http://www.voip-info.org/tiki-index.php?page=Asterisk%20Cisco%20CallManager%20Express%20Integration
dakemp wrote:
Hi,
We have Asterisk (CVS-HEAD-11/23/04-10:52:47) and Callmanager 4.1(2)
connected together using a SIP trunk, have both way dialing and are using
the Asterisk Box as a
Outstanding!
This completes the usability features of the scheduler. I have
a couple enhancements to make, such a CDR like facility to allow
examining past conferences to see who participated.
For the list members that have been following my app_cbmysql and
Web-MeetMe progress, look for an
The configuration for X-Lite in sip.conf:
[177209]
;Turn off silence suppression in X-Lite (Transmit Silence=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
;regexten=1234 ; When they register, create extension 1234
;username=xlite1
On Apr 7, 2005 8:36 PM, kaiser [EMAIL PROTECTED] wrote:
Hi , all:
Anyone try sip channel with canreinvite=yes?
sometimes we see a new INVITE will be send to UA immediately after user
hangup the call.
It makes the phone ring again after hangup.
Anyone know what happen?
It not always, maybe
Hello,
I am using CVS-NHEAD-03/29/05-15:51:16 and testing iax realtime. I have
configured a test account on iax.conf:
[test]
type=friend
context=test
username=test
auth=md5
secret=testing
host=dynamic
disallow=all
allow=ilbc
allow=gsm
callerid=1010
trunk=no
qualify=no
Then I insert an entry on
On Fri, 8 Apr 2005, Ronald Wiplinger wrote:
Is priority n already supported as next one???
I'm running CVS-HEAD. I don't know when it ('n') appeared.
Steve Edwards wrote:
On Thu, 7 Apr 2005, Jason Brown wrote:
Does anyone have a working failover outbound calls that I could sponge a
hint from? i.e.
Hello *users,
I would like to know how one would go about to allow every-one that
wishes to connect to my * machine to connect without a registration
being placed in the conf files. Would this be achieved through a
Database that will lookup the UserName and Password and if it does not
exist
Hi all * user
I have TDM FXO (4) connected with TELULAR (CELL Phone Device) and they
answer without ringing , and also when it goes to phone service provider
message like You Have dial wrong number please dial correct number...
without any ring and my cdr shows this call answered. Is there any
On Thu, 2005-04-07 at 20:23, Craig Guy wrote:
As an initial troubleshoot, can you preserve the original .tiff file from
rxfax and see if it is being received correctly or corrupted to determine if
the issue is in related to asteriks or somewhere downstream in the fax
processing to email part.
Hello *users,
I would like to know how one would go about to allow every-one that
wishes to connect to my * machine to connect without a registration
being placed in the conf files. Would this be achieved through a
Database that will lookup the UserName and Password and if it does not
exist
On Thu, 2005-04-07 at 17:50, Guy Decarpentrie wrote:
Le jeudi 7 Avril 2005 16:43, Chris Blake a écrit :
Greetings *`s,
I am trying to get faxes rec`d by * to be passed over to an email
address, and although the fax is being rec`d, it is not being
transmitted to the email address :
Hi all,
I currently have a setup where my users dial in to a dedicated DID that
sends them to VoiceMailMain(). this works fine except for the fact that
nobody can remember the number! (they already have to remember the main
number, their personal number, fax number and mobile number)
What I
Le vendredi 8 Avril 2005 09:04, Chris Blake a écrit :
On Thu, 2005-04-07 at 20:23, Craig Guy wrote:
As an initial troubleshoot, can you preserve the original .tiff file from
rxfax and see if it is being received correctly or corrupted to determine
if the issue is in related to asteriks or
I've googled and yet I've found nothing which describes this error.
This is Gentoo on Asterisk 1.0.7. Will try CVS later to see if it will
help resolve this error.
[res_features.so]Warning, flexible rate not heavily tested!
WARNING[25868]: loader.c:258
ast_load_resource:
Has anyone had any experience with the IAX2 phones being marketed by
Netweb?
I have received one and am waiting for a second one. There is an
extensive wiki page discussing the first phone which is now obsolete,
the 302. I'd agree with most of what is said there. However, for the
price these
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Josiah Bryan
Sent: Thursday, April 07, 2005 3:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Interception
There is no way to do
Hello Asterisk community,
After numerous request from various companies where
we have implemented * as a phone system and also
from many other * users all over the
world,yesterday wereleased the
1st version of Asterisk module for
Call Accounting Mate (www.callaccounting.ws) . As some of
Hi,
I'd like to setup delayed dial under Asterisk. That means that at the caller
side I set up number *YY and call Asterisk PBX (XXX... is number of
Asterisk PBX, * means pause (2 secs), YY is internal number).
