I have foloved instructions at this web pages
http://www.omniis.com/ntsgr/cms/page.asp?688 and now I'm able to call
contacts from Outlook. Now I have few questions. When I place a call, my
phone rings before * tries to dial out. Is it posible that * first dials
out, and when other side picks up
Not at all, I am right with you. I am listening to what
Digium is saying and letting them spin their resources on it. They say they have
it working, they say it should work, and they say they will do whatever it takes
to make it work. I personally am finding this rather interesting being in t
I fixed it on my system it now reads the date and time correctly from
the filenames but the by - part is missing in the joined file names as
far as I can tell so I had to leave that part out. This is AAH 1.5 if
that makes any difference. I'll be upgrading to 2.2 after a while but
this install
Kerry is filtering out what he doesn't want to hear so I think breath is being wasted here - practical experince will educate...On 1/4/06, C F <
[EMAIL PROTECTED]> wrote:Look at this rather:
http://www.voip-info.org/wiki/view/Asterisk+tips+findmeLike BJ said try avoiding inband call progress on Zap
Within asterisk, is it possible to detect that an incoming call is a
direct dialing, or forwarded via another place? When a call is being
forwarded via a 3rd party (say, SBC), will it have some indication in
the call packet?
Thanks!
Frank
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1. yes turn off fax detect.
2. i have never seen a PBX do this maybe an old key
system. the dial plan is based on the phone. phones
like the Cisco 7960 have easy to configure dial plans
so you don't have to dial #.
3. X100P card are about a crapy as the get. you will
get a loot better sound from
We currently have about 60 cisco 7940's, which were converted from cisco
call manager to be used for asterisk. We're running 1.2.1 stable on 4
systems (primary server, backup server, gateway, and voicemail). The
phone lines come into the gateway on a digium te405p. The problem we're
having is th
On 1/3/06, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote:
> Have you removing the asterisk include directory before trying version
> 1.2? I think it might be /usr/include/asterisk/ in many cases.
Thanks. Looks like this and make clean worked.
Michael
>
> Michael Stearne wrote:
> > I
Asterisk::LDAP is a perl module for generating Asterisk Stable (versions
1.0 and 1.2) compatible configuration files from an LDAP directory tree.
The package includes everything you need to get started, including the
module itself, schema files and example code.
http://projects.alkaloid.net/conte
Can you provide some details about the system, what version of Asterisk, what kind of phones, what kind of phone lines, etc.
Erick
On 1/3/06, Aaron Daniel <[EMAIL PROTECTED]> wrote:
I've got a slight problem with echo. Basically, most of the outgoingphone calls on our system echo, but as far as
- Original Message -
From: "Mike McMullen" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, January 03, 2006 8:31 PM
Subject: [Asterisk-Users] Raw Hangup messages with IAX2?
Hi All,
I am running asterisk 1.2. I have a softphone conn
Hi All,
I am running asterisk 1.2. I have a softphone connecting from a coworkers
home through their router using IAX2 through our router at the office. Both
have port 4569 for TCP and UDP opened and forwarded to the right pc
and server.
I'm seeing Raw Hangup , src=0. dst=10787
messages show
Mojo with Horan & Company, LLC wrote:
Wow, example by me. I don't read the Wiki enough lately ;)
on this topic - I had to remove a few blank lines in config.php after I
renamed it or I got a header error and nothing displayed - now it is
working as designed
but shows line like
August 20,
And please excuse the formatting there, darn mail client
Pete wrote:
I have some stuck channels that I think I'm going to have to bounce
Asterisk to get rid of, but am curious to know what they are and how
they've managed to accumulate. The show up with a channel identifier
of '(None)' a
Look at this rather:
http://www.voip-info.org/wiki/view/Asterisk+tips+findme
Like BJ said try avoiding inband call progress on Zaptel.
On 1/3/06, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> If I have to move the outbounds to a ITSP I will, but Digium swears this is
> "supposed to work" so I am let
I have some stuck channels that I think I'm going to have to bounce
Asterisk to get rid of, but am curious to know what they are and how
they've managed to accumulate. The show up with a channel identifier of
'(None)' as in the output below, and do not show up in the soft hangup
list, and so c
In Asterisk v1.2.1 check the "featuremap" section of the "features.conf"
file. You also need to add the "w" or "W" option to your "Dial" cmd where
appropriate. So with the feature mapping below pressing *1 would start
recording.
