Hi you all,
Thanks for the help, got it working! The mpg123 in combination with the
mpg123 directory (executable MUST be in /usr/local/bin AND in /usr/bin)
was the problem that MOH was not working
Thanks!
Jeroen
Brian West wrote:
put mpg123 in /usr/bin
bkw
On Tue, 19 Aug 2003,
Hello,
Is it possible to limit the number of user in a
particular conference room?
Foong
so i call from a sip phone (grandstream) to
a cell via x100p
PSTN side hears everything nice, no echo.
on the SIP side I hear myself about .1 to .2 sec
later...
any thoughts on how to resolve this.
mucho thanks to everyone that has been helpful :)
john
should be CVS
Foong
- Original Message -
From: Steven Thomas [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 20, 2003 2:42 PM
Subject: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for
Chan_h323
Hi,
Can someone tell me where to find the stated
I thought that the CVS would only contain the lastest code - being:
OpenH323: v1.12.2
PWLib: v1.5.2
Is this not the case?
Thanks
Regards,
Steven Thomas
you can do
cvs update -r v1_11_7
to get version 1.11.7 for openh323
Foong
- Original Message -
From: Steven Thomas [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 20, 2003 2:51 PM
Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB
for Chan_h323
Thanks - because of my ignorance using the CVS archive - could you please
give me the full command - thanks.
Regards,
Steven Thomas
Is there a way to determine what codec the remote server
wants to use in a SIP session for an incoming call by looking at something,
possiby sip debug?
export CVSROOT=:pserver:[EMAIL PROTECTED]:/cvsroot/openh323
cvs login
CVS password: press enter
cd /root
cvs checkout openh323
cd openh323
cvs update -r v1_11_7
I usually get the latest version then down grade to older version, If you
know how to get the older version directly, let me know.
Is there a way to determine what codec the remote server wants to use in
a SIP session for an incoming call by looking at something, possiby sip
debug?
Take a look in the archives this was covered a couple of days ago..
the command you are looking for is sip show channels.. and then look in
Citeren Chee Foong [EMAIL PROTECTED]:
Hello,
Is it possible to limit the number of user in a particular conference room?
Foong
Hi,
I think the easiest way is to create a counter that adds one when a user joins
and subtracts when a user leaves or hangs up. A few simple AGI scripts
I already tried that, it says unknown.
I suspect it is requiring the G723 codec.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut .
Sent: Wednesday, August 20, 2003 3:28 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP using which
At 7:27 AM + 8/20/03, WipeOut . wrote:
Is there a way to determine what codec the remote server wants to use in
a SIP session for an incoming call by looking at something, possiby sip
debug?
Take a look in the archives this was covered a couple of days ago..
the command you are looking
Hi,
While trying to update latest CVS, during make install to zaptel, I
got weird error message (down under).
Anyone had same kind of problem? What would be the solution?
-Johanna
_
fi
/sbin/depmod -a
depmod: cannot read ELF header from
Actually I got it working right before I gave up (I had the wrong line
in my config commented out)
But now I get these messages when I try to playback a recording:
NOTICE[16401]: File channel.c, Line 1406 (ast_set_write_format): Unable
to find a path from GSM to G723
WARNING[16401]: File
Are the dialogic DTI series cards supported in
asterisk? I know there's standard API, but I don't know if it supports
only the cards listed on the digium site, or if it will support ALL dialogic
cards.. Sorry, I *AM* a newbie to this stuff, just trying to get my hands
on a good card.
Hello,
I`ve upgraded my Snom 100 to the new version of firmware that is
snom100-2.00n-SIP.bin, and they did fix the GSM, that is the nice news,
it is very clear and nice almost indistinguishable from the G.711. But
there still a problem, when doing transfers or for example diling the
500 (
Hello,
I have problem setting up queue.
Everything works nice, but I would like to have
some kind of announcement while playing MusicOnHold.
Is it possible?
If yes how I can set it up.
Bartek
On Wed, 2003-08-20 at 02:55, Florian Overkamp wrote:
Citeren Chee Foong [EMAIL PROTECTED]:
Hello,
Is it possible to limit the number of user in a particular conference room?
Foong
Hi,
I think the easiest way is to create a counter that adds one when a user joins
and
On Wed, 2003-08-20 at 02:07, Johanna Kangas wrote:
Hi,
While trying to update latest CVS, during make install to zaptel, I
got weird error message (down under).