Has anyone experience with receiving such calls ? How should I setup
hello
i am using phone with g723 and gw is complient for
g723.then why after 200 oK i am getting this.
can any one tell me why i am getting.
Apr 8 16:14:05 NOTICE[5750]: channel.c:1833
set_format: Unable to find a path from g723 to slin
Apr 8 16:14:05 WARNING[5750]: channel.c:2263
I have been using the PA168 IAX phone. It works well with *. I think
it is a good entry level type of phone.
cheers,
Wing
On Apr 8, 2005 3:47 PM, Wilson Pickett [EMAIL PROTECTED] wrote:
Has anyone had any experience with the IAX2 phones being marketed by
Netweb?
I have received one and
Have you tried the latest CVS, there was a bug relating to ALERTING which
was fixed yesterday...
-Original Message-
From: Ugur GUNCER [mailto:[EMAIL PROTECTED]
Sent: 08 April 2005 04:54
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Answering
Hi all,
I found that my Firefly Softphone is not able to register to
Asterisk.
However, if I define the following lines on
extensions.conf
[from-sip-external]
;appended by raymond 24 marexten =
_997402.,1,Dial,SIP/[EMAIL PROTECTED],trexten =
_997412.,1,Dial,SIP/[EMAIL PROTECTED],trexten
Hi,
I also define:
The same thing with context [sip] in extensions.conf but it doesn't works so
that why I cut-and-paste those lines:
exten = _997402.,1,Dial,SIP/[EMAIL PROTECTED],tr
exten = _997412.,1,Dial,SIP/[EMAIL PROTECTED],tr
exten = _997492.,1,Dial,SIP/[EMAIL PROTECTED],tr
from context
Gabriel Millerd wrote:
I have been struggling with oh323 compilation for some time now. I am
trying to use the voip-info suggested walk through that points to here
...
http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en
... which asks for versions OpenH323 (v1.13.5)
Hi,
I also 'spent' some times there banging my head on the wall.
Please read CAREFULLY :
http://www.inaccessnetworks.com/projects/asterisk-oh323
use only the mentioned version
the compilation linking seem to be rather sensitive
(for info, I use chs 29 march 05 their module 0.7.2)
read also
Hi,
is it possible to use Asterisk with T110P and CAS (channel associated
signalling)?
Thanks,
David
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Hello,
I am using oh323 and i think there is a bug. When i enter any digits,
There is a white space following the digits.
E.G. when i enter 333 oh323 responds 3 3 3
Because of that the DTMF does not get recognized.
Has anybody encountered and solved this problem?
Any hints will be greatly
On Thu, Apr 07, 2005 at 03:57:11PM -0400, William M. Sandiford wrote:
Hello:
I've read through the list archives and found tonnes of threads on this topic
but there has been no definitive answer, so hopefully someone can give me one.
Can a proper 2-way audio call be established when the
I believe it would work, besides that you may wish that each gateway may only
be used once at a time.
While the gateway may not bother if you use it twice, your accountant may, if
they send you a second invoice for another flat rate.
The soltuion would be to use setgroup / checkgroup to make
Cisco TAC service told me that they will not support RFC 2848/3265 for
the 7960 phones
So no busy status line notification with subscribe/notify system. This
is really a bad news for me.
So they are not planning to backport
Mike:
Depending upon your application requirements voicemail is pretty
simple. In sip.conf define a context for the peer from which inbound
sip connections will arrive. Add to the peer whatever config options
seem appropriate (allow=codec, context, etc,).
In extensions.conf define the same
You can use ChanIsAvail to confirm that specific
trunk is available before routing your call.