[featuremap]
blindxfer => #1; Blind transfer, d
If I have to move the outbounds to a ITSP I will, but Digium swears this is
"supposed to work" so I am letting the work on the solution. If they finally
give up, at least there are alternative options.
-Kerry
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
Leo Ann Boon wrote:
Darrick Hartman wrote:
I'm attempting to use an asterisk box with a Digium TDM01B as
voicemail for an existing Meridian/Norstar PBX with an ATA-2
adapter. We're having problems where hangup is not always (but
sometimes) detected. It's not detected probably 70% of the tim
On 1/3/06, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> The magic setting is callprogress=yes, however, we have this working
> properly in the lab but not at this particular client location right now.
> Strange, but true.
> -Kerry
>
You're going to have very unpredictable results with that setting
Michael Stearne wrote:
I am having trouble with FC3.
After doing a yum update (of 1264 packages) I still cannont compile
1.2.1 from source:
make[1]: `libedit.a' is up to date.
make[1]: Leaving directory `/usr/src/asterisk-1.2.1/editline'
make[1]: Entering directory `/usr/src/asterisk-1.2.1/db1-
Look like this thread will become distro war ;)
Im using FC3 - * 1.2.1,
make sure #up2date b4 compile from tar.gz source
working fine...
Latest CentOS/Debian should be choice for production.
L
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I have used Fedora previously and have always had something not work correctly.
My understanding is that Fedora is the beta testing ground for new features
before they get put into RHEL.
Kind of like CVS head vs. Stable concept.
That is why they want you to pay for their RHEL.
That is the appeal
Is there a way to tell asterisk to start recording DURING a call?
Can I flash, then *XX, and then flash back to my call while asterisk
records?
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To UNSUBSCRIBE or up
That exactly what I had to do to get it working. Very very weird...
Seems like a bug in 1.3
Lee Archer wrote:
Actually it worked, but only after I defaulted all the settings on the
phone and let it pick the config up fresh.
Anyone know if there is any headset config options to default to
LOL I need to read the list completely too before I respond.
happy everything and merry too!
Tomislav Parcina wrote:
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED]
says...
Do I have to do something more? Does it work for anybody else?
Is there any other way to combine in and out soundfil
Wow, example by me. I don't read the Wiki enough lately ;)
while you are recording, are there *-in and *-out files being created?
Within seconds of hanging up, do they disappear to be replaced by a
single file? In my php I search for records using the following:
ls -1t --color=never --almos
I wrote a patch to do just this quite a while ago. Have been using it
in production since Asterisk 1.0.6. Here's the bug tracker link:
http://bugs.digium.com/view.php?id=4297
action_bridge-updated-10-12.txt is newer/better not written by me :)
http://bugs.digium.com/view.php?id=5841 is anot
On Tue, Jan 03, 2006 at 06:43:16PM -0500, Michael Stearne wrote:
> I am having trouble with FC3.
>
> After doing a yum update (of 1264 packages) I still cannont compile
> 1.2.1 from source:
>
> make[1]: `libedit.a' is up to date.
> make[1]: Leaving directory `/usr/src/asterisk-1.2.1/editline'
> m
I'm attempting to use an asterisk box with a Digium TDM01B as voicemail
for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're
having problems where hangup is not always (but sometimes) detected.
It's not detected probably 70% of the time or more. (The users transfer
callers to an
Look at the Manager command Redirect
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jason D. Wolfe
> Sent: Tuesday, January 03, 2006 7:12 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] bridging two active calls
>
> Can s
Don't forget ChanIsAvail()
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: Tuesday, January 03, 2006 3:43 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] How do you check whether a
> c
Thanks! I'll try that.
On 1/3/06, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote:
> Have you removing the asterisk include directory before trying version
> 1.2? I think it might be /usr/include/asterisk/ in many cases.
>
> Michael Stearne wrote:
> > I am having trouble with FC3.
> >
The magic setting is callprogress=yes, however, we have this working
properly in the lab but not at this particular client location right now.
Strange, but true.
