Anybody here added oh323 to @homeasterisk? I have compiled and add the
oh323. I am wondering if anybody able to add the oh323 under web interface
AMP? If anybody did it or know how to do it, please let me know. It has
option for sip, IAX.. why not add h323 !!
Thanks
On Fri, Apr 08, 2005 at 07:01:08PM -0500, Eric Rees wrote:
I have asterisk installed on a Dell 2850 dual-Xeon 3.0Ghz box with 2GB
of memory. This is serving about 75 sip clients, Polycom500's and
600's. We are running into problems with the memory. Asterisk, right
now, is using about 1.8GB
I have a DID from livevoip coming into * as SIP/gsm.My phone is a
sipura 2000 and the sip.conf for the sipura only allows g726. When I
dial the sipura on an incoming call to connect the channels the sipura
returns the error Media Type Not Available. If I set sip.conf to
allow ulaw for the
Hi Mat,
I can easily replicate the problem. I just put an entry on the iax
table for mysql, fire up iax soft client and BOOM .. asterisk core
dumps. What's weird is sip is working fine using realtime. Here is a
gdb backtrace. Not really a programmer. Hope someone helps. Thanks.
#0 0x00beeec0
snacktime wrote:
I have a DID from livevoip coming into * as SIP/gsm.My phone is a
sipura 2000 and the sip.conf for the sipura only allows g726. When I
dial the sipura on an incoming call to connect the channels the sipura
returns the error Media Type Not Available. If I set sip.conf to
On Apr 9, 2005 12:14 AM, Brian Capouch [EMAIL PROTECTED] wrote:
snacktime wrote:
I have a DID from livevoip coming into * as SIP/gsm.My phone is a
sipura 2000 and the sip.conf for the sipura only allows g726. When I
dial the sipura on an incoming call to connect the channels the sipura
I have installed Astcc and everything works fine.
Except one issue, right after the card number is
entered, Astcc prematurly sets the 'inuse' field in
the 'cards' table to 1 to indicate someone is using
the card. So if i entered the card number and i then
hangup without dialing any number,
Hi Henry
staff member can take the call). If there is another way to do this with
anaologue lines, i'm open to suggestions. I have looked at using a
You used to be able (still can probably) do this with a thing called
auxiliary working from BT on analogue lines. Two lines with one
number, the
In article [EMAIL PROTECTED],
BJ Weschke [EMAIL PROTECTED] wrote:
What version of * are you running? There was a bug that was posted a
few weeks back where when not using the q option it was possible for
legs of the conference to get further separated from each other
(sometimes up to 3-5 secs
Thanks for the replies.
I've got a bit further now after running the agi script manually,
turns out the asterisk
perl module was missing. I've got that and it gets somewhat further now.
However I am running RT 3.4.1, so I assume by what you say Kris its
probably not going
to work?
As soon as I
When I do see problems, It happens near capacity. All of the systems I
will be running will be using all 92 lines when they are being used.
Has anyone tested any of the new Dell systems in a production
environment with this kind of load? Specifically I am looking at the
1850
We are using spandsp but find it unusable in a
commercial environment, we are looking at changing to a dedicated hylafax server
using an eicon diva PRI/E1-30 via asterisk.We know the server on it's own
is a reliable configour only uncertainty is how good Asterisk is at
handling pass
Hi,
Get yourself a 'Fritz!Card PCI' - also marketed by BT themselves as a
'BT Speedway ISDN' adapter - these seem to be the most cheap and
supported of low-end ISDN2 adapters
Will do - they seem pretty inexpensive (even for the BT Speedway card is
only about £35). From doing a bit of poking,
Yes the digium cards are relatively cheap compared to traditional telephony
cards. A four port Eicon BRI card costs as much as the digium 4 port E1 so
on a per channel basis (8 vs 120) the digium is very reasonable. Must think
in terms of bang for buck before opening mouth next time.