Wojtek
- Original Message -
From:
Jason Brown
To: asterisk-users@lists.digium.com
Sent: Thursday, April 07, 2005 9:59
PM
Subject: [Asterisk-Users] failover
outbound
Dinesh Nair wrote:
On 04/01/05 00:00 Matthew Boehm said the following:
Steve Underwood wrote:
And your EU bias is clearly demonstrated by this. I've never seen a
BRI product outside he EU. :-)
Come to Houston, TX. We were running a BRI for quite some time before
upgrading to a T1.
ahem, ISDN
Depends what you mean by in use. You will find BRI listed as a service
option in most countries. including China and the US. Installed lines
is different matter. They are so rare in most places that if you order
one it will be the technician's first install, and they will have enough
problems
Hi Robert,
I just set this up today for dialing international using a calling card
account.
usually we call 0120 982 433
wait for voice prompt
then dial the number
i set it up so the user only has to prefix with 011 then the number like
this:
[brastel]
exten = _011.,1,Dial(SIP/[EMAIL
Richard Dutton wrote:
Hi,
I've seen from the Asterisk Hardware list that the Dialogic D/300JCT-1E1 and
D/600JCT-2E1 cards are supported by Asterisk, can anyone tell me if the
D/300SC-1E1 and D/600SC-2E1 cards are as a client has quite a few of these
particular model and would like to use them in
David Hajek wrote:
Hi,
is it possible to use Asterisk with T110P and CAS (channel associated
signalling)?
There are hundreds of CAS protocols. Quite a few currently work with the
T110P.
Regards,
Steve
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Hi,
For madplay, install it, then put this into your musiconhold.conf
(adjusting the paths, of course):
[classes]
default =
custom:/usr/local/share/asterisk/mohmp3/,/usr/local/bin/madplay -Q -z
--fade-in --mono -R 8000 --output=raw:-
Subjectively, the quality is a little worse than with
Date: Fri, 08 Apr 2005 09:20:26 +0200
From: Chris Blake [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Fax to email problem
To: Guy Decarpentrie [EMAIL PROTECTED]
On Thu, 2005-04-07 at 17:50, Guy Decarpentrie wrote:
Le jeudi 7 Avril 2005 16:43, Chris Blake a écrit :
Greetings *`s,
On Fri, Apr 08, 2005 at 09:14:48AM +0200, Etienne Pretorius wrote:
Hello *users,
I would like to know how one would go about to allow every-one that
wishes to connect to my * machine to connect without a registration
being placed in the conf files. Would this be achieved through a
Date: Fri, 8 Apr 2005 16:21:03 +0900
From: Mick Hastings [EMAIL PROTECTED]
Subject: [Asterisk-Users] external access to voicemail?
Hi all,
I currently have a setup where my users dial in to a dedicated DID that
sends them to VoiceMailMain(). this works fine except for the fact that
nobody
SNIP
If you look at a 'iax2 debug' log you will see things like:
Tx-Frame Retry[000] -- OSeqno: 006 ISeqno: 007 Type: DTMF
Subclass: 6
Timestamp: 15832ms SCall: 2 DCall: 00167
[217.160.244.186:4569]
which seem to indicate the codes are making to my local asterisk
box,
or
The configuration for X-Lite in sip.conf:
[177209]
;Turn off silence suppression in X-Lite (Transmit Silence=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
;regexten=1234 ; When they register, create extension 1234
I have a STUN server running on my Asterisk box which seems to work
for most of my SIP clients.. but some of them seem to require NAT=yes
turned on. If I go further and turn QUALIFY=yes to on, is there a
reason I need to keep running a STUN server? If so, what's the
difference?
hello
Any one know how to resolve NAT issue.
PublicIp(UA)-Asterisk on
publicIP--privateIP(UA) its not working
PrivateIP(UA)-Asterisk on
publicIP--publicIP(UA) its working
how to reslove this issue
Thanks
Kamran
__
Do you
Hi ,
The linejack use the DSP compression for IAX2 ?
Think's .
kenshin .
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Hello,
I am using oh323 and i think there is a bug. When i enter any digits,
There is a white space following the digits.
E.G. when i enter 333 oh323 responds 3 3 3
Because of that the DTMF does not get recognized.
Has anybody encountered and solved this problem?
Any hints will be greatly
If you look at a 'iax2 debug' log you will see things like:
Tx-Frame Retry[000] -- OSeqno: 006 ISeqno: 007 Type: DTMF
Subclass: 6
Timestamp: 15832ms SCall: 2 DCall: 00167
[217.160.244.186:4569]
which seem to indicate the codes are making to my local asterisk
box,
More importantly it's a standardized DNS record to reliably locate any
service whether it's voip or whatever using weighting and
prioritization. :-)
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Friday, April 08, 2005 12:11 AM
To: Asterisk Users Mailing List -
I've never actually core dumped but I *have* been able to hang asterisk
a couple times, I believed my problem was when I lost my mysql
connection. Why it lost connection is a mystery, the servers are on the
same testswitch. :/
I forgot which head ver it was, a couple weeks ago.
-Original
Top posting for consistency...
