Hi,
I am very new to asterisk and appreciate somebody to help me at the following..
I have sucessfuly install [EMAIL PROTECTED] at my linux box and current
running 3 Xlite clients on my LAN without problem (behind firewall and
router). Is there any possible that I can connect to another Asterisk
--- Ian Pattison [EMAIL PROTECTED] wrote:
If I understand your question correctly ztcgf is not
a module, it's merely a rudimentary diagnostic
utility. Run ztcfg -vv to get info on your zaptel
hardware.
Thanks Ian:
When I installed the X100P, I followed the
installation instruction and it
Hello friends,
iam getting problems in transfering the call from one
agent to another agent
iam useing the asterisk-1.0.7 and sj phone
i have one queue called 30 for that one
i have two agents 9020 anb 9024 and normal memeber
1001
so when 1001 calls to the queue 30 it goes and connect
to
Hi,
Il giorno ven, 15-04-2005 alle 18:33 -0400, Ian Pattison ha scritto:
If I understand your question correctly ztcgf is not a module, it's merely a
rudimentary diagnostic utility. Run ztcfg -vv to get info on your zaptel
hardware.
Not exactly. ztcfg is the tool that applies the
Gavin Hamill wrote:
On Friday 15 April 2005 22:50, Olivier MONNET wrote:
Hello,
It can be that you need power on your ISDN bus.
[..]
I have just added an ISDN NT to power the bus:
where does one go to find broken NT1 boxes?! We have three ISDN2e NT1s but
they are all active and working, I don't
In the UK BT do not provide NT1 seperately and the BT ISDN2e grey boxes
only provide emergency power as they do not have a mains connection.
So I do not believe power is the issue...I strongly suspect you need the
100 ohm terminators and you need to check that your ISDN BRI crossover
cable is
Hello All,
I was wondering if anyone has a solution to the following:
I have a dialplan to route to an extension, but if the extension is busy
I want the user to get a message, and hold. I want them to be able to
press 1 to leave a message if they do not wish to hold.
If the extension rings but
Slightly OT but if you know much about networks, please read and maybe you can suggest some things.
I have been running asterisk for about 1 year. Currently running 1.0.6
with 2 X100P and 1 TDM400P with 3 FXS modules on a Pentium III-800 with
512M RAM. The system performs very well for both POTS
On Fri, Apr 15, 2005 at 11:48:29PM -0700, chawki hammoud wrote:
--- Ian Pattison [EMAIL PROTECTED] wrote:
If I understand your question correctly ztcgf is not
a module, it's merely a rudimentary diagnostic
utility. Run ztcfg -vv to get info on your zaptel
hardware.
Thanks Ian:
Hi Robert,
extensions.conf grammar is very easy:
To change 602085551212 to really dialing 6011442085551212:
exten = _602.,n,Dial,IAX2/user:[EMAIL PROTECTED]/6011442${EXTEN:3}
Everything that starts with 602 - take out the first 3 digits, and add
instead 6011442 before the number.
For more info
Title: FXO GW Dial in/out syntax
Hi all,
I have a non-branded FXO Gateway connected to 4 analog lines at the office. The situation is that I figured out how to make it dial in with the following entries:
In sip.conf:
[4003]
username=4003
fromuser=4003
dtmfmode=rfc2833
type=friend
Are all your ISP using the same upstream provider?
Wilson Pickett wrote:
Slightly OT but if you know much about networks, please read and maybe
you can suggest some things.
I have been running asterisk for about 1 year. Currently running 1.0.6
with 2 X100P and 1 TDM400P with 3 FXS modules on a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi list!
I'm just wondering if anyone has any experience with [EMAIL PROTECTED]
connecting to an ISDN BRI line, and if so, could you tell me what
hardware will work well with [EMAIL PROTECTED]
I've been looking at the Fritz!PCI but it seems (though
Hi,
Thanks for helping me out.
I want to clear out few more points
1) zaptel cards receive PCM from PSTN. In what form do they give it to
asterisk. Do Zaptel cards CODE/DECODE PCM from PSTN to RTP or zaptel cards
forward PCM to asterisk which converts it to RTP.
2) If asterisk does that
Newbie Question
Has anybody installed [EMAIL PROTECTED] on VMware Workstation (w/ WMware
Tools)successfully?
Thanks,
Sean
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Hello everyone:
I've had X100P running on asterisk 1.0.6 for about two
months. Each time I start linux, I manually modprobe
zaptel, wcfxo, and ztcfg before I start asterisk.