As for the
This would be a good solution but be aware that at this time the Fritz! may
not handle DID (specifically PTP mode). The AVM drivers will not support
DID. The mISDN drivers and fritz! cards do seem to handle DID but chan_capi
doesn't pass the call to Asterisk (although you can see the call coming
Hello
I am in the process of putting together a short term calling card
solution that is rapidly deployable for charity events, and would
apreciate some guidence on hardware dimensioning for the solution
I have a test system running on an old P3 laptop, so in principle the
solution works : It is
Hi,
Thanks for the tip - is there a better ISDN card (i don't mind paying
extra) for compatibility with Asterisk? Is there any Digium hardware
that will do what i need to do? I'm basically looking for a really
reliable solution, with (relatively) easy setup and good compatibility,
and don't mind
If your hardware isn't getting clean data to spandsp, why should it be
able to get clean data to a hylafax box? Unless you fix the config
problem that stops spandsp working, there is no reason to expect a
pass-through to a modem bank and hylafax to work.
Regards,
Steve
Kevin Brennan wrote:
We
when a call comes the astcc-accountnum plays and ask
the caller about the card number and after playing
astcc-accountnum a period of time is given for the
caller to dial his card number but the problem here
is the short of the time given ,and i dont know where
and how can i setup the time.
Henry Owens wrote:
Hi,
Get yourself a 'Fritz!Card PCI' - also marketed by BT themselves as a
'BT Speedway ISDN' adapter - these seem to be the most cheap and
supported of low-end ISDN2 adapters
Will do - they seem pretty inexpensive (even for the BT Speedway card is
only about £35). From doing a
John,
Thanks very much for the detailed response, that sounds pretty much like
what i'm looking for (1x BT ISDN2e and 1x analogue). Are you using one
of the Digium 4 port BRI cards, or what hardware are you using?
It would be my intention to use the ISDN primarily for incoming, and
VoIP for
Serves you right for offering a bait and switch deal. If you are selling
unlimited that's what it should be. Why would you be surprised if someone
wants to use the unlimited feature?
What's wrong with selling a 1000 minutes for $10 plan? I guess you are
afraid someone will then offer an
I'm new to phone systems and phone wiring and I couldn't find an answer
to this question on the wiki or google.
My understanding is that a standard residential/business phone line
carries the signal over 2 wires. Your 4-wire RJ11 wiring supports 2
phone lines. Given that each line takes 2
Ok - point taken - but we're running Asterisk as a SIP/PSTN gateway and we
don't seem to have any other noticable problems, ok fax is more sensitive.
We've tried different versions of spandsp and it does not fix anything, ok
perhaps this shows problem is not spandsp - so where/how to start
Hi Remco,
-Original Message-
I'm using wengo for my outgoing calls (SIP). However,
whenever a number is
busy, asterisk plays a message that the number you dialed is not
available instead of a busy signal.
How can I get the 'normal' PSTN tones (like number not in use
tone or
Everyone,
I beg pardon to probably demand help of what had discussed many times,
earlier. But I really stuck and earlier replies couldn't help me out.
My first szenario connects two servers via IAX2. One is static IP the second
is a nated dnyamic host. I could register the dynamic host
One important question i ask my self is whether my asterisk server (it uses
nat, which in public uses a dns alias as well), needs to register itself
(with the register statement in iax.conf) at a host not behind a router?
Would this be mandatory in any case asterisk is behind a router, or can I
I am in the proscess of integrating a clients remote and head office
phone systems. Currenty each office has their own PBX and trunk lines. I
am recommending that they put in an Asterisk server at the Head office
with a WAN link to the remote office and switch to IP phones. Trunk
lines at the
I use following settings in shorewall:
(for connections established to the firewall)
ACCEPT netfwudp 4569,5060,1:2
(all outgoing connections are permitted)
Someone, please, comment on that to attest! I appreciate...
A.Fittering
--
Handyrechnung zu hoch? Tipp: SMS und MMS
Can you please detail the steps you have taken to successfully compile this
on @home asterisk?
Regards
Mike
- Original Message -
From: CM Rahman Jr. [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Saturday, April 09,
MemTotal: 2074808 kB
MemFree:417420 kB
Buffers: 39396 kB
Cached:1547124 kB
SwapCached: 0 kB
Active: 471180 kB
Inactive: 1131508 kB
HighTotal: 1179392 kB
HighFree: 233536 kB
LowTotal: 895416 kB
LowFree:183884 kB
SwapTotal:
Henry Owens wrote:
John,
Thanks very much for the detailed response, that sounds pretty much like
what i'm looking for (1x BT ISDN2e and 1x analogue). Are you using one
of the Digium 4 port BRI cards, or what hardware are you using?
