On Wed, 2004-04-21 at 01:03, Fran Boon wrote:
On Tue, 2004-04-20 at 23:21, Mark Elkins wrote:
No matter what is dialled - I always go out on the 'Default' line.
Swapping order makes no difference. If I comment out the 'default' - it
does match the 'Cell' pattern - and works.
Should this actually attempt more than a single ping before claiming the
remote is unreachable?
ie, one packet (out of the two - one request + one reply) might be lost
or intermittent congestion might be involved.
Perhaps a config option for setting number of consecutive ping requests
are
Good day all
I'm still looking for a SIP client that will work on fedora core 1?
Thanks
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On Wed, Apr 21, 2004 at 09:20:54AM +0200, Altus Snyman spake thusly:
I'm still looking for a SIP client that will work on fedora core 1?
Thanks
linphone? www.linphone.org
--
Tracy Reed The attachment is a digital signature.
http://copilotconsulting.com More info:
Do you have a copy for me,the page seems to be closed and it redirect me
to http://swpat.ffii.org/ and I cant read that
Thanks
On Wed, 2004-04-21 at 09:16, Tracy R Reed wrote:
On Wed, Apr 21, 2004 at 09:20:54AM +0200, Altus Snyman spake thusly:
I'm still looking for a SIP client that will work
Hi,
I've been searching on an error I'm getting trying to compile against
uClibc, related to enum support. I found reference in an earlier thread
(http://lists.digium.com/pipermail/asterisk-users/2003-June/014176.html)
to a patch adding an Makefile option to remove enum support. Anyone
have
---BeginMessage---
---BeginMessage---
I have just installed the Alsa drivers
for my 2.4.18-14 kernel (RH8). I have configured
the sound card ok with alsaconf and tested
with the aplay , works fine. But when I run
asterisk it says..
---
[chan_alsa.so] = (ALSA
Hi,
I am new in asterisk and i've bought a X100p and a TDM400...
First of all, how can i verify my config files ?
Secondly, when i'm trying to pass a call to the outside, i ve a Notice
about appdial.c (l 554) telling me: unable to create channel of type Zap
...and i don't understand...
Finally,
I have been running af Asterisk server Version 0.7.2 for a while now
But I I wanted to upgrade my version to the new 0.9.0 or the CVS 1.0 Stable.
But when I install one of the new asterisk servers I having lots of troubles
with the IAX connection between my servers.
When I start the 0.7.2
On Wed, 21 Apr 2004, Jeremy Jones wrote:
I've been searching on an error I'm getting trying to compile against
uClibc, related to enum support. I found reference in an earlier thread
(http://lists.digium.com/pipermail/asterisk-users/2003-June/014176.html)
to a patch adding an Makefile
It doesn't look very hard. FreeBSD supports recursive mutexes. It is
just a matter of getting the appropriate defines. I'm going to look at
this.
Tom
On Tue, 20 Apr 2004, Olle E. Johansson wrote:
The recent addition of recursive mutexes to Asterisk is causing a lot of problems
on
On Wed, 21 Apr 2004, Tom wrote:
It doesn't look very hard. FreeBSD supports recursive mutexes. It is
just a matter of getting the appropriate defines. I'm going to look at
this.
On my Gentoo system I had to add #define _GNU_SOURCE to lock.h just
before it #includes pthread.h.
That
Thanks a lot,
Which ports U used ? I tried some ... the same error.
Only if I comment out the line it works
The other problem ist hat all natted IAX clients go Unmonitored (CLIIAX2
show peers) if I disable the Qualify=yes tag in IAX.conf.
If I activate qualify all go UNREACHABLE and cannot make
We want to inform the asterisk community members that we have released
the Appradius project under the GPL License and it is already available
for download.
It contains app_radius.so and cdr_radius.so applications.
The current version of Appradius project supports full RADIUS authorize
and
On Wed, 21 Apr 2004, Altus Snyman wrote:
Do you have a copy for me,the page seems to be closed and it redirect me
to http://swpat.ffii.org/ and I cant read that
You must have missed the link on that page... To enter linphone.org,
click here.
