that is "* Name". It
appears what you really want is a manager Action to
On 17:05, Wed 29 Nov 06, Scott Keagy wrote:
> If you have enough users where this comes up as a real issue, I'd recommend
> migrating to Asterisk Realtime voicemail, then can have row-level locking
> etc. if you use the right kind of storage engine... I've found problems using
> the dial-by-name
Greetings,
I am cutting my teeth with SIP phones and my first issue is getting a
Cisco 7940 to Authenticate with my VoIP provider (BBTelsys).
I did read some notes on the vo-ip website about 7.5 being the better
firmware version. Has anyone had trouble with 8.2 and SIP registering?
Should I just
phpldapadmin is pretty nice. I was using 2-3 different ldap clients to get
the job done until I got over my php bias and installed it. It lets me do
everything I want, without crashing.
On 11/27/06, Steven Baker <[EMAIL PROTECTED]> wrote:
Hello All,
we are using asterisk+openldap. Do is there a
> -Original Message-
> From: James Texter [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, November 29, 2006 3:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] What's up with the Manager Interface?!?!
>
>
> Doug,
> Your issue isn't with t
Hi all
For some dumb reason I decided to upgrade from Mandriva 2006 to 2007, thinking
I could install asterisk all over again. Anyway I did install asterisk, zaptel
and libpri. After install I ran modprobe zaptel which said "zaptel not found".
Thanks to help on this mailing list I had a fix
In article <[EMAIL PROTECTED]>,
Richard Lyman <[EMAIL PROTECTED]> wrote:
> just wait till you get a 'hiccup' that causes a line to get cut off,
> drop a char, and continue on next line.
> (examples below)
I've made heavy use of the Manager interface for over 2 years now, and
have never seen the
Anyone know if it posible to make voice promps native g726 or g711 format?
___
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To UNSUBSCRIBE or update options visit:
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> -Original Message-
> From: Steve Edwards [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, November 29, 2006 2:55 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] What's up with the Manager Interface?!?!
>
>
> On Wed, 29 Nov 2006, Douglas Garst
I have asterisk 1.2.12.1 running with several client phone options. Our
echo cancellation is finally working great. The only problem I seem to
be having is there is background noise including beeping sounds at
regular intervals no matter which phone we use. Does anyone know why?
We are using a d
I have been using Enswitch. Has some bugs but over all works great. It's not
open source but worth the money.
- Original Message -
From: "Guillermo Salas M." <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, November 29, 2006 3:12 AM
S
If you have enough users where this comes up as a real issue, I'd recommend
migrating to Asterisk Realtime voicemail, then can have row-level locking etc.
if you use the right kind of storage engine... I've found problems using the
dial-by-name directory with realtime voicemail, but it seems you
Doug,
Your issue isn't with the manager. It's with the CLI output you are
trying to hijack via manager :D If you run "sip show peer 2944093" in the
CLI, you'll see a blank line, followed by a line that is "* Name". It
appears what you really want is a manager Action to show a sip peer, in
wh
On Wed, 29 Nov 2006, Douglas Garstang wrote:
G. Here's another example...
Action: Command
Command: sip show peer 2944093
Response: Follows
Privilege: Command
* Name : 2944093
Secret :
MD5Secret:
Context : 180o_CallStart
Subscr.Cont. : 180o_WatchBLF
Why the HE
I'm wondering if anyone is having problems when
multiple users concurrently change their voicemail
passwords.
Consider the following scenario (based on
vm_change_password() in app_voicemail.c):
- user1 wishes to change his password so
voicemail.conf is opened and read into a buffer
- user1 change
Thanks for the response!!!
I enabled debuging in the menuselect configuration for compiling
asterisk 1.4 beta3. In logging.conf enabled debug loggin to the
/var/log/asterisk/debug file and to the console. Restarted (not just
reload) asterisk and there is plenty of general debugging info in the
> -Original Message-
> From: Douglas Garstang
> Sent: Wednesday, November 29, 2006 12:26 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] What's up with the Manager Interface?!?!