Today, all of the sudden I am not able to load ztcfg
and I get the error message
FATAL: Module ztcfg not
On Saturday 16 April 2005 09:33, Tim Robinson wrote:
So I do not believe power is the issue...I strongly suspect you need the
100 ohm terminators and you need to check that your ISDN BRI crossover
cable is wired correctly. i.e. the RX pins on one RJ45 connect to the
TX pins of the other.
All,
Further to my note below, I now have incoming working - yipee! (and seem
to have identified a problem with the G711A codec in the latest sipura
firmware - although need to do some checking). This box sounds great
compared to the echo ridden FXO and gives me an FXS for very little more
cash.
Julien Levi wrote:
I'm beginning to think a chroot environment may be the best route - *
fails to compile as it loads the wrong headers from libpri and zaptel
(those from the may 2004 install, the new ones get put in
/usr/local/new/usr/include).
I got it to compile by editing the makefile
Frtiz is a nightmare although it is cheap and I have seen it
working. I have been trying to install it for some days without success but one
thing is for sure: you have to use the right Kernel (they are available for
2.4.20 and 2.6something). There are no clear instructions to install it,
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
Note that this may have strange results if you need
to load more than
one module, as ztcfg will fail after the load of the
first module and
modprobe will return a failure message.
Yes, earlier when i first installed X100P, I got the
error message
On Fri, 15 Apr 2005, Paul Hewlett wrote:
I am running * 1.0.6 with 8 analogue phone lines connected to 2 cards - lspci
reveals these as :
03:04.0 Communication controller: Tiger Jet Network Inc. Model 300 128k
03:05.0 Communication controller: Tiger Jet Network Inc. Model 300 128k
The
I tried many different possible ways to us speed dialing, however, I
end up in the default context, where the number does not match anything,
... with the result Playing 'demo-congrats'
I also could not figure out how to use the tabs Queues and Agents
I have not found a new version over
Hi everyone:
I have my asterisk 1.0.6 installed on Manrake 10.1. I
had problems using ulaw codec, RTT was skyrocketing in
few seconds so calls experienced huge delays. Then I
started changing my codecs in iax.conf file and
monitor the send and receive speed rate from Mandrake
Control Center. I
On Apr 16, 2005, at 4:32 AM, parijat wrote:
Hi,
Thanks for helping me out.
I want to clear out few more points
1) zaptel cards receive PCM from PSTN.
They receive it in either uLaw, aLaw, or signed linear PCM, depending
on the model and what it's connected to.
In what form do they give it to
I would use mtr, leave it running, and
look for the node that is causing the problem to suddenly develop packet loss.
You should see very clearly where the problem is.
Also, try visualroute on a windows
computer connected to the same network.
Chris Mason
www.anguillaguide.com
What type of modem are you using. If Zyxel or other consumer level try
restarting and feel the box to see if it is hot. I do like the
consumer level products but heat is there enemy. This could also be
caused by out of date DNS server settings.
On 4/16/05, Wilson Pickett [EMAIL PROTECTED] wrote:
Hi,
Il giorno sab, 16-04-2005 alle 13:30 +0200, Robson Ribeiro ha scritto:
Frtiz is a nightmare although it is cheap and I have seen it working.
I have been trying to install it for some days without success but one
thing is for sure: you have to use the right Kernel (they are
available for
I have a question regarding setting the CallerID, more specifically the
Caller Name. In all of my menus I set the current Caller Name so it
displays what menu they are in when the phone rings for my users. We run
seperate companies so it's easy for us to distinguish how to answer the
On 13:30, Sat 16 Apr 05, Robson Ribeiro wrote:
Frtiz is a nightmare although it is cheap and I have seen it working. I have
been trying to install it for some days without success but one thing is for
sure: you have to use the right Kernel (they are available for 2.4.20 and
2.6.something).
The just released version 0.90 of IPSwitchBoard now also supports Zap
Channels.
* Zap support added - thank you to Daniel for letting me use his server and
to Tim for patiently testing
* Configuration page can now be password protected
* Option to show technology on buttons (SIP, IAX2, Zap)
*
Hi Ronald,
The author has just posted a new version :-) Download:
http://ipswitchboard.thorben.dk
You need to type in the context on the configuration page, then it should
work.
Thorben
Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse
news:[EMAIL PROTECTED]
I tried many different
Search the wiki for campon. It may help point you in the right direction.