On the readability of swpat.ffii.org, there are
Sorry. 1004 hz...Im forgetting the 4 hertz...you're correct...must of been
a while since the function was used. When I called today to see if it was
still up, it connected, burped, and stayed up fine from then on. I'll bet
there were some transistors that hadn't seen electrons in a few
Szenario:
UA(Grandstream) = PROXY(SER) = GATEWAY(*) = PSTN
After sending the SIP ACK From Gateway (*)
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK86c0bd474ea746b5
From: Me sip:[EMAIL PROTECTED];tag=0f63d269bc25545d
To: sip:[EMAIL PROTECTED];tag=as05df60b5
Hello list,
I have many IP hardphone like Siemens 300 basic ( old ) , cisco ata..
etc I need: G711 from
old phones must be convert to G729 via asterisk and send to provider I have this
problem:
oh323 (last version): -
asterisk work with this driver ok for old phones, if I
IAX was removed on newer versions replace it with IAX2 just make sure to
change on your extensions.conf IAX/user:[EMAIL PROTECTED]/201 to
IAX2/user:[EMAIL PROTECTED]/201
Good luck
Miguel
On Wed, 2004-04-21 at 03:44, Jan Madsen wrote:
I have been running af Asterisk server Version 0.7.2 for a
Hi,
I have a couple of questions about MeetMe and call queues. Im
still pretty new to Asterisk, but already having to write a Service Center call
manager for it (which I might add, our director has agreed to make open
source!).
MeetMe:
How can I get MeetMe (does it even do this)
update your crappy hardware :)?? atleast with sip you will be able to
allow both codecs.
Miguel
On Mon, 2004-04-19 at 07:51, Serge wrote:
Hello list,
I have many IP hardphone like Siemens 300 basic ( old ) , cisco ata..
etc
I need: G711 from old phones must be convert to G729 via
Hello all,
Just to update,
Instruction's can be found at www.omniis.com/asttapi, including where to
download it from. This is update 0.02, this now includes a little
feedback from Asterisk so that when click to dial has occurred then it
is indicated at the start and the end of the call.
Now
There is no IE ( information element) in the isdn setup for this
indicator. Of course with ISUP(SS7) FGD trunks it is delivered in the OLI ISUP
parameter
On Tue, 20 Apr 2004, James Sharp wrote:
Quickie: Does anyone out there have experience with PRI delivery of ANI II
information?
Hi
My motto is to connect two computers on the same
network with Voip without using any special hardware,i
have downloaded Asterisk, I was suggested to use
LinPhone as a soft phone as it is very easy to install
I have installed Asterisk on my computer and iam using
it as a server.
And whe i
Hallo Altus Snyman
On Wed, 21 Apr 2004 09:54:42 +0200 you wrote:
Do you have a copy for me,the page seems to be closed and it redirect me
to http://swpat.ffii.org/ and I cant read that
Thanks
Try this:
http://www.linphone.org/linphone.php?lang=usrubrique=1
On Wed, 2004-04-21 at 09:16,
On Wed, 21 Apr 2004, Altus Snyman wrote:
Do you have a copy for me,the page seems to be closed and it redirect me
to http://swpat.ffii.org/ and I cant read that
http://www.linphone.org/linphone.php
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Scott,
We have 2 PRI spans on a TE405P, and we're sending faxes out 22 channels
concurrently out one span into the other. We were trying to stress our fax
application, but I fear we may have been stressing Asterisk (or the TE405P)
just a little too much as well.
Here's a grep for WARNING from
On Wed, 21 Apr 2004, Steven Kokinos wrote:
which is exactly 15 instances of asterisk. this is certainly a usual way of
running for many different applications, but i was not aware asterisk was
one of them. i would think there was something haywire going on, however, if
i start a single
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adam
Goryachev
Sent: Wednesday, April 21, 2004 2:29 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Repeated Notice: (UN/REACHABLE)
Should this actually attempt more than a single ping before claiming
I am having some difficulty getting a T100P card to work with my PRI.
When I attempt to make an outbound call via:
exten = 1004,1,Dial(Zap/g1/NPANXX)
I see the following on the asterisk console:
-- Executing Dial(SIP/sbruton-b8ce, Zap/g1/NPANXX) in new stack
Apr 21 08:18:48
Anybody out there use Ser along with *? Any advantages disadvantages? Is
this even a good idea?
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Hi,
Currently using voiptalk.org and the quality is getting
really bad.
I would like a second provider preferably in UK,
anyone got any suggestions?
Ta.
On Wed, Apr 21, 2004 at 03:24:06PM +0100, Craig Waddington wrote:
Currently using voiptalk.org and the quality is getting really bad.
I would like a second provider preferably in UK, anyone got any suggestions?
That's the trouble with running VoIP over contended public Internet.
Find someone
Sean Bruton wrote:
I am having some difficulty getting a T100P card to work with my PRI.