I'm using an AVM Fritz card with chan_capi. They're pretty cheap
Eric Rees wrote:
MemTotal: 2074808 kB
MemFree:417420 kB
Buffers: 39396 kB
Cached:1547124 kB
SwapCached: 0 kB
Active: 471180 kB
That's a total memory usage for the entire OS of only 107MB:
(Total-Free)-Cached.
Tony
Dear all ...
I'm experiencing terrible trouble with crackling and noise on an
analogue line connected to an X100P (compatible) card. I've checked the
line with a normal analogue phone and it works fine, clear as a bell,
but any outgoing or incoming calls to Asterisk are almost completely
drowned
Rich Adamson wrote:
Serves you right for offering a bait and switch deal. If you are selling
unlimited that's what it should be. Why would you be surprised if someone
wants to use the unlimited feature?
What's wrong with selling a 1000 minutes for $10 plan? I guess you are
afraid someone will then
I had the same problem at one site. We could not receive faxes with spandsp
reliably. Our solution that seems to have worked with no problems so far
was to use a SPA-2000 to a fax machine.
- Original Message -
From: Kevin Brennan [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Folks,
Let's try trimming the replies. I'm sick of wading through 100 lines of
reply to find a single line comment.
Chris Mason
www.anguillaguide.com
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I am trying to put together a matrix. Please send me links, corrections,
additions, flames, etc.
http://www.geekgazette.com/index.php?option=com_contenttask=viewid=25Item
id=26
-Kerry
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It Stuart...Wonder if We're long lost cousins or something...Name here
is Bill Ford...
Anyway...It sounds like a mechanical problem. Maybe something as
simple as dirty contacts on the RJ-11 on the X100P. You say you've
checked the line...but have you replaced the cable from the demark to
the
How do I change the language when I do commands from the manager interface?
It seems that if I originate a call to a mailbox it will always speak
English. I have set the language to da in sip.conf general context, but it
still speaks English. I have no problems when using a phone, everything is
in
How can play music when is clients phone ringing in dial progress.
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I have a similar situation but it seems to vary from call to call
sometimes.
Using 2 digium genuine x100p's in a dell with riser card.
I'm wondering if it is something to do with the riser because it doesn't
seem to matter if I swap various cords, positions, etc.
Cheers,
Dean
Dial(SIP/100,30,tm)
On Apr 9, 2005 5:50 PM, Ugur GUNCER [EMAIL PROTECTED] wrote:
How can play music when is clients phone ringing in dial progress.
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Dean Collins wrote ...
Using 2 digium genuine x100p's in a dell with riser card.
I'm wondering if it is something to do with the riser because
it doesn't seem to matter if I swap various cords, positions, etc.
Right, that's interesting. My card too is in a Dell (2550) with a riser
card.
I have a very similar problem that
I have been grappling with for a while. I've got a genuine TDM400P
with four FXS ports and am using an Eicon Server quad BRI ISDN (using
CAPI) for external calls.
To date we have had no luck at all in diagnosing this problem as we too
have periodic problems
Thorben Jensen a écrit :
How do I change the language when I do commands from the manager interface?
It seems that if I originate a call to a mailbox it will always speak
English. I have set the language to da in sip.conf general context, but it
still speaks English. I have no problems when using
Ugur GUNCER wrote:
How can play music when is clients phone ringing in dial progress.
Usually you read the documentation.
At the Asterisk CLI do a show applications to show you what Asterisk
apps are available. Also see musiconhold.conf.sample in the Asterisk
source directory (in the configs
Drew Einhorn wrote:
The ATA generates it's own dialtone, and waits for
the user to dial a number, before sending anything
to the * box. So one of the first examples in the
in the Brief Introduction to Dialplans from
Vol. 1 of the Asterisk Documentation Project.
[incoming]
exten =
[marco-voicerec]
exten = s,1,noop(${ARG1})
exten = s,2,Background(custom/recordwarn)
A nice thought, to name macros for Mark, marco. Won't work in the
dialplan though.