- Original Message -
From: Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, April 16, 2005 3:43 AM
Subject:
Hi,
we've just posted a completed interview with Kevin Fleming on the Daily
Asterisk News.
You can access the interview from here:
http://www.sineapps.com/news.php?rssid=667
:)
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily
Crashes all the time.
Object Reference not set to an instance of an object as
IPS.ipdkpbx.GetBtnTest(String btnName)
Chris Mason
www.anguillaguide.com
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Thorben Jensen
Sent: Saturday,
Hallo,
anyone with experience with USR Sportster PCI BRI cards with Asterisk?
Also, is there any Digium officially supported BRI card?
Francesco
[EMAIL PROTECTED] wrote on 16/04/2005 14.24.29:
On 13:30, Sat 16 Apr 05, Robson Ribeiro wrote:
Frtiz is a nightmare although it is cheap and I
Hi Razza,
I don't know what country you are in, or what your country's telephone
numbers look like, but it seems from your dialplan that if you dial an
outside number it needs to start with 0X.
So if you dial 012345, the Sipura will dial 012345 on the fxo port.
If your line needs to dial 12345,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Frtiz is a nightmare
Is Fritz the only option really for ISDN BRI (without paying $400+)? I'm
just wondering if there is a more viable, not-as-cheap alternative to
the Fritz cards?
Thanks!
Henry.
Robson Ribeiro wrote:
Frtiz is a nightmare
Hi Chris,
Thank you for reporting,
Pls. download again, I have fixed the bug (I hope).
Thorben
Chris Mason (Lists) [EMAIL PROTECTED] skrev i en meddelelse
news:[EMAIL PROTECTED]
Crashes all the time.
Object Reference not set to an instance of an object as
IPS.ipdkpbx.GetBtnTest(String
Wilson Pickett [EMAIL PROTECTED] wrote:
About 10 days ago, I started seeing message where ALL providers and phones
would become UNREACHABLE for about 10 seconds, then the same packagae of
messages saying they were now REACHABLE. This happens about every 20-60
minutes, usually at least once
Hi all,
i'm getting about 1-3 NOTICE[1915]: rtp.c:453 ast_rtp_read: RTP:
Received packet with bad UDP checksum message per call on CVS HEAD
from 31 Mar. which seems some changes regarding rtpchecksums is made
at that time.
setting rtpchecksums to no or yes in rtp.conf doesn't make any sense.
now
On Apr 13, 2005, at 8:36 AM, Eric Wieling wrote:
The only time PLC makes sense is thwn you are converting FROM VoIP to
something else. So PLC would be done on chan_sip or chan_IAX, or
chan_h323 on the receiving end. This is for 1.0.x.
I don't even see where PLC was mentioned in this thread..
I am trying for 1 month to use IConnect with * an I am not satisfied with the
sound quality. Can someone to recommend me some VOIP to PTSN provider.
Thanks a lot Ilija Poznic
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I installed asterisk 1.0.7 successfully on VMware workstation with
fedora 3 as guest.
Of course without any hardware only pure asterisk. It works fine for
testing.
SCollins wrote:
Newbie Question
Has anybody installed [EMAIL PROTECTED] on VMware Workstation (w/ WMware
Tools)successfully?
Hardware Pentium 1.4 Gig 1 Meg ram 1
FXO100 Card Sipura 2000 Local Network Router SMC Codec 711
Asterisk @ home (lastest)
On average it take almost 10 13 Secs to make an
outbound call to a local number.
Is this a normal time ? Is there something that can be done
to cut this time
I noticed that there is some interest in MGCP slave operation for Asterisk
to enable it to work with the ATT Callvantage offering. I have tried the
FXS/FXO connection to Asterisk and the Linksys TA with little success.
Dropped calls are the biggest problem, which does not occur when the phone
is
On 2005/04/16, at 15:48, chawki hammoud wrote:
Do you know where to add the modprobe lines so I don't
have to manually do that after each boot?