I think we can stop this thread now. As Brandon pointed out, the dtmf
issue only effects a small percentage of their DID's, and the source
of that issue is outside their direct control (as is true with many
itsp's that obtain DIDs from third parties, which you are
I made patch
But when i wrote make im taking errors
.
./gentone ringtone 440 480
Wavelength 1 (in samples): 18.18182
Minimum samples (1): 200 (11.00.3 wavelengths)
Wavelength 1 (in samples): 16.7
Minimum samples (1): 50 (3.00.3 wavelengths)
Need 200 samples
Wrote
Hello,
It was written to manage asterisk in a postgres database, not MySQL. It
was written to add sip_users, sip_peers, dialplans etc. If you are still
interested, I will send you the php.
As I have written, it is for postgres, not MySQL.
Spencer
Marshall,
I am interested in seeing what
- Original Message -
From: Gavin Hamill [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, April 05, 2005 2:28 PM
Subject: Re: [Asterisk-Users] Asterisk and HylaFAX integration
On Tuesday 05 April 2005 14:13, Lee Howard wrote:
I successfully run a HylaFAX server
I was quoted about $700/month if I was within my downtown area for ISDN PRI.
So your price is in the right ball park.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of snacktime
Sent: Thursday, April 07, 2005 6:24 PM
To: Asterisk Users Mailing List -
Jean-Michel Hiver [EMAIL PROTECTED] writes:
Jean-Michel Hiver wrote:
Oops, sorry for the list reply :/
Actually, why does the Reply-To point to the Asterisk Users mailing
list? This breaks the reply to sender only / reply to all / list reply
functionality of my mailer. It's really broken
On Fri, 2005-04-08 at 02:57, San Singhania wrote:
Hello Asterisk community,
After numerous request from various companies where we have
implemented * as a phone system and also
from many other * users all over the world, yesterday we released the
1st version of Asterisk module for
Call
Spencer,
I am interested for your asterisk manager.
Can you send for me?
[]s
Douglas Conrad
G.Marshall escreveu:
Hello,
It was written to manage asterisk in a postgres database, not MySQL. It
was written to add sip_users, sip_peers, dialplans etc. If you are still
interested, I will send
Call XO www.xo.com
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of snacktime
Sent: Thursday, April 07, 2005 5:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Getting a good deal on a PRI
I have connected my home phone line into my asterisk box via
an X100P, but have noticed that asterisk doesnt check the line for
dialtone before dialing, barging in on any non-asterisk call which is taking
place.
I see from the voip-info.org wishlist that there is an
outstanding item to
Damon Estep [EMAIL PROTECTED] writes:
http://groups-beta.google.com/group/Asterisk-test
Stuff shows up fast! Anyone have insight on this, did I miss something?
Apparently, somebody created that group on google groups and subscribed
it to the * mailing list. As long as registered, anybody can
Hi
is the syntax of this call file
correct?
because wheniit to
/var/spool/asterisk/outgoing, the CLI shows"unknown keyword" for all the
keywords used (i.e. channel, MaxRetries,...).
1.call
Channel:Zap/g2/5148367580
MaxRetries:2 RetryTime:60
WaitTime:30Context:extensions
Follwing the information from the wiki
(http://www.voip-info.org/wiki-Asterisk+phone+snom) and the mailing list, I
have been able to get my Snom 190 to monitor extension states accurately.
I have noticed a couple oddities, however, that I am hoping I can get
explanation on so that I can know more
a couple other lists that I am on got notices last night that they were
added to google groups. I wonder if this is a google marketing ploy,
seek out all lists and subscribe them then spam the various lists
informing the individuals that instead of seeing it free in your email
box you can make
is there an easier way to ask through the manager api
what the connected channel is for a given channel.
Example: I dont know the session number for SIP/401
but I what to know what channel SIP/401 is connected to.
SIP/401 is presently something like SIP/401- type session number
and the
Looking at alternative VoIP providers and I found Teliax. One of the
features listed on their pay-as-you-go plan is unlimited
incoming/outgoing connections.
I am working on setting up a conference calling system for some of our
traveling salepeople to call into for their weekly staff meetings.
Yep, I've seen it and from reading http://www.voxilla.com it's a
pretty common problem.