-Kerry
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: S
Jason Wolfe
[EMAIL PROTECTED]
c (770) 561-6956
This e-mail transmission may contain information that is proprietary,
privileged and/or confidential and is intended exclusively for the person(s)
to whom it is addressed. Any use, copying, retention or disclosure by any
person other than the intend
I'm a bit confused on how you get your calls to Asterisk, what I mean is: are you phoning into asterisk via a sip user? in this case, which one?, if not is it iax or though a zap channel?anyway, here some tips:For your first problem it seems it has to do with what I pointed above, check that the us
Has anyone checked out the UNIDEN ELBT-595(http://www.uniden.com/elbt/index.html)It supposedly is a handset that can provide the same services. I have not seen any info about * compatibility though.
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Have you removing the asterisk include directory before trying version
1.2? I think it might be /usr/include/asterisk/ in many cases.
Michael Stearne wrote:
I am having trouble with FC3.
After doing a yum update (of 1264 packages) I still cannont compile
1.2.1 from source:
make[1]: `libedit.
Can someone give me some direction on how to bridge two calls on an asterisk
server. I'm originating two calls using asterisk java manager and after
some processing in each of the individual dialplans, I want to connect the
calls together.
Jason Wolfe
[EMAIL PROTECTED]
c (770) 561-6956
This e-ma
I'm not sure I follow - too much Christmas eggnog, perhaps...
My understanding is that one of the spans is always primary - Asterisk
will get its timing from that span unless no timing signal is present
there, when it will go to the secondary source etc.
Just to avoid any confusion, our c
I'm supposed to have a "mostly" canned script that will do this done
already. It will pull the list of people to call out of a db and play
them the file specified in the db table. Contact me offlist if you're
interested. It will be done real soon but I'm not done testing yet.
Darren Wiebe
[
Asterisk wrote:
In my case I would be using DNS round robin. So a UA would only be
registering to one * server at a time. So wouldn't in fact be an
active/passive?
No. You have said that you want the _other_ servers to be aware of that
phone's registration and be able to deliver calls to
I am having trouble with FC3.
After doing a yum update (of 1264 packages) I still cannont compile
1.2.1 from source:
make[1]: `libedit.a' is up to date.
make[1]: Leaving directory `/usr/src/asterisk-1.2.1/editline'
make[1]: Entering directory `/usr/src/asterisk-1.2.1/db1-ast'
make[1]: `libdb1.a'
Title: Dialer
You actually aren't far from it. If the system only needs
to play the same file to each person, a simple script can be used to pull from a
database and create call files. Asterisk will use the call files to place the
calls and play a sound. A few minutes of searching on that sho
you could try setting the * box to pull timing from each pri connected
to it and set the nortel to be a master for that circuit and see if that
helps any
Casey
Anthony Rodgers wrote:
Greetings, everyone, and Happy New Year!
I have a question relating to running two PRIs into a single TE411P.
Title: Dialer
Hello All,
I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear.
First off let me premise this with "no, this is absolutely not for doing call marketing".
I need to make my Asterisk box call a gr
Hi,
Hope someone can help me-Asterisk isnt behaving as I would expect
and I think its down to my contexts.
There are two things I cant fathom.
Firstly I want to record an IVR and so have created a user 20005 and
a context called createmenu. I am using SER in front of asterisk so I
changed the
On 1/3/06, Technical Support <[EMAIL PROTECTED]> wrote:
> We do a lot of installs on Fedora (slowly becoming our favorite). Initially
> clients asked for FC because of compatibility with Red Hat, great package
> management, etc. With FC4, you get a great set of packages, and not a lot
> of add-o
Greetings, everyone, and Happy New Year!
I have a question relating to running two PRIs into a single TE411P. We
have been experiencing echo, noisy MOH, poor audio call quality and so
forth that started at around the time we introduced the second PRI into
the equation. Here is our zaptel.conf:
Ariel Batista wrote:
Iaxtel has been down for some time now.
But to get in contact with digium via your asterisk box all you need is
to set this dialing rule up.
exten => 500,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ;Call Digium
exten => 500,2,Congestion
Cool, I didn't think of that
I've got a slight problem with echo. Basically, most of the outgoing
phone calls on our system echo, but as far as I can tell, the incoming
echo has been relatively fixed, with just a bit of work left to do on
it. I read somewhere that asterisk doesn't echo cancel on outgoing
calls, am I wron
>
> Large packets might be okay if your objective is simply to send maximum
> data with the lowest overhead. But, if there is any mix of traffic at all
> on the wireless facility, small and more frequent packets will provide
> better response to multiple sessions/applications/users.