On Redhat 9.0 and Yellow Dog 3.0.1 I , I did this to modprobe and ztcfg
on boot:
cd /usr/src/zaptel ;or wherever the zaptel source is located
make
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kumara
Jayaweera
Sent: Friday, April 15, 2005 6:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How do I make Extention in my Asterisk PBX
Hi All, I have a working Asterisk PBX in my
All,
I'm working on an Asterisk 1.0.7 system that is acting as a B2BUA SIP
gateway. canreinvite=no is set in the global section of sip.conf, and
it's important that it be there. I have
Cisco --- Asterisk --- Multiple destinations
Some destinations support both G711 and G729, but some only
Hi Lee. Actually is very easy to connect to asterisk servers. Please
read the IAX related documentation in voip-info:
http://www.voip-info.org/wiki-Asterisk+config+iax.conf
and if you have troubles, ask here :-)
Best Regards
On 4/16/05, lee siang fong [EMAIL PROTECTED] wrote:
Hi,
I am
Hardware - Pentium 1.4 Gig - 1 Meg ram - 1 FXO100 Card - Sipura 2000 -
Local Network Router SMC -Codec 711 - Asterisk @ home (lastest)
On average it take almost 10 - 13 Secs to make an outbound call to a local
number.
Is this a normal time ? Is there something that can be done to cut this
Today I ran a different box with Windows XP on a different router and
disconnected the phones from the filter. I ran pingplotter, which is
like mtr for windows and was able to determine that there is nearly
always about 8 minutes between these glitches.
I pinged a server of mine in the US and the
L.S.
Subjectline says it all. I made a mistake upgrading the firmware.
Got anothet\r one working allright. Dis- and re-assembling the
device is no problem.
Thanks for any advice,
Loek---
When we run out of nice things to say, it's a good indicator that an
argument is not following productive
Any one have Asterisk running on a Openbrick [1] box ?
[1] www.openbrick.org
--
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
e-mail: [EMAIL PROTECTED]
www : http://www.telconet.net
http://www.telcocarrier.net
Linux User:
/etc/modprobe.conf on slackware..
On Sat, 2005-04-16 at 08:05, goldhorse wrote:
On 2005/04/16, at 15:48, chawki hammoud wrote:
Do you know where to add the modprobe lines so I don't
have to manually do that after each boot?
On Redhat 9.0 and Yellow Dog 3.0.1 I , I did this to
Question 1:
If I am going to be selling hardware phones to the enduser do there
accounts have to be SIP or can they still be IAX (I find IAX is better
for firewalls)?
Question 2:
Is there a way to limit connections? For example, extension 7895483 can
only have 1 call in progress at a time
Hi,
I have a Panasonic Cordless phone and want to use the built-in answer
machine instead of an asterisk voice mailbox. The problem is now that
the answer machine plays the announcement and exactly when it wants to
record, asterisk reports a zap Hangup. The caller never even hears the beep.
Any
I have a problem connecting asterisk with wipphone...
Does anybody has made this sip connection..?
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Hello,
I'm having trouble getting BT100 to identify NAT type reliably for Asterisk.
My setup is as follows:
- Asterisk is on the open internet 142.x.x.41
- BT100 phones are behind NATs
- I use STUN for my BT100 : 142.x.x.41 (same server as Asterisk)
- BT100 firmware (tried .16,.18,.23 same
Hi Bryan, thanks, but we found a
combination of slots that resulted in the TDM (and our BRI ISDN card)
getting a unique IP address.
IBM has a lot to answer for with their xSeries 206 server - looks like
they've given the BIOS a lobotomy or something, as it doesn't allow you
to configure much.
Tomas,
Yes, BT100 is a little picky on the use of Stun Servers. For example, it
will not work at all with Vovida Stun server. Also, Stun negotiation takes
some time. So if you rebooted the phone, I would suggest waiting 15-30
seconds until phone syncs up with Stun server and requests binding. You
This answers a lot of questions
- I am in fact using Vovida STUN (so I have to find a replacement)
- I don't have 2IPs on the Asterisk server - so that's wrong too
Thanks for your help!! :-)
Tomas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
One more question ... I did a search on Google for STUN servers and didn't
find any other open source server other than Vovida's
What other open source Stun servers are there? And if there are none, what
commercial one have you found to work well with BT100?
Thanks again,
Tomas
-Original
Thanks,
I'll look into it...
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry
Devito
Sent: Saturday, April 16, 2005 9:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Extensions busy queue
Search
Lucky you, my spa-3000 likes to dial 911. So far the local cops have
been nice about it though. (my mobile number ends in 9110)
I have been having trouble getting quality and tone transmission right.
Seems to be a delay, but there is less than 5ms ping time. G729 is 'ok'
and ulaw seems
Tomas,
There is mystun on sourceforge, but I think the only way to down load it is
to build it from cvs source. I normally use public stun servers from
grandstream or xten.