When I attempt to make an outbound call via:
exten = 1004,1,Dial(Zap/g1/NPANXX)
There is a real number here, ^^ right?
I see the following on the asterisk console:
-- Executing
Yes, but, I am talking about this world.
Ive got 2mb up/down with qos, just need another (good) provider.
If I can try a few and see which is best.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Kennedy
Sent: 21 April 2004 15:32
To: [EMAIL
Hi Darren-
In my situation, the frame rejects/retries appear not to cause problems -
even thousands of them, however I found that when I got a lot of unknown
error 500 messages, they would be associated with stuck channels, ie
channels that appeared to be in use when they are not. The stuck
On Wed, 2004-04-21 at 09:32, Steve Kennedy wrote:
That's the trouble with running VoIP over contended public Internet.
Find someone who can offer you connectivity with QoS and then has QoS
across their network for VoIP traffic.
LOL! I've not found any providers that offer QoS on their network
On Wed, Apr 21, 2004 at 08:37:12AM -0600, Michael Welter wrote:
Sean Bruton wrote:
I am having some difficulty getting a T100P card to work with my PRI.
When I attempt to make an outbound call via:
exten = 1004,1,Dial(Zap/g1/NPANXX)
There is a real number here, ^^ right?
On Wed, Apr 21, 2004 at 08:37:12AM -0600, Michael Welter wrote:
Sean Bruton wrote:
I am having some difficulty getting a T100P card to work with my PRI.
When I attempt to make an outbound call via:
exten = 1004,1,Dial(Zap/g1/NPANXX)
There is a real number here, ^^ right?
Craig,
2mb up/down with QoS doesn't mean anything, especially when you hit the
Internet. What is better is to look at the exact route of your calls and
then determine whether maybe there are some other issues. For instance,
we had a customer with Ciscos who was reporting choppy audio. However,
In the UK, with the sort of equipment that BT has in its network, you're
lucky to even get adsl going through! ISPs can only provide QoS up to a
certain boundary. After that it is out of their control!
Tan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Wait till DDOS/extortion scams start hitting voip providers!
Panny
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, April 21, 2004 3:43 PM
Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider
On Wed, 2004-04-21 at 09:32, Steve
On 21/04/04 08:37 -0500, Sean Bruton wrote:
I am having some difficulty getting a T100P card to work with my PRI.
When I attempt to make an outbound call via:
exten = 1004,1,Dial(Zap/g1/NPANXX)
I see the following on the asterisk console:
-- Executing Dial(SIP/sbruton-b8ce,
Steve Kennedy wrote:
On Wed, Apr 21, 2004 at 03:24:06PM +0100, Craig Waddington wrote:
Currently using voiptalk.org and the quality is getting really bad.
I would like a second provider preferably in UK, anyone got any suggestions?
That's the trouble with running VoIP over contended
On Wed, Apr 21, 2004 at 04:02:21PM +0100, [EMAIL PROTECTED] wrote:
In the UK, with the sort of equipment that BT has in its network, you're
lucky to even get adsl going through! ISPs can only provide QoS up to a
certain boundary. After that it is out of their control!
It depends on what
Haven't used them, but on my travels have come across:
http://www.magrathea-telecom.co.uk
Like I said, I don't know anything about them, but seem to remember that they are an
IAX provider.
Cheers
Matt
Wait till DDOS/extortion scams start hitting voip providers!
Panny
- Original
*CLI zap show channels
Chan Extension Context Language MusicOnHold
1default
... (2-22 are the same)
23default
On Wed, Apr 21, 2004 at 11:05:53AM -0500, Jason Stewart wrote:
On 21/04/04 08:37 -0500, Sean
Thanks Tan.
I will look into it my end. Unfortunately it isn't happening from just
one location, and a variety of phones. The quality used to be perfect,
the odd call would be a little jittery/choppy, but now most are like
that, I am running asterisk stable with eicon diva cards. 3.0Ghz dell
2GB
I'm getting the following when sending to a specific fax machine. Any
ideas?
File name is '/var/spool/asterisk/email2fax/7F2SOeYJiU.tif'
Changed from phase 0 to 2
Slow carrier up
Slow carrier down
Slow carrier up
NSF: 20 00 00 11 80 00 8a 49 10 53 54 49 52 4c 49 4e 47 20 43 4f 56
49 4e 47 54 00
Hahahhaaa your right there Tan.
List, don't get me wrong, voiptalk are very good, service, support,
price, I am just having some issues which may be my end.
I was just wanting to try some iax providers out to see what worked best
for us.
Hopefully will get sorted.