If you turn on debugging what you'll see is that the Sipura has
mistakenly detected a DTMF code in the audio stream and is relaying it
by repeating the signal (very loudly I might add)
So this
So am I (sorry to drop in like this). I'm a programmer and I'm open to
start a project like this based on this attempt. Let me know.On Apr 8, 2005 9:46 AM, Douglas Conrad [EMAIL PROTECTED] wrote:Spencer,I am interested for your asterisk manager.Can you send for me?[]sDouglas ConradG.Marshall
Just documenting this issue and how I solved it for future reference on the
list, hope it helps someone:
I blew away my primary Asterisk install just because I felt it wasn't as
clean as it could be. I wanted to put on the latest AMP 1.0.007 (which, by
the way, totally rocks) and everything went
On Friday 08 April 2005 10:04 am, Jerry Geis wrote:
is there an easier way to ask through the manager api
what the connected channel is for a given channel.
Example: I dont know the session number for SIP/401
but I what to know what channel SIP/401 is connected to.
SIP/401 is presently
Hello All, I upgraded (installed) [EMAIL PROTECTED] .8 from .4. Now my
SPA-2000 will not stay registered. When the it needs to reregister it
may or may not. Line1 might be able too when Line2 can't and so on. When
on a call it will drop out. I did upgrade the SPA-2000.
Any Help would be great..
Looking at alternative VoIP providers and I found Teliax. One of the
features listed on their pay-as-you-go plan is unlimited
incoming/outgoing connections.
I am working on setting up a conference calling system for some of our
traveling salepeople to call into for their weekly staff
Spencer,
I am interested too for your asterisk manager.
Can you send for me?
Regards,
Fred OGUER
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Douglas
Conrad
Envoyé : vendredi 8 avril 2005 15:46
À : Asterisk Users Mailing List - Non-Commercial
trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] writes:
a couple other lists that I am on got notices last night that they were
added to google groups. I wonder if this is a google marketing ploy,
seek out all lists and subscribe them then spam the various lists
informing the
Hi Jonathon.
The boxes are at work, but I am pretty sure the FXO box (6port) is a
Micronet SP5050/S and the FXS box (2 port) is the Micronet SP5002/S.
http://www.micronet.com.tw
I recommend you move to the Digium users forum. I have taken this question
there. Not much feeback so far, but it is
The itsp I spoke with about concurrency limitations said they limited
due to overuse by calling card app providers.
By regulating the number of concurrent calls, they can maintain load and
quality for all users on the server(s).
Not being able to know your maximum line potential would be pretty
Right you are Michael.
I have some Multitech MVP200s and they do work indeed. Only problem is mine
are too old to do SIP. I know Asterisk does not do T.39 but as it only needs
to ALLOW the codec when devices are communicating with each other, it can't
be too hard to get working. Perhaps the
Malcolm Taylor wrote:
I have connected my home
phone line into my asterisk box via
an X100P, but have noticed that asterisk doesnt check the line for
dialtone before dialing, barging in on any non-asterisk call which is
taking
place.
I see from the
voip-info.org wishlist
Just wanted to let people know that the
VoicePulse Connect service has problems today.
Im personally experiencing dropped
calls within 2 seconds of an incoming phone call.
I talked to a tech who would not disclose
many details about the problem, saying their upstream provider is
Guillermo Salas M wrote:
On Fri, 2005-04-08 at 02:57, San Singhania wrote:
Hello Asterisk community,
After numerous request from various companies where we have
implemented * as a phone system and also
from many other * users all over the world, yesterday we released the
1st version of
Colin Anderson wrote:
So, AMP 1.10.007 from SourceForge seems to have this problem, anyone
upgrading won't run into this problem but a new install you will.
Just wondering, did you download AMP-1.10.007a bugfix release ? I have
installed it a few days ago and it went fine. (somewhere beginning
I have an installation next week. This asterisk box has a
PRI card (for the inbound PRI) and a TDM400P with 3 FXS cards in it (for 2 fax
machines and a credit card machine)
What do you have to do to get * to see the TDM400P? It sees
the PRI card and associated channels but I cant get
Jacob Cazzell wrote:
Looking at alternative VoIP providers and I found Teliax. One of the
features listed on their pay-as-you-go plan is unlimited
incoming/outgoing connections.
I am working on setting up a conference calling system for some of our
traveling salepeople to call into for their
no firefox no linux no asterisk .
Bye .