Hth
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian
Sent:
Ben Price wrote:
Question 1:
If I am going to be selling hardware phones to the enduser do there
accounts have to be SIP or can they still be IAX (I find IAX is better
for firewalls)?
Think about the scale you want to go. If you have many phones you may
want to install a SIP Proxy (e.g. SER)
We have just posted a review of the Sipura SPA-1001 ATA.
http://geekgazette.com/index.php?option=com_contenttask=viewid=26
Kerry Garrison
http://www.geekgazette.com
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Thanks very much. I'm still learning things. Dialing rules isn't something I've fully come to grips with yet. Now
that I see your explaination it's dead easy.
I'll put this in and give it a go.
Thanks again!
Cheers!!!
Shahar Livne wrote:
Hi Robert,
extensions.conf grammar is very easy:
To
Good point. Here is another Suggestion. Why not use the existing analog
phones to their PBX and go out to channel banks for their phone line
trunks. Then go to Asterisk for the rest. They don't have 700 trunks.
This will save on equipment costs and you will get some of the benefits
of
Hmm still not really working. This is what I have in a context that I'm part
of:
[outbound_world]
; Dial rule for China
exten = _686.,1,Dial(IAX2/user:[EMAIL PROTECTED]/011${EXTEN:1})
; Dial rules for UK
exten = _60[1278].,1,Dial(IAX2/user:[EMAIL PROTECTED]/01144${EXTEN:2})
; Dial everything
Hi list!
I'm having trouble receiving faxes. After the second page the line is
disconnected and my HP all-in-one LaserJet 3350 displays a 'communication
error'
I've worked my way through the list back to November and googled around
extensively but didn't find a solution. I did find several
I use voipjet and am quite pleased. Good enough rates and no
noticeable quality issues.
http://www.voipjet.com
Plus, you can even test it before you buy.
-Andy
On 4/16/05, Ilija Poznic [EMAIL PROTECTED] wrote:
I am trying for 1 month to use IConnect with * an I am not satisfied with the
The cat 3 issue depends on your phone, if you went with VoIP phones,
you would need to make sure that it could be set/forced to 10Mbps. I
have only used Cisco phones, and, save the 7910 and 7902 (may a few
others), they all are fully capable of doing 100Mbps because of their
internal switch (you
Vikram,
Would they really be able to tell if I have VOIP and POTS terminating
on the PBX? Theoretically, its not like I'll be using this 100% of
the time for sending VOIP calls to the POTS line. Probably maybe once
or twice a month? It's main function is to act as a PBX with
voicemail and
Got it sorted.. it was an issue with the order the exten's were processed. Found the page on the wiki about forcing
your own sort.. it's all working now.
Thanks for the hand in finding the light switch :)
Cheers!!!
Robert P. McKenzie wrote:
Hmm still not really working. This is what I have in
Hello,
I have a strange problem . I am using Asterisk CVS and
when I wnat to use two port SIP gateways like CISCO ATA , Linksys or
Sipura I can only use one port the other port does not register . I
can only use one of the ports.
Can anybody help?
Ehsanul Karim
Hi,
For those that were having the same line noise problem that we were, an
update:
* Our TDM400P *was* sharing an IRQ, despite the output from 'cat
/proc/interrupts' showing that it wasn't. Running 'lspci -v'
showed that it was and we had to perform some card juggling to get
Andy Hamilton wrote:
I use voipjet and am quite pleased. Good enough rates and no
noticeable quality issues.
http://www.voipjet.com
Plus, you can even test it before you buy.
On their pricing page, they have:-
There are some providers who can terminate some, but not all, 1800
numbers for free.
I've found something that seems to be the norm when I connect to some
of the smaller providers out there. At first I thought it was my
setup, but yesterday I was trying out voicepulse connect, and they are
the first ones that I have not had this problem with. Their quality
also seems to be on
Chris Hills [EMAIL PROTECTED] wrote:
[...]
There are some providers who can terminate some, but not all, 1800
numbers for free. (If they could terminate all 1800 numbers for
free, then we'd use them!)
I don't understand - I thought all 1800 numbers were free?
They're not like UK 0800s - the
Hi,
I am having a few issue withs [EMAIL PROTECTED] 0.9.
1) Setting up a sip connection, with voicemail to use with (eyebeam/X-pro)
softphone. I can receive voicemail no problem and even in this revision
the MWI seems to work correctly, though when i try to go to the message
center, (*98) and
Hi,
I am having a few issue withs [EMAIL PROTECTED] 0.9.