>
>
> > -Original Message-
> > From: Michael Collin
Hi List,
I have a Polycom 601 that when the user is on the phone they only hear one beep
and the CID of the second incoming call is not shown. Is there a way to have
the CID show up for the second call ? And a way to configure the phone to beep
more often if there is another call coming in. The
I thought I sent this out.. but don't see it so apologies if it
went already.
I am trying to get streaming MOH working but haven't been able to.. I
am running 1.2.x
Based on people's suggestions in other e-mails I've tried:
[scanner]
mode=custom
application=/usr/local/bin/mpg123 -s --mono -y
I get an error when I do a make install
[EMAIL PROTECTED] app_swift-0.9.5]# make install
gcc -g -Wall -D_REENTRANT -D_GNU_SOURCE -fPIC -DCHANNEL_HAS_CID
-DNEW_CONFIG -I/opt/swift/include -c -o app_swift.o app_swift.c
app_swift.c:49: warning: type defaults to `int' in declaration of
`STANDARD_
Hi Noah,
Noah Miller wrote:
> Hi Norbert -
>
> Just a thought: You could go the other way - share a volume on a
> separate webserver, and have the asterisk box connect to the webserver
> via NFS as a client, and store the voicemail on the NFS share. While
> I don't have any exact numbers, it seem
On Wed, 2006-11-29 at 16:38 +0100, Timothy Parez wrote:
> I get the following with debug on:
>
> P[ 3] I IND :SETUP oad:497978546 dad:50556010 pid:10 state:none
> P[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:
> P[ 3] --> info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0
> P[ 3]
Hi Vincent -
Here's what I did on the X-Lite at home in the Topology section:
IP address : Discover global address
STUN server : Discover server
Port used on local computer : Manually specify range 8000-8019
Here are the ports that I forwarded from my NAT router at home:
UDP 5060
UDP 3478 (STUN
> Here's a good example. I'm trying to get SIP blf. I managed to split
my
> result into a list of lines by splitting on ANY of \r\n, \n or \r. I
was
> going use the column headings from the third line as my keys for my
> dictionary/hash, rather than hard coding them. Notice anything? The
'Call
> I
Hi Norbert -
>> I want to store all of my voicemail stuff in a database so that I can
>> give users web access to it, but I don't want to run web services on my
>> * server itself. If it is all in a DB, I can have a web box and a
>> separate SQL box and none of it should affect *.
>
> Yes, you
> -Original Message-
> From: Matt Florell [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, November 29, 2006 12:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] What's up with the Manager Interface?!?!
>
>
> On 11/29/06, Douglas Garstang <[EM
On 11/29/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
The Asterisk Manager Interface is driving me nuts.
Whoever wrote it should be drawn and quartered.
You would need a ton of rope and a few hundred horses for that :)
The Manager API code is distributed across dozens of source files in
the
Dear List,
I'm looking for a coder/developer that can modify oh323 return codes on
asterisk
Example on based on SIP and h323.
Right now we are receiving :
Call Rejected (code 21)
Network Out of Order (code 38)
Need to able to replace dose codes with
-> No Circuit/Channel Available (code 34)
P
We are thinking of setting up an Asterisk system to route calls between 2 of
our factories. Our idea is to connect an Asterisk box to each PBX and then use
SIP(or IAX) to truck between the 2 systems on our internal network.
I would be interested in any ideas regarding the connection points:
> -Original Message-
> From: Michael Collins [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, November 29, 2006 11:20 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] What's up with the Manager Interface?!?!
>
>
> > Sometimes the data comes back
Eric,
It looks like the definition for PTHREAD_MUTEX_RECURSIVE is within an
#ifdef __USE_UNIX98 (on Fedora Core 6, anyway). You could try defining
it within the Makefile. Similar to the _GNU_SOURCE definition in the
app_cepstral.so: app_cepstral.c stanza.
Bob...
On Wed, 2006-11-29 at 13:38 -05
Douglas Garstang wrote:
The Asterisk Manager Interface is driving me nuts.
Whoever wrote it should be drawn and quartered.