>
> I have a cl
Title: Uvox streams
Have anyone been able to use the uvox streams found with Winamp 5.0?
I cannot seam to find a Linux player for it.
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> I'm proposing a wireless distribution system in infraestructure mode for a
> valley with no cables and distant houses, respect to Asterisk, I can get a
> company I work with to provide me with E1 and use LCR to get 4 different
> telcos based on prefix.
>
> I've been reading a very interesting
Thanks Steven,
that is a good way of achieving this.
Budoka.
_
Don't just search. Find. Check out the new MSN Search!
http://search.msn.click-url.com/go/onm00200636ave/direct/01/
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I have used and subscribed to them.
They do not allow the re-assigning of Callerid.. so you cannot do your own
callerid, like you can with other providers. all calls through them come
out with a callerid that they have set in their system. This is alright
for those that dont use callerid, thus int
BTW
Places that sell CellSockets that are know to work.
http://www.cyber-telecom.net/store/900/1800 GSM. No phone needed just SIM card.
http://www.cellantenna.com/Dockingstations/cellsocket.htmcellular phone accessory that allows you to dock your cellular phone and integrate it with your Land Li
>If the two servers service distinctly separate groups of endpoints,
they
>can share the same table since they won't care about the other server's
>entries. If the two servers service the same endpoints but in an
>active/passive arrangement, that would also work.
In my case I would be using DNS
Has anyone heard of or used sipbroker?
http://www.sipbroker.com/sipbroker/action/login
I was doing some brief looking around with them, and tried their PSTN
gateway, but was unable to get it to complete to calls with providers
such as at vonage (Where they claim I can peer).
I did an enum search f
The cyber-telecom is cheaper compared to the doc-n-talk unit.
http://cyber-telecom.net/store/product_info.php?cPath=1&products_id=29
however, they both work for 800/900/1800 bands.
Any products to work in the GSM 850mhz arena?
In our country, GSM runs 850mhz.
Brian: Do you need fancy features or
We cheated with ours.
One our ECS we did a "change dialplan parameters" and added
routing pattern 31 to the "ETA"
Once this was done we programmed routing pattern 31 to send
via the appropriate signaling/trunk group out to Asterisk. The end result
was that if the Avaya didnt know the ext
We do a lot of installs on Fedora (slowly becoming our favorite). Initially
clients asked for FC because of compatibility with Red Hat, great package
management, etc. With FC4, you get a great set of packages, and not a lot
of add-ons required.
Asterisk has perfect compatibility with FC3 & FC4
Just type in the asterisk command line: show channels or sip show channels type "help" also to take a look at the other commands availableAlyed How do you check whether a channel is active and the number of calls on it?Is it simple and complicated?/Obelix---
The unit dials whatever asterisk tells it to, although it seems to take a second or two for the mobile to start dialling.
It passes the Caller Name from the cellular phone's directory, together with the CLID on inbound calls.
On 1/3/06, Paul Dugas <[EMAIL PROTECTED]> wrote:
On Mon, 2006-01-02 at
I checked and the digium card was sharing with the onboard usb
controller. I disabled the USB in the BIOS and now it's on it's own
IRQ. The sound quality already sounds better. Thanks for the help!!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom V
Quoting Matt <[EMAIL PROTECTED]>:
I may not be using the right terminology. If I can make concurrent calls through
a particular provider, how can I tell the number of concurrent calls running on
that providers account. Perhaps trunk is the right word?
Is there a way to tell programmatically, thro
I don't think you mean how do you see how many calls are on a
channel.. 1 call per channel.
SIP:
sip show channels
IAX2:
iax2 show channels
ZAP:
zap show channels
To see details on a specific channel
sip show channel BLAH
iax2 show channel BLAH
zap show channel BLAH
On 1/3/06, Obelix <[EMAIL P
On Tue, Jan 03, 2006 at 04:33:49PM -, Brett, Gary wrote:
> Hi
>
> I wish to install asterisk 1.2 (the latest tar.gz from the site not the
> CVS version) on an HP box with a TE110P (single port E1/T1)
>
> My question is which OS would be preferred in this configuration Fedora Core
> 1 or F
On Tue, 3 Jan 2006, Steve Beaumont wrote:
> All,
>
> I seem to have a problem with Asterisk 1.2.1.
>
> Version 1.0.?? used to allow me to set the Type of Service bits to ef I.e.