Anyone had same kind of problem? What would be the solution?
OPEN EYES AND READ.
Your problem is in hisax, not zaptel.
On Wed, 2003-08-20 at 00:50, Dave Cotton wrote:
On Tue, 2003-08-19 at 22:42, Steven Critchfield wrote:
So far I have received 43 since 3am till 3:45pm
According to mails in the ser list it's there also, and around the same
time of day.
Sounds appropriate since I have received a bounce
I upgraded to the latest CVS yesterday (and this morning again), and
whenever I execute the reload command Asterisk seems to hang. While the
current calls aren't dropped, no new calls can be made. The CLI isn't
responding properly either. The only way to get going again is to exit
the CLI and stop
yes start it with asterisk -gc
watch and see what the error is.
bkw
On Wed, 20 Aug 2003, Marcus Adolfsson wrote:
I upgraded to the latest CVS yesterday (and this morning again), and
whenever I execute the reload command Asterisk seems to hang. While the
current calls aren't
You can also check www.openh323.org/bin/
bkw
On Wed, 20 Aug 2003, Steven Thomas wrote:
Thanks - because of my ignorance using the CVS archive - could you please
give me the full command - thanks.
Regards,
Steven Thomas
Chee Foong
It is possible to connect ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank to Asterisk
?
Somebody offered me that hardware, but I do not know if thats
good hardware for Asterisk.
rgs,
Bartosz
MOH requires that Asterisk transcodes (It also has to transcode to for
PSTN calls and voicemail and playing any sound files). Asterisk can't
transcode to or from G723. Nope. Doesn't work. May very well never
work. Use a different codec.
On Wed, 2003-08-20 at 03:47, Andrew Joakimsen wrote:
Conference call problem - do not have any special hardware added to the
system yet.
Did the following:
* Uncommented the ztdummy.c in the Makefile (/zaptel) - recompiled all
* Extensions.conf
exten = 2675,1,meetme,2675
* meetme.conf
conf = 2675
When I dial 2675 I get the
Hi,
I've seen some postings on the Directory application, but haven't seen
too many resolution postings. Has anyone experienced where the
Directory app doesn't even answer when called? For example, using the
config below, dialing 899 results in just a continual ringing sound.
extensions.conf
The FXO ports will only allow you to connect phone lines, not actual
phones, but since FXO ports are more expensive in general than FXS ones,
it's likely you could find someone to trade. We probably should have a
list dedicated to trading/selling/buying asterisk related hardware, but
failing that
Jeroen wrote:
Conference call problem - do not have any special hardware added to
the system yet.
Did the following:
* Uncommented the ztdummy.c in the Makefile (/zaptel) - recompiled
all [...] Any ideas?
When you do an lsmod, is ztdummy listed?
If you do a depmod -a is there any output,
On Wed, 2003-08-20 at 14:36, Steven Critchfield wrote:
And do you expect a crack dealer to stop selling crack? This won't
change till enough lusers are educated.
No Gates got of the hook last time, too political to discuss here but
perhaps John Dvorak's article hits the nail on the head.
On Wed, 2003-08-20 at 07:58, Mark Spencer wrote:
The FXO ports will only allow you to connect phone lines, not actual
phones, but since FXO ports are more expensive in general than FXS ones,
it's likely you could find someone to trade. We probably should have a
list dedicated to
I want to connect analog telephone lines only. The analog lines telecom
gives you
:)
- Original Message -
From: Steve Meyers [EMAIL PROTECTED]
To: Asterisk List [EMAIL PROTECTED]
Sent: Wednesday, August 20, 2003 11:34 AM
Subject: Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port
Hi Almaw,
The following:
* Asterisk up running
* lsmod - no ztdummy module loaded
* depmod -a - no output
So I tried to modprobe the ztdummy --- with result! Conference is
running without problems .. do you knwo if there is a manual or
something like that which summarises all these
Then yes, it will work and do what you're looking for it to do.
On Wed, 20 Aug 2003, Bartosz Jozwiak wrote:
I want to connect analog telephone lines only. The analog lines telecom
gives you
:)
- Original Message -
From: Steve Meyers [EMAIL PROTECTED]
To: Asterisk List [EMAIL
Thanks for your help.