Jalal
Le vendredi 08 avril 2005 22:52 +0800, Ronald Wiplinger a crit :
Guillermo Salas M wrote:
On Fri, 2005-04-08 at 02:57, San Singhania wrote:
Hello Asterisk community,
After numerous request from various companies where we have
Bruno Hertz wrote:
<>Jean-Michel
Hiver [EMAIL PROTECTED] writes:
Jean-Michel Hiver wrote:
Oops, sorry for the list reply :/
<>Actually, why does the Reply-To point to
the Asterisk Users mailing
list? This breaks the reply to sender only / reply to all / list reply
functionality of my
Just wondering, did you download AMP-1.10.007a bugfix release ? I have
installed it a few days ago and it went fine. (somewhere beginning this
week I guess)
Cheers.
Kristof.
I didn't note which one it was, just clicked the topmost link on the
download page from SourceForge. I did take a
I should connect to a gateway and got following info:
Username = Password = NONE(not very secure!!!)
SIP
port 5060
IP address
For a trunk line dial 1234 and continue the number you want to reach at
PSTN.
codex g723 (I guess it should be g723.1)
vpbx*CLI
-- Executing NoOp(SIP/615-127a,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello, I installed asterisk (1.0.7) on my Gentoo (2.6.11-gentoo-r3) box
with udev support, also installed zaptel (1.0.7). I have a TDM31B
correctly installed. My problem comes right after I modprobe the card
and I execute 'ztcfg -vv', it gives me the
Jalal wrote:
no firefox no linux no asterisk .
You guys should check really before posting, this link works fine for me
in both Konqueror and Firefox, and additionally:
http://uptime.netcraft.com/up/graph/?host=www.callaccounting.ws
The site appears to be running Linux.
Who cares?
Bye .
A word of caution, we ran that same setup
for a while and then bagged the TDM400P in favor of 2 Sipura SPA2000 ATAs. The TDM400P
kept locking up and the SPA2000 never has. No problems getting fax from * to
the SPA2000 via g.711 over a FastE LAN.
I am not sure if the TDM400P has gotten
Hi,
I have a problem with the Asterisk server.
When I call from an IP Phone registered to the
Asterisk server, the connection is established and I can hear what the other
person says but this other person does not hear me. In fact, the Asterisk sends
an Invite message to the VoIP
What do you have to do to get * to see the TDM400P? It sees the PRI card
and associated channels
but I can't get the TDM400P to work - no matter what mix of channel numbers
I use ztcfg doesn't
like it.
My config with a Digium PRI card and a TDM400P, just finished yesterday
working fine:
Matt wrote:
I have a STUN server running on my Asterisk box which seems to work
for most of my SIP clients.. but some of them seem to require NAT=yes
turned on. If I go further and turn QUALIFY=yes to on, is there a
reason I need to keep running a STUN server? If so, what's the
difference?
I
Hello,
I am working for a charity in the UK and I am projecting a new phone system.
We would like to connect our two-wire telephones (40 or so) to an ADIT
600 channel bank, and connect that into an Asterisk box via the CMG card
or T1 card.
I have been in talks with Carrier Access about the
On Apr 8, 2005, at 6:28 AM, Steve Mann wrote:
I was quoted about $700/month if I was within my downtown area for
ISDN PRI.
So your price is in the right ball park.
XO quoted me $500 for a PRI in downtown Seattle about 6 months ago. I
suspect that you could beat that with a bit of shopping
Maybe following options:
1-) Get another channel bank from ebay at low cost. Which will also need another
T1 card;
2-) Use 40 voip phones at 50 USD each and you no longer need the card neither
the channel bank. But a reliable local network ;
Selon Peter Hoppe [EMAIL PROTECTED]:
Hello,
I am
On Friday 08 April 2005 16:35, Peter Hoppe wrote:
Hello,
I am working for a charity in the UK and I am projecting a new phone
system.
So - would there be any other way to connect 40+ telephones (two wire)
into an asterisk box? Are there any voip gateways that actually conform
to SIP
Call Accounting is such an important issue for me it is literally a make or
break component, without it I will not be able to deploy Asterisk at our
resort. If I have to use a windows computer to download and run the client
end of the software, so be it. At least the software will work and I will
On Fri, Apr 08, 2005 at 00:16:17 -0500,
Brian Capouch [EMAIL PROTECTED] wrote:
Jean-Michel Hiver wrote:
Jean-Michel Hiver wrote:
Oops, sorry for the list reply :/
Actually, why does the Reply-To point to the Asterisk Users mailing
list? This breaks the reply to sender only / reply to
What does it mean, and how can I fix it?
Apr 8 23:50:23 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown
SIP command 'PUBLISH' from '192.168.250.108'
Apr 8 23:50:24 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown
SIP command 'PUBLISH' from '192.168.250.108'
Apr 8 23:50:24
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