Also, *8 is usually used for picking up a ringing phone. See features.conf.
___
My first szenario connects two servers via IAX2. One is static IP the second
is a nated dnyamic host. I could register the dynamic host succesfully at
the static one. Routing calls to it with my dialplan gets denied/rejected
due to missing authority on the remote side. I REALLY put this up
Forgot to mention - we are using
an IBM xSeries 206 Server, so the Dell riser card may not be the issue
if we are having the same problem.
FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz
Damian Funnell
Kerry Garrison wrote:
I am trying to put together a matrix. Please send me links, corrections,
additions, flames, etc.
http://www.geekgazette.com/index.php?option=com_contenttask=viewid=25Item
id=26
-Kerry
Kerry,
you did a great job, ... (I made a bookmark of it!!!)
However, I wanted to find
Oh my gosh! I've been staring so long at it that I didn't even see my
typo. I was not talking about *8.I am using the prefix of 8 instead of
9. Like 8401234.
Regards,
Chris
- Original Message -
From: Wilson Pickett [EMAIL PROTECTED]
To: Asterisk Users Mailing List
Hello,
have successfully installed Asterisk 1.o with H.323 driver and made
configuration:
GW (Hardware)- GnuGK - Asterisk
and i call into asterisk from the PSTN network and it's work fine, but i need to
make conversion from SIP small gateways to H.323. I need to make configuration
like that:
Kamran Ahmad wrote:
hello
how to pass G723.1 to other side is there any
softphone using g723.1. i want to use G723.1 in my
voice communication.
Microsoft Netmeeting can use G723.1
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Hi,
Try the OH323 implementation, we found it works better. Everyone has
different experiences oviously..
Cheers,
Sahil
On Sat, 9 Apr 2005, Adam Rybak wrote:
Hello,
have successfully installed Asterisk 1.o with H.323 driver and made
configuration:
GW (Hardware)- GnuGK - Asterisk
and i call
I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and
my * server.
My SPA is behind a NAT accessing a server which is also behind a NAT but SIP
and RTP ports are forwarded to it.
My SPA can successfully register. It can call another extension which is
inside the * local
Hi all, I am trying to set up two asterisk servers (SrvA and SrvB), and
what I want to get done is that if I dial 1X on SrvB the call must be
routed to extension X on SrvA and if I dial 2X on SrvA the call must be
routed to extension X on SrvB. I've read the www.voip-info.org wiki
abouta
I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can handle.
Anybody had his hands on this card or knows some details ?
regards
m.
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Message: 11
Date: Sat, 9 Apr 2005 08:21:16 -0700
From: Kerry Garrison [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] unlimited iax termination
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type:
izo wrote:
I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can handle.
Anybody had his hands on this card or knows some details ?
Please God, if you can hear me, don't let them use a TigerJet chipet.
--
I sent this earlier today. I didn't see my copy of the mail arrive back.
Does anyone know if I am supposed to get back any of my posts or is
there a setting I need to change.
If it has been reflected properly this morning, please accept my
applogies for the re-send.
David
--
Hello
I
Jim Sturtevant wrote:
I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and
my * server.
My SPA is behind a NAT accessing a server which is also behind a NAT but SIP
and RTP ports are forwarded to it.
My SPA can successfully register. It can call another extension which
Cytowanie Sahil Gupta [EMAIL PROTECTED]:
[...]
Hi,
Try the OH323 implementation, we found it works better. Everyone has
different experiences oviously..
Thanks, just compiled oh323 0.6.5. But still don't know how force asterisk to
act as protocol converter.
Regards, Adam
I am currently trying to solve this problem aswell with a TDM400p card and
going out the FXO port to the PSTN ..
If anyone runs into a solution, would be great news.
T
On Sat, 9 Apr 2005, Stuart Ford wrote:
Dear all ...
I'm experiencing terrible trouble with crackling and noise on an
Last night I signed up for a FWD account and was
hoping to use iax to connect thier server. I have been unable to connect as of
yet. I get a:
Registration of '64' rejected: Registration
Refused.