-Original Message-
On Wednesday 21 April 2004 16:12, Matt wrote:
Haven't used them, but on my travels have come across:
http://www.magrathea-telecom.co.uk
Like I said, I don't know anything about them, but seem to remember that
they are an IAX provider.
I haven't used Magrathea for anything 'production' yet,
I am having trouble locating webvmail on my * server.
Is this a seprate porgram or does it come with *. I
am running version
asterick*CLI show version
Asterisk CVS-03/26/04-17:08:20 built by
[EMAIL PROTECTED] on a i686 running Linux
asterick*CLI
Thanks
Kurt
make webvmail
from your source directory. Then, point your browser to:
http://your_ip/cgi-bin/vmail.cgi
Regards,
Gus
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Kurt
Enviado el: Miercoles, 21 de Abril de 2004 12:36 p.m.
Para: [EMAIL PROTECTED]
Asunto:
That's the trouble with running VoIP over contended public Internet.
Find someone who can offer you connectivity with QoS and then has QoS
across their network for VoIP traffic.
LOL! I've not found any providers that offer QoS on their network other
than a small regional ISP that put
I did a quick test with the danish numbers in say.c patch (04-20-04 02:11)
and found this..
*1 -- Executing SayNumber(SIP/1000-497f, 1) in new stack
-- Playing 'digits/1' (language 'da')
*2 -- Executing SayNumber(SIP/1000-497f, 100) in new stack
-- Playing 'digits/1' (language 'da')
Hi Eric !
I have the same problem with Canon fax mashine as you have. I have wrote an
email to Steve (developer of spandsp) some weeks ago and got no answer how
to fix the problem :(
Looks like connection was dropped by fax mashine without any reason
- Original Message -
From: Eric
Dear Scott
i have notice the same type of warning plus another with my TE410P with not
very high load 30 IN/OUT line, i use only two span in this moment.
What is only warning? or not?
Thanks in advance
Dimitri
Apr 21 10:27:35 WARNING[966674]: Unable to forward
Next question:
After doing your rerecommendation was able to get
to the main web page. I trtriedogging in using one of
the vmvmailccounts (I am to assume that the login and
password is what I have set up in vovoicemailoconfor
mail boxes) and I got login incorrect. Do i need to
change
permission
I am currently helping a friend build an Asterisk PBX that spans
several cities using anything from T1s to DSL connections to
link remote SIP phones, IAX gateways, etc. to a central Asterisk
PBX server that serves up voicemail, features, etc. The biggest problem
that I have had with this system
Has anyone got Asterisk talking successfully to an Altigen PBX using h323?
I can successfully make calls from Asterisk to Altigen, but calls from
Altigen to Asterisk get a fast busy.
There seems to be a lack of h323 example files (or maybe I'm looking in the
wrong places) as well as a severe lack
On Wed, 21 Apr 2004, Clif Jones wrote:
I am currently helping a friend build an Asterisk PBX that spans
several cities using anything from T1s to DSL connections to
link remote SIP phones, IAX gateways, etc. to a central Asterisk
PBX server that serves up voicemail, features, etc. The
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adam
Goryachev
Sent: Wednesday, April 21, 2004 2:29 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Repeated Notice: (UN/REACHABLE)
Should this actually attempt more than a single ping before
Sören!
Tusen Tack :-)
I'll add your input and will see what I can do to fix this. Does the other danish
people agree?
For the rest of you - please add your input to the bugtracker. For those
of you who have earlier contributed with patches, answer my e-mails!
If I don't get disclaimers from the
Any good ideas would be appreciated!
We use a package called Nagios to monitor our servers, which works quite
well. It has the ability to track service and host dependencies so you don't
get flooded with a bunch of service down alerts when the real cause is a
bad switch (or similar).
It would
No, you don't need to change permissions. Check in your voicemail.conf the
user password for accounts.
I don't know how vmail.cgi works with multiple contexts, or if you have
mysql/pgsql support with app_voicemail.
See http://www.voip-info.org/wiki-Asterisk+gui+vmail.cgi for more details.
Dimitri-
I'm not sure about the message Unable to forward voice - I don't get that
one on my systems.
The frame reject messages are the same ones I was talking about - these
are load related, I'm pretty sure.
Regards,
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto
Almost all of our fax machines are Canon, so it's kinda tough to test
TxFax.
On Wed, 2004-04-21 at 11:35, Serge Oleinikov wrote:
Hi Eric !