Sometimes the data comes back separated by \r\n, and sometimes it's separated
by \n.
The whole thing is completely inconsistent, and trying to write any kind of API
for it
I am installling on a scratch asterisk running white box linux (fedora)
Does anyone know where to find them after the rpm runs. I am looking for
ircd and the perl dependancies. The instructions make a ton of
assumptions, so I am not sure what is happening here.
Jordan Novak
Senior Telecommunicati
Hall, Eric M. wrote:
Using
this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt
This
is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3
I
get the following errors on make install
Any
help would be GREAT!
Thanks
Howdy,
Anybody have any ideas on how to record to a different file each time a call
is transferred by means of the transfer button on Polycom phones? I basically
need to be able to execute StopMonitor and an AGI script each time a call is
transferred without using features.conf for transfers. T
> Sometimes the data comes back separated by \r\n, and sometimes it's
> separated by \n.
> The whole thing is completely inconsistent, and trying to write any
kind
> of API for it is -GHASTLY-
Doug,
What language(s) are you using? Just curious. I've been tinkering with
Perl, POE, and POE::Compo
Ok. So, it's not that I can make Asterisk listen on multiple ports for any
SIP friend, but I could override the port on an individual SIP friend.
So, instead of having something like:
bindport=5060,5080,5081,5082
in the general section of sip.conf, I need to just have
bindport=5060
in the gene
Wow. I didn't know you could do that. So, I could have something like this
in sip.conf:
bindport=5060,5080,5081,5082
and it will make Asterisk listen on all those 4 ports?
- Daniel
-Original Message-
From: "Joseph" <[EMAIL PROTECTED]>
Sent: Wed, November 29, 2006 1:31 am
To: [EMAIL PRO
That's not possible. These are residential people who hardly know enough
to hook up their PAP2 with detailed step-by-step instructions on hand and
support on the phone :)
Thanks,
Daniel
-Original Message-
From: "Tom Lynn" <[EMAIL PROTECTED]>
Sent: Wed, November 29, 2006 12:28 am
To: "Aste
The Asterisk Manager Interface is driving me nuts.
Whoever wrote it should be drawn and quartered.
Sometimes the data comes back separated by \r\n, and sometimes it's separated
by \n.
The whole thing is completely inconsistent, and trying to write any kind of API
for it is -GHASTLY-
Doug.
_
I am attempting my first go at a simple AGI application using PHP (Getting
Asterisk to SAY PHONETIC ABC). I have dabbled with PHP but I am by no means
a professional standard developer.
Can't really say what is wrong with your code since I never did an AGI
in PHP without this class : http://phpa
On 11/20/06, Ralph Liebessohn <[EMAIL PROTECTED]> wrote:
On 11/20/06, Alex Robar <[EMAIL PROTECTED]> wrote:
> Hi Ralph,
>
> Have you setup your PAP2 to allow the 729 codec? I believe you actually
> have to tell it that it's allowed to use that codec before it will work.
>
> Cheers,
> Alex
>
>
Fran when you say "specify the next hop" do you mean the S0 line be an
extension in sip.conf or a context in extensions.conf?
Or should the line simply be tacked on to my [default] context?
Larry
Fran Oliveira wrote:
I think it is wrong. You should specify the next hop with some like this
S0<
Setup:
Office A:
router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv
Asterisk: v.1.2.4
static IP
Office B:
router: Linksys WRT54GL running Linksys firmware v4.30.2
Asterisk: v.1.2.7.1
dynamic IP (using dyndns name)
Office A is set up with refresh dns and cron job for iax2 reload e
Tomer Horn wrote:
Hello list,
I am curious here if anybody here got an experience connecting Avaya to
Asterisk using H323 / T1. I am completely lack of experience with Avaya
and I wanna know if anybody here has connected Avaya to Asterisk using
H323 and managed to stabilize it. Google provide
Using this link
http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt
This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3
I get the following errors on make install
Any help would be GREAT!
Thanks
[CC] app_cepstral.c -> app_cepstral.o
In file
I've got to the point with FWD and IAX that I just connect directly via SIP
to IPKall, using my Asterisk box's address as the proxy. It simply works
better and eliminates another point of failure at FWD.