I used the iax section of http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWDto
Hi all,
My VoIP provider (race.com) doesn't send name info with CallerID, so I wrote
an AGI script that does the following:
1) If it's a toll free number (800|888|877|866), set the CallerID name to
TollFree Caller
2) Use curl to look up the number in Google phonebook
3) If a business listing,
How can I configure AgentLogin to connect the agent to a MeetMe
conference?
Or, can I achieve similar functionality through other means?
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867
|
| How do I change the language when I do commands from the manager
| interface?
| It seems that if I originate a call to a mailbox it will always speak
| English. I have set the language to da in sip.conf general context, but
| it
| still speaks English. I have no problems when using a phone,
On Sat, 9 Apr 2005 11:57:20 -0700, Scott Wolfe wrote
Last night I signed up for a FWD account and was
hoping to use iax to connect thier server. I have been unable to connect as of
yet. I get
a:
Registration of '64' rejected: Registration
Refused.
I used the iax section
like it says, the equivalent of 20 E1's or 28 T1's
and I guess you know how many channels a E1 or T1 PRI is
On Sat, 9 Apr 2005, izo wrote:
I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can handle.
Anybody
Thank you for your reply. There is a wealth of information on the wiki,
etc. I turned on RTP debug and the SPA is not sending it's public IP it is
sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere...
The SPA is behind a NAT and traversing the public IP network to get to
Hi,
On Apr 9, 2005 2:57 PM, Scott Wolfe [EMAIL PROTECTED] wrote:
I used the iax section of
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
to try and help me get this going.
I followed the directions below, and things are still working. You
must activate iax through fwd. Check
Jim Sturtevant wrote:
Thank you for your reply. There is a wealth of information on the wiki,
etc. I turned on RTP debug and the SPA is not sending it's public IP it is
sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere...
nat=yes makes Asterisk use the public IP that is
Thank you for your reply. There is a wealth of information on the
wiki, etc. I turned on RTP debug and the SPA is not sending it's
public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP
packets are going nowhere...
Do I understand your question correctly:
You have an SPA behind
I've helped a lot of people on the mailing lists and on IRC #asterisk.
and wanted to let people know that I will be in Europe between May 19
and June 21. Stockholm (VON 2005), Brussels (holiday/vacation),
Amsterdam (holiday/vacation), and Madrid (Astricon). There are
several weeks during my
Hi,
During boot I am getting an error that says the following:
Syntax error near unexpected token 'f'i'
/etc/rc3.d/S09zaptel line 92
Any ideas what might be causing this? I am using Fedora 3 with latest
Asterisk build
Thanks
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During boot I am getting an error that says the following:
Syntax error near unexpected token 'f'i'
/etc/rc3.d/S09zaptel line 92
Maybe you should look at line 92 in that file and see what's up with
it? Or post it here...
--Luki
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In your second option using a STUN server would I need to setup my own STUN
server?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Saturday, April 09, 2005 12:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial
In your second option using a STUN server would I need to setup my
own STUN server?
No, use FWD or xten's STUN servers.
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
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On Apr 8, 2005, at 9:40 PM, Drew Einhorn wrote:
...But how do we get the intial prompt to play
on an ATA?
On many ATAs you can have it do a hot-line dial -- start a call when
the phone is picked up. Perhaps you can have your ATA dial
@servername (no phone number, just the @ sign and the server
Pardon my answering myself (and for the long post). But I do have it
sort of working, and I come back with information on the GS HT-488, as
well as questions related to SIP / DTMF issues.
The GS HT-488 acts as a PSTN pass through device for 4 rings. If the
phone attached to the FXS port hasn't
I've seen this with @home.
Either trunk (under amp) and then dial(sip/trunk_name/extension)
or
Dial(IAX2/user_name:[EMAIL PROTECTED]/s)
James
On Fri, 18 Mar 2005 07:08:17 -0600, Matt Schulte [EMAIL PROTECTED]
wrote:
Per Mike's issue here, we're noticing this problem with older versions
of
On Sat, 9 Apr 2005, Stuart Ford wrote:
Dear all ...
I'm experiencing terrible trouble with crackling and noise on an
analogue line connected to an X100P (compatible) card. I've checked the
line with a normal analogue phone and it works fine, clear as a bell,
but any outgoing or incoming calls
I'm looking into the TelIAX pay-as-you-go plan. I'm assuming that they charge
incoming calls minutes as well? Is there the $0.02 connection fee for the
incoming call as well?