> tos=0xb8 Diffserv EF (Expedited Forwarding) bits. This seems not to be working
> with 1.2.1:-
>
>
> Jan 3 19:26:2
it does support ilbc, alaw, ulaw and gsm. I've tryied all but get the same results with all of them the phone doesn't hangs up, but cannot hear anything in my endpoint. Alyed Return-Path: <[EMAIL PROTECTED]> Tue Jan 03 12:47:02 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164
Sounds like an IRQ sharing issue with the card. This comes up alot.
Do a lspci -vb and find the card and check if the IRQ is being shared
with another device in your system.
On 1/3/06, Jason Adams <[EMAIL PROTECTED]> wrote:
>
> Hello All,
>
> We are experiencing "heavy static" on our analog line
My SysAdmin tells me that this script is an Asterisk watchdog, not a SIP
watchdog. Surely someone else out there has had a provider that loses its
connection from time to time and needs to reconnect?
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Mess
All,
I seem to have a problem with Asterisk 1.2.1.
Version 1.0.?? used to allow me to set the Type of Service bits to ef
I.e. tos=0xb8 Diffserv EF (Expedited Forwarding) bits. This seems not to
be working with 1.2.1:-
Jan 3 19:26:26 VERBOSE[2702] logger.c: == Using TOS bits 0
Jan 3 19:
> > With the current *RT release?
>
> Yes. The crux of the issue that you can't have two servers responsible
> for updating the same records in the table, and that you can't have two
> servers both expected to react to changes in those records on an
> instantaneous basis (which is why you can't
For anyone interested, our company released a PHP/MySQL based content
manager for the Cisco 79XX series IP Phones compatible with the SIP load
yesterday.
It's available via: http://www.sourceforge.net/projects/open79xxdir
Best wishes,
-Corey
*
Th
> Since the device status system relies on it, I rewrote the
> incominglimit and outgoinglimit into the combined call-limit.
> The keywords "incominglimit" and "outgoinglimit" will be removed, but
> call-limit will stay.
>
> /O
Olle///
What happens when it not a simple phone/ATA but a providers
How do you check whether a channel is active and the number of calls on it?
Is it simple and complicated?
/Obelix
This message was sent using IMP, the Internet Messaging Program.
___
Tryed what Eric suggested in the other thread (changing in sip.conf: allow=all for disallow=all allow=somecodec) so now the call is not being hanged up, but cannot hear anything. Tryied it with ilbc,alaw, ulaw and gsm I still think it sould be a matter of RTP addressing since I get the fol
Eric "ManxPower" Wieling wrote:
Use a codec your phone supports like ulaw.
Assuming he is using SJphone, that I understand, would support iLBC even
in the free version ?
Alyed Tzompa wrote:
made the changes in sip.conf so now it reads:
disallow=all
allow ilbc
now I when the call is pl
Does anyone have exporience with the SetTransferCapability application?
I'm trying to use it, but it does not give the expected result.
My configuration is like this:
Telco<--->Definity<--->Asterisk<--->Brooktrout PRI card
The Definity communicates with the Asterisk using the Bearer 3.1K audio
s
Mike Fedyk wrote:
With the current *RT release?
Yes. The crux of the issue that you can't have two servers responsible
for updating the same records in the table, and that you can't have two
servers both expected to react to changes in those records on an
instantaneous basis (which is why y
SJphone supports ilbc, anyway tryed it with ulaw, alaw and gsm (all of them supported by SJphone), but the behaviour is the same. That's why I thought this sould be a RTP addressing stuff Alyed Return-Path: <[EMAIL PROTECTED]> Tue Jan 03 11:46:59 2006Received: from bourbon.fnords.org [209.16.72.1
On 1/3/06, Bogdan Moldovan <[EMAIL PROTECTED]> wrote:
> IMHO use FC4.
>
> Also after the install of the OS and all the required packages do a 'yum
> update'.
I am using FC3 right now with 1.0.9 and I am having a problem updating
to 1.2.1. I am trying to avoid upgrading to FC4 and I'll try a yum
u
Hi.
I'm proposing a wireless distribution system in infraestructure mode for a
valley with no cables and distant houses, respect to Asterisk, I can get a
company I work with to provide me with E1 and use LCR to get 4 different
telcos based on prefix.