I have the same problem with Canon fax mashine as you have. I have wrote an
email to Steve (developer of spandsp) some weeks ago and got no answer how
to
On Tue, 2004-04-20 at 18:07, [EMAIL PROTECTED] wrote:
worked came to one ring only now. Thank you very much. If I use TE410 or
TE405 instead of X100P. do it make that first ring disappear?
Excuse me if already answered, but broken threads means I don't know
till I am done reading all my mail if
On Wed, 2004-04-21 at 18:40, CW_ASN wrote:
No, you don't need to change permissions. Check in your voicemail.conf the
user password for accounts.
I don't know how vmail.cgi works with multiple contexts, or if you have
mysql/pgsql support with app_voicemail.
You need to log in with [EMAIL
I'm interested in hearing success stories in tying things like Asterisk
YELLOW
and RED alarms and network problems into a central alarm reporting solution.
The most common problems that I have found are:
1. Someone unplugs a X100P from the Dmarc and nobody knows until people
complain
Hello,
In my sip.conf I have:
;Register and forward FWD numbers to internal extensions
register = FWDNUMBER:[EMAIL PROTECTED]/9500
Which should register Asterisk at FWD and then when any calls are made to
FWDNUMBER those calls should be forwarded to extension 9500 as specified in
the
Hi,
Apologies in advance for the lengthy email.
I'm new to asterisk and have trouble with isdn4linux.
The setup is very basic like this:
winxp --- asterisk winxp
x-lite | x-lite
|
pstn
The hardware involved is:
OK,
I've fixed the '#' transfer problem. We setup a macro for dialing staff
extensions, however, it was missing the 'tr' options on the Dial
application:
[macro-staff-extension]
; Macro for Staff Extensions
exten = s,1,Dial(${ARG2},20,tr) --
exten = s,2,Voicemail(u${ARG1})
exten =
Ian McLaughlin wrote:
Has anyone got Asterisk talking successfully to an Altigen PBX using h323?
I can successfully make calls from Asterisk to Altigen, but calls from
Altigen to Asterisk get a fast busy.
There seems to be a lack of h323 example files (or maybe I'm looking in the
wrong places) as
Dear Scott
the reject warning is a bug? I must put in bug track?
Thanks in advance
Dimitri
On Wednesday 21 April 2004 07:44 pm, Scott Stingel wrote:
Dimitri-
I'm not sure about the message Unable to forward voice - I don't get that
one on my systems.
The frame reject messages are the
Dear
i have buy two license of G729 codec and i have install/registered as
documented but after i start Asterisk -vvvcng i notice this warning and if
i made call the CLI say No compatible codec! How can i solve this problem?
Thanks in advance
Dimitri
It would be VERY nice if TxFax exited with a non-zero return code if the
fax was not actually sent. In the case of the problem with sending to
the Canon fax machine, this always returns 0:
$status = $AGI-exec(TxFAX, $fax_filename|caller);
On Wed, 2004-04-21 at 12:46, Eric Wieling wrote:
Almost
We are using ser together with *. Ser is used as a SIP proxy/registrar, * is
used as a sip - pstn gateway and voicemail/forward/conference server.
Advanteges - scalable, very large number of sip clients with easier
radius/database user management, advanced sip logic/routing options, better
sip
Hello,
I'm considering using Asterisk with some type of Cisco phone, and currently
considering either the 7940 or 7960 because of its more-complete functionality
(compared to the 7905).
I'm currently wondering:
Do all the expected functions (transfer, conference, voice mail, message
Yeah, primarily fired at them from large telcos with infinite bandwidth...
At 11:01 4/21/2004, you wrote:
Wait till DDOS/extortion scams start hitting voip providers!
Panny
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I am using a CallManager Server at
work and all my phones register with it. I have built an Asterisk server and am
connecting to the CCM with OH323. This is working great and am going to get more
features working as time permits. I have calls going both ways between these
servers. The
On Apr 21, 2004, at 12:20 PM, David Carter wrote:
Hello,
I'm considering using Asterisk with some type of Cisco phone, and
currently
considering either the 7940 or 7960 because of its more-complete
functionality
(compared to the 7905).
I'm currently wondering:
Do all the expected functions
Thanks, those are the advantages I needed to hear. Is there any special
config I need to do to either * or SER? Do I just set SER as a friend in
sip.conf? Still looking for documentation on using the two together.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Dear Scott
the reject warning is a bug? I must put in bug track?
Thanks in advance
Dimitri
No, not a bug I don't think. A warning that the framer driver was not able
to keep up with the PRI bit stream.