I also find it helps keep things a little more organized since I can assign
my own internal S
Hi,
I've setup a trixbox 1.2.2 instalation with digium b410p. It uses misdn
driver.
I'm able to place calls and receive calls using this interface.
One thing that happens is when i dial an a number from a sip client to an
number that is routed through b410p and the called party rejects the call
Hi,
Sorry, uncomenting that actually worked.
Now I need to filter on the last two numbers, that shoulnd't be to hard
I guess.
Tim.
Giorgio Incantalupo schreef:
Hi Patrick,
it seems like Asteirsk cannot match any number, try uncomment the two
following lines:
;exten => _X.,1,Dial(SIP/tim
jason wrote:
> last I had heard, pretty much all FWD accounts that were created in the
> past year or so no longer work with IAX. Still don't know why.
>
> Timothy Parez wrote:
>> I've got the same problem here.
>> It can't register anymore --> timeout
>>
>>
>> Brian Capouch schreef:
>>> I hadn't
I get the following with debug on:
P[ 3] I IND :SETUP oad:497978546 dad:50556010 pid:10 state:none
P[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:
P[ 3] --> info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0
P[ 3] --> Bearer: Speech
P[ 3] --> Codec: Alaw
P[ 0] --> * NEW CHANNEL
Hello list,
I am curious here if anybody here got an experience connecting Avaya to
Asterisk using H323 / T1. I am completely lack of experience with Avaya
and I wanna know if anybody here has connected Avaya to Asterisk using
H323 and managed to stabilize it. Google provides mixed comments
r
Hi,
I did, that was my first try,
but it didn't work.
Giorgio Incantalupo schreef:
Hi Patrick,
it seems like Asteirsk cannot match any number, try uncomment the two
following lines:
;exten => _X.,1,Dial(SIP/timothy,30,r)
;exten => _X.,2,Hangup()
Giorgio Incantalupo
On Wed, 2006
Still doesn't work for me.
Still get timeout
Michael Graves schreef:
I'm travelling today but I was just able to use Firefly to login to
FWD via IAX2. I called the echo test with no problems other than the
lousy network in this hotel.
My Astlinux server Also reports that it's registered with
Hello list
We have a situation where calls need to be transfered to another extension.
We are using # to accomplish this but we found this is only working for
calls answered at the original called extension. If the call has been
forwarded to another extension or if the call has been picked up by
Noah Miller wrote:
> Hi Peder -
>
>> Is the storage of actual voicemail messages in a database still limited
>> to ODBC? If so, why?
>
> Yes. Why? Nobody has developed a voicemail solution that directly
> connects to a *SQL database for message storage.
A clear answer :-) Although a sad one :-(
B
Derek Whitten wrote:
> Norbert Zawodsky wrote:
>
>> RR wrote:
>>
>>
>>> Mate, I can't say it with authority but I'm almost certain that the
>>> only DB that a specific driver was written for is MySQL. I think if
>>> you use res_mysql.o you should be able to talk to mySql directly
>>> witho
Thanks for all the help guys.
I cannot load the new SIP image straight on as the SCCP image is very old.
i read the FAQs posted on the lists and it tells me I need to upgrade the
SCCP image to at least 7 before I can load the SIP image.
This is the problem I am having. I cannot load SIP until I
I'm travelling today but I was just able to use Firefly to login to FWD via
IAX2. I called the echo test with no problems other than the lousy network in
this hotel.
My Astlinux server Also reports that it's registered with FWD via IAX2. My
account is a couple years old.
Michael
On Wed, 29 N
Hi Patrick,
it seems like Asteirsk cannot match any number, try uncomment the two
following lines:
> ;exten => _X.,1,Dial(SIP/timothy,30,r)
> ;exten => _X.,2,Hangup()
Giorgio Incantalupo
> On Wed, 2006-11-29 at 13:26 +0100, Timothy Parez wrote:
>> Hi,
>>
>> I'm able to place outgoing calls using
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