Bart
- Original Message -
From: Steve Creel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 20, 2003 11:47 AM
Subject: Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank
Then yes, it will work and do what you're looking for it to
Hello,
Again one more question about
hardware.
What could you suggest me to buy.
I need hardware to connect let's say 4 analog
lines. (FXO).
This hardware should "talk" to Asterisk of
course..
Thanks very much for some advices :)
Bartek
Hello,
I've had quite a few requests for this info, so I thought I'd copy this
to the list as well. Since I don't really monitor the list anymore,
queries should be directed back to me if you have problems.
On Wed, 2003-08-20 at 04:08, Sjur Eivind Usken wrote:
Dear Ben,
I saw your posting
Did you enable echocancel and echocancelwhilebridged?
Did you put them in the correct location in the zapata.conf ? It has to be
before the channel statement (this is what threw me for a week)
If you tail -f debug in the /var/log/asterisk you can watch the call and see
if echo cancel was kicking
Also dont forget FXO cards must be L2 with minimum REV K firmware to support
Caller ID.
see http://www.wwworks-inc.com/asterisk/
Then yes, it will work and do what you're looking for it to do.
On Wed, 20 Aug 2003, Bartosz Jozwiak wrote:
I want to connect analog telephone lines only. The
Martin - et all,
I'm having the same issue. I have a PRI T1 on a T100P with six 7940 Cisco
Phones w/SIP load 4.4. What hardware do u have? The worst part is that my
system will sometimes just busy out even if I do not issue a reload command!
However if I issue reload it's a sure thing * will
Digium makes a 4 port card. It'd be hard to get 4 lines with quicknet
hardware.
Bartosz Jozwiak wrote:
Hello,
Again one more question about hardware.
What could you suggest me to buy.
I need hardware to connect let's say 4 analog lines. (FXO).
This hardware should talk to Asterisk of course..
Does the Asterisk server support VAD (aka Silence
Suppression)? I want my SIP phones (7960's) to use VAD when dialing out the
x100P interface. I know the phone can do VAD , can the Asterisk server be setup
to do it? and if so, where do I set the configuration?
Thanks
Lee Goodman
you can download current release tarballs from openh323.org. I just put
them up yesterday
Steven Thomas wrote:
Thanks - because of my ignorance using the CVS archive - could you please
give me the full command - thanks.
Regards,
Steven Thomas
Okay,
I am facing a move in two months to newly renovated space. I
have to decide *this week* between:
A) Pull LAN and phone cables, prepare to move and expand our
traditional PBX (Panasonic KX-TD1232 and VPS200).
or
B) Pull only LAN cables, go VoIP, use Asterisk as PBX.
It is *not* an
Does anyone have X-Lite build 1059 working fully with Asterisk?
The GSM Codec works very well now but we have problems when using G711
in that when I setup a ping between the two sites and then watch the
latency, it steadily increases and starts at about 150ms and goes up to
2500ms within about
I have three Cisco 7960 phones/SIP 5.3 using two Wildcard X100Ps and IAX
service from Nufone. It worked fine on my earlier installed CVS from
6/10. I have not noticed any random hangs, altough it has only been
running for two days.
Thanks,
Marcus
-Original Message-
From: [EMAIL
At 10:42 AM 8/20/2003 -0500, you wrote:
I've literally read the last year's worth of posts to
asterisk-users to get a feel for the situation. Since you
don't see posts of the form installed it, just working, no
problems very often, you could get the opinion that everyone has
problems since that
What is your SIP registration timeout?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Marcus
Adolfsson
Sent: Wednesday, August 20, 2003 11:50 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] reload not working
I have three Cisco 7960 phones/SIP 5.3
Okay,
I am facing a move in two months to newly renovated space. I
have to decide *this week* between:
A) Pull LAN and phone cables, prepare to move and expand our
traditional PBX (Panasonic KX-TD1232 and VPS200).
or
B) Pull only LAN cables, go VoIP, use Asterisk as PBX.