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and
On Wed, 2004-04-21 at 18:41, Ernest W. Lessenger wrote:
We use a package called Nagios to monitor our servers, which works quite
well. It has the ability to track service and host dependencies so you don't
get flooded with a bunch of service down alerts when the real cause is a
bad switch (or
Hello,
From: Dawid Mielnik [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Ser and Asterisk together
Date: Wed, 21 Apr 2004 21:15:21 +0200
We are using ser together with *. Ser is used as a SIP proxy/registrar, *
is
used as a sip - pstn gateway and voicemail/forward/conference server.
Advanteges
On Wed, 2004-04-21 at 21:02, AJ Grinnell wrote:
Thanks, those are the advantages I needed to hear.
FWD SipGate apparently have this config:
http://www.voip-info.org/wiki-Asterisk+at+large
Is there any special
config I need to do to either * or SER? Do I just set SER as a friend in
sip.conf?
On Wed, Apr 21, 2004 at 01:17:23PM -0700, Scott Stingel wrote:
Dear Scott
the reject warning is a bug? I must put in bug track?
Thanks in advance
Dimitri
No, not a bug I don't think. A warning that the framer driver was not able
to keep up with the PRI bit stream.
Zaptel
On Wed, Apr 21, 2004 at 04:18:51PM +0100, Craig Waddington wrote:
1 1 ms 2 ms 1 ms 10.5.0.1
217 ms14 ms14 ms 195.10.119.94
317 ms14 ms14 ms 195.10.119.158
422 ms14 ms15 ms 217.23.160.1
515 ms15 ms31 ms 217.23.162.122
As far as I understand how voicemail is integrated into Asterisk, it
seems that SIP channels poll MWI directly from the filesystem.
Is it possible (feasible?) to have something like :
- a central voicemail server
- which has an IAX peer with a mailbox= line with tens of VM boxes
- this peer has
Im totally a newbee at *
Im confused.
Ive got a FWD account, and it works on the winboxen. Ive got * up and can
do the echotest etc, so its working.
I want to get FWD working, and all the pages ive seen on setup are most
confusing.
Is FWD setup like IAXTEL? Do i plug in my FWD info in the same
I have some Grandstream BT101 SIP phones. Work great (so far).
I have some Planet VIP-101T H323 phones... how do I make them
look/feel/act like a SIP phone
I can dial to them from both Trunk + SIP's
(ie - I've added 'oh323' libraries)
What config do I add so that if I dial the * IP -
Hi all,
I have 3 * boxes all running the same OS and software version. Machine A
has an X100P card, machines B and C do not. They all have the same
dialplan.
I can dial from machine A to either of the other 2 with no problem. I can
dial from either of the other 2 to machine A with no problem. I
On Wed, 2004-04-21 at 17:31, Mark Phillips wrote:
Hi all,
I have 3 * boxes all running the same OS and software version. Machine A
has an X100P card, machines B and C do not. They all have the same
dialplan.
I can dial from machine A to either of the other 2 with no problem. I can
dial
Look here:
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: tmpm [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, April 21, 2004 11:50 AM
Subject: [Asterisk-Users] Ok, Im confused
Im totally a
We have set up an Asterisk PBX managing a EuroPRI in Italy.
We have conneccted to the asterisk PBX some Cisco IP Phones and a Panasonic
PBX with 10 analogic phones.
If we dial an unassigned telephone number we are not able to listen to
PSTN
Operator message telling that the subscriber does not
Quickie: Does anyone out there have experience with PRI
delivery of ANI II information?
Our carrier appends it to the DNIS. For instance, if I
call from my cellphone, we get:
877852000263
Where 8778520002 is the dialed number, and 63 are the info
digits.
So your carrier provides you
Thanks Jim,
But that page started my trip off to confusionbeen theretried it 10
different ways...still no joy.
I'll go through it once again, maybe Im missing something, I dont know. Im
about ready to boot the penguin to the curb...
I know its in there...I think Ive got it all
At 9:16 PM +0100 on 4/21/04, Fran Boon wrote:
On Wed, 2004-04-21 at 18:41, Ernest W. Lessenger wrote:
We use a package called Nagios to monitor our servers, which works quite
well. It has the ability to track service and host dependencies so you don't
get flooded with a bunch of service down
Quickie: Does anyone out there have experience with PRI
delivery of ANI II information?
There is no IE ( information element) in the isdn setup for
this indicator. Of course with ISUP(SS7) FGD trunks it is
delivered in the OLI ISUP parameter
Of course! ;-) But we're not quite there with
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