I've been reading a very interesting paper
It should be possible, but I think that the problem lies in the connection between Avaya and Asterisk, let me explain: your Avaya most likely uses different connection parametrs for handling the calls coming from /going to PSTN and the ones coming from /going to Asterisk. If you need to pass let
Mikael Magnusson wrote:
Olle E Johansson wrote / skrev:
Andreas Koch wrote:
Hello,
how is it possible to connect (register) more the one Phone to One
Sip-Acoount.
With, for example sipgate.de this is not a special feature, it is
common.
We have users, what like to have more then one Phon
Eric "ManxPower" Wieling wrote:
Joseph Rothstein wrote:
I am setting up 10 SNOM 320s for a customer, and there seems to be a
problem
with call-limit and hints.
Here is my sip config for one phone:
[944]
type=friend
context=x
language=de
accountcode=x
notifyringing=yes
host=dynamic
dt
Is it possible to adjust the timings of
how long asterisk listens before it trys to match or is there something else
that I should be looking for? It acts as though asterisk is seeing a match in
the dial plan for the very first dtmf tone.
Thanks
Adam
From:
[EMAIL PROTECTED
Use a codec your phone supports like ulaw.
Alyed Tzompa wrote:
made the changes in sip.conf so now it reads:
disallow=all
allow ilbc
now I when the call is placed it is not hanged up, but I cannot hear
anything. I think it's becasue Asterisk is sending the RTP's to a wrong
address (my
inter
I use IP Kall to forward my missed cell phone calls to. This way, if my
phone is off, or out of a service area, calls will go to my * box.
Concurrently, all incoming calls to my * box cause it to dial my local
extensions at home, my extension at work, and my cell phone via NuFone.
Problem: A loop
Hello Armin,
On Mo, 2 Jan 2006 Armin Schindler wrote:
> I don't think it is necessary to exclude it. Just build chan_capi-cm and
> overwrite chan_capi.so as well as remove the app_capi* modules from your
> installation.
>
> Armin
Many thanks, it is working.
Karsten
__
We use the Dock-N-Talk at our office. We purchased a TDM with 4 FXOs,
the first three go to PSTN and the fourth into the Dock-N-Talk. Right
now I have the users choose to use the cell service by prefixing the
outgoing string with a certain digit; they can remember when it's
better to use the
Hello
All,
We are experiencing
"heavy static" on our analog lines when dialing into our asterisk server.
We have a Digium TDM04B with 4 FXO modules. Two of the
modules are connected to our analog lines from our local
telco.
This doesn't happen
with every call. It's definitely a rand
I can vouch for the CentOS line as being compatible with Asterisk 1.2
for all intents and purposes I've thrown at it. (Polycom 50x and 30x,
all SIP, asterisk has TDM with 4 FXOs), using g729 codecs but ulaw
sounded great. IM(h)O, they are great distributions. I would even
expect they'd be mor
I can use PBX from local LAN. I can also receive SIP calls thru the net.
Instead, i've got problem when trying to register the sip client and i'm
not onto the PBX LAN.
I get:
NOTICE: chan_sip.c:7708 handle_request: Registration from
'' failed for 'HOME_PUBLIC_IP'
This is the PBX location:
made the changes in sip.conf so now it reads: disallow=all allow ilbc now I when the call is placed it is not hanged up, but I cannot hear anything. I think it's becasue Asterisk is sending the RTP's to a wrong address (my internal IP). Looked at the sip debug and got the following: -- Execu
I'm currently using CentOS 4.2 in my home install on a P3-600/512MB/40GB
HDD with a X100P clone and it works great. Using Asterisk 1.2.1.
Ryan
> Any thoughts on CentOS-4.2?
> It is based on RHEL4 update2.
> It has the 2.6 Kernel.
>
> I am currently using CentOS-3.5, which is based on RHEL3 update
I generally use CentOS. Haven't tried CentOS 4 with Asterisk yet, but I'm
sure it'd work fine.
It's generally less of a "moving target" than Fedora is as far as updates
are concerned. CentOS 3.x will get updates as long as Red Hat is providing
them whereas FC1 servers and FC2 servers we set up a
Jason D. Wolfe a écrit :
Hello,
If I use an IAX termination service to connect outgoing VoIP calls to a PSTN
will I have answer supervision so that my script won't initiate too early?
I'm not sure to understand you. If you don't use Answer() before you use
Dial(), asterisk won't answer unti
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