Thanks,
Jared
From: [EMAIL PROTECTED] on behalf of Mohit Muthanna
Sent: Fri
Yes and yes.
On Apr 9, 2005 6:03 PM, Bellows, Jared [EMAIL PROTECTED] wrote:
I'm looking into the TelIAX pay-as-you-go plan. I'm assuming that they
charge incoming calls minutes as well? Is there the $0.02 connection fee for
the incoming call as well?
Thanks,
Jared
Serves you right for offering a bait and switch deal. If you are selling
unlimited that's what it should be. Why would you be surprised if someone
wants to use the unlimited feature?
What's wrong with selling a 1000 minutes for $10 plan? I guess you are
afraid someone will then offer an
I am trying to put together a matrix. Please send me links, corrections,
additions, flames, etc.
http://www.geekgazette.com/index.php?option=com_contenttask=viewid=25Item
id=26
Go look at the list on digium's site, free world dialup's site,
the wiki, google, etc.
I'm experiencing terrible trouble with crackling and noise on an
analogue line connected to an X100P (compatible) card. I've checked the
line with a normal analogue phone and it works fine, clear as a bell,
but any outgoing or incoming calls to Asterisk are almost completely
drowned out by
Rich Adamson wrote:
Sure you can, in most cases. Just check the fine print in their
service agreements, or whatever else they publish. If its not
their, call them as a prospective customer. If they don't answer,
then why bother to do business with them as that's going to be
about the same level of
I've wondered about this as well. I suggest posting a bug to the bug tracker
and see if you can get a clarification or better yet, get someone to fix
this. It would be nice to override the clearing of the vars for Local
channels.
MATT---
-Original Message-
From: Dan Austin
On April 9, 2005 02:13 pm, Eric Wieling wrote:
izo wrote:
I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can
handle. Anybody had his hands on this card or knows some details ?
Please God, if you can
On Apr 9, 2005 5:03 PM, Bellows, Jared [EMAIL PROTECTED] wrote:
I'm looking into the TelIAX pay-as-you-go plan. I'm assuming that they
charge incoming calls minutes as well? Is there the $0.02 connection fee for
the incoming call as well?
That's the only thing they do that I could do
Brian McSpadden wrote:
On Apr 9, 2005 5:03 PM, Bellows, Jared [EMAIL PROTECTED] wrote:
I'm looking into the TelIAX pay-as-you-go plan. I'm assuming that they charge incoming calls minutes as well? Is there the $0.02 connection fee for the incoming call as well?
That's the only thing they do
Rich Adamson wrote ...
What country are you in, and does the chipset on the compat card
support the telco standards in your country?
I'm in the UK. The card was bought in the UK, but from Ebay, so I suppose it
could have originated from anywhere. The card dials and answers calls
without a
Andrew Kohlsmith wrote:
On April 9, 2005 02:13 pm, Eric Wieling wrote:
izo wrote:
I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can
handle. Anybody had his hands on this card or knows some details ?
Please
OK so now you have an IP address. Did you login and configure the
Sipura?
On Apr 7, 2005, at 1:04 AM, Rich Adamson wrote:
I wish to configure my Sipura with static
IP. I have set the static
IP, but there is registration failure on doing so. Could you please
tell me how
I'm aware of the context=menu feature in queue.conf.
This feature only works while the caller is waiting for an agent.
What I want to do is allow the caller to press * during the conversation
with the agent and exit the queue application without hanging up.
On Mon, 4 Apr 2005, Richard Lyman
On April 9, 2005 08:25 pm, Eric Wieling wrote:
Which specific Digium card does not use the TigerJet chip (as shown in
lspci)?
TE405P:
05:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev
01)
I imagine the TE410 and TE110 are both also similarly lspci'd.
-A.
I enjoy using the Adit 600 with the new FXS cards via the controller T1
interfaces. Works well. I do have concerns with using the CMG card via
MGCP. Has anyone done this? How is it working?
On Apr 8, 2005, at 12:50 PM, Matt Schulte wrote:
Word of warning, get the version 5 or higher FXS cards
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