1) Setting up a sip connection, with voicemail to use with (eyebeam/X-pro)
softphone. I can receive voicemail no problem and even in this revision
the MWI seems to work correctly, though when i try to go to the message
center, (*98) and
I use Askerisk in my home as my home phone system. I use to make long
distance calls because of the cost savings that I can get. I also
share my phone system with my neighbor via a wireless link so that he
can also use * to make long distance calls at a savings. He also uses
that as a second
Judging by the price of my single phone line it would be hard to have
much of a margin even with a T1.
First of all, a T1 is cheaper than 23 POTS lines, as you noticed, but not by
a huge amount. However, what you must remember is that with T1, you have
DIDs. That means you can have 23 numbers
See inline reply:
On Sat, 16 Apr 2005, Henry Devito wrote:
1) Setting up a sip connection, with voicemail to use with (eyebeam/X-pro)
softphone. I can receive voicemail no problem and even in this revision
the MWI seems to work correctly, though when i try to go to the message
center,
Andrew Niemantsverdriet wrote:
I use Askerisk in my home as my home phone system. I use to make long
distance calls because of the cost savings that I can get. I also
share my phone system with my neighbor via a wireless link so that he
can also use * to make long distance calls at a savings. He
800 numbers are free to the caller because the recipient pays the charge.
Voipjet has no way to get paid anything for carrying the calls, hence they
are unwilling to use their resources to move calls with no revenue.
Can you blame them? :)
/edg
--On Saturday, April 16, 2005 9:44 PM +0100 Chris
Hi List,
What is the good client softphone for windows that connects to my Asterisk
Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php
Give it a try
N.B.: I should release an updated version really soon now
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Senerio
multiple * boxes connecting to a central * box with T1 card via IAX2.
1box 1 abd 2 work fine all the time
box 3 - after approx 10-15 minutes with no calls - central box with T1 card
fails to deliver incoming calls to box 3.
Connectivity is good, * exten-2-exten good
in order to allow
Will sip/iax devices designed for European use also work in Russia?
I'm specifically looking at using the Sipura ata's if anyone can
confirm they work.
Chris
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Hi,
I am looking
around for phone for a receptionist that has a list of extensions and so forth
like the cisco expansion module. But I had read
somewhere that the expansion module does not work with asterisk. Please
confirm.
Thank You
--
We are needing a IAX softphone with support for supervised transfer
(among others). We have tried the following ones without success,
either because it does not support supervised transfer, or because
some other codec issue that made it unusable:
iaxComm
Diax
IaxPhone
IAXTelefon
Do
Senerio
multiple * boxes connecting to a central * box with T1 card via IAX2.
1box 1 abd 2 work fine all the time
box 3 - after approx 10-15 minutes with no calls - central box with T1
card
fails to deliver incoming calls to box 3.
Connectivity is good, * exten-2-exten good
Ok, I am
central box connects to PSTN with T1 card
boxes 1-3 use IAX to connect to central box via IP and deliver in/out bound
call to PSTN.
we have multiple division that connect to a central LD PRI.
really quit simple and using a LD PRI gives us much better LD pricing.
sorry about the confusion. Help
That's the nice thing about VOIP devices... Ethernet is the same the world over
no matter what the phone system is like. So long as you've got comptible power
you're all set.
Ian
snacktime [EMAIL PROTECTED] 16/04/2005 19:20
Will sip/iax devices designed for European use also work in Russia?
Can anybody recommend relabel four port FXS ATA adapter?
Or any independent reviews.
--
#Joseph
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The way to do this in my opinion is to stay with analog phones in the
room and *not* ip phones for a couple of reasons:
With Cellphones the way they are right now, you will never recover
and/or justify the costs ($60 per phone plus wiring for IP, vs $5 per
phone and no wiring for analog).
Over
On Sat, 2005-04-16 at 20:04 -0600, Joseph wrote:
Can anybody recommend relabel four port FXS ATA adapter?
Or any independent reviews.
Is anybody using SIP and IAX 1 to 4 port FXS - Nokphil Gatway, 1-4 port
FXS?
--
#Joseph
___
Asterisk-Users
I have a number of situations where in the past I would get a native
bridge, IAX to IAX - e.g., call coming in on an IAX VOIP line being
forwarded to PSTN via another IAX termination provider, or calls made
from my IAXY to a PSTN number via my Asterisk box to a VOIP provider.
I find the native
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