It
astman or gastman would tell you this info. And yes we us it in
production right now. Works better than anything we have had previously.
bkw
On Wed, 20 Aug 2003, Ernest W. Lessenger wrote:
At 10:42 AM 8/20/2003 -0500, you wrote:
I've literally read the last year's worth of posts to
Brian,
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/enum.conf': Not found (No such file or
directory)
== Parsing '/etc/asterisk/rtp.conf': Not found (No such file or
directory)
== RTP Allocating from port range 5000 - 31000
-- Reloading module
Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP phones
(7960's) to use VAD when dialing out the x100P interface. I know the phone can do
VAD , can the Asterisk server be setup to do it? and if so, where do I set the
configuration?
Thanks
Lee Goodman
Can you do remote loopup from your switches side ... and look asterisk's
T1 and check if your transmission is ok ?
regards
Martin
On Tue, 19 Aug 2003, Barry Porch wrote:
Martin,
Here is the trace you asked for. It's quite lengthy so I'm attaching it
as a text file. The way I generated
Is anyone out there using an AudioCodes MP108 8-Port FXO Analog Gateway
(SIP) with asterisk to support both inbound and outbound calling? If so,
I'm interested to hear how it works, and I'd love to see some example confs
(both in sip.conf and on the MP108).
This product has been recommended to
VAD is evil. I hate it. I find when we used it.. you keep asking people
to repeat stuff all the time.. and it was just anoying.
bkw
On Wed, 20 Aug 2003, WipeOut . wrote:
Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP
phones (7960's) to use VAD when dialing
I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one
another using IAX/IAX2 trunks.
I've managed to get a semi-functional NAT Firewall working as a PBX
(with Asterisk running directly on the firewall itself), but there are
issues with bind()ing to various interfaces which is
I would use the latest CVS for one. And try again.
bkw
On Wed, 20 Aug 2003, Ian Blenke wrote:
I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one
another using IAX/IAX2 trunks.
I've managed to get a semi-functional NAT Firewall working as a PBX
(with Asterisk running
I always keep known working code and libs at
http://www.nufone.net/downloads
Jeremy McNamara
Steven Thomas wrote:
Hi,
Can someone tell me where to find the stated correct versions of Openh323
and PWLIB for Chan_h323? The README states the versions required are:
Open H.323 v1.11.7
Or you can jump on #asterisk
bkw
On Wed, 20 Aug 2003, John Brown wrote:
We are getting ready to replace our old Panasonic PBX with
an Asterisk system. I'd say its ready for prime time.
THe other thing is to have a good consultant in your back pocket
for those now how do I do this.
I can
On Wed, 20 Aug 2003, Mike Ciholas wrote:
I am facing a move in two months to newly renovated space. I
have to decide *this week* between:
A) Pull LAN and phone cables, prepare to move and expand our
traditional PBX (Panasonic KX-TD1232 and VPS200).
or
B) Pull only LAN cables, go VoIP, use
Brian West wrote:
I would use the latest CVS for one. And try again.
Unfortunately, I've tried numerous times to get a current CVS trunk
snapshot to talk to *anything*, to no avail. Even getting my Grandstream
phones to register with it was an apparent excersize in futility.
Dropping back to
Hello,
Does somebody knows how to connect Adtran Total
Access to Asterisk, is it with T1 ?
bart
Mike Ciholas wrote:
Okay,
I am facing a move in two months to newly renovated space. I
have to decide *this week* between:
A) Pull LAN and phone cables, prepare to move and expand our
traditional PBX (Panasonic KX-TD1232 and VPS200).
or
B) Pull only LAN cables, go VoIP, use
When I turn on VAD on cisco ATA186, asterisk shows:
Aug 20 11:12:20 NOTICE[618513]: File rtp.c, Line 239 (process_rfc3389):
RFC3389 support incomplete. Turn off on client if possible
RCF3389 defines Payload for Comfort Noise, that is used with VAD.
So I
And if one cannot use a different codec?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Wednesday, August 20, 2003 9:51 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] G723 (was SIP using which codec?)
MOH requires that
I'm in almost the same situation as you. However, I'm mostly worried that
the customer service desk here will start to complain that they can't tell
how many calls are in the queue any more (our current phone tells us how
many calls are ringing, on hold, etc).
yea we ran into that as well, we used
On Mon, 18 Aug 2003, Mark Spencer wrote:
It's up one directly. It just moved.
Run make in h323 then do make install on asterisk again.
On Mon, 18 Aug 2003, John Fortman wrote:
What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp,
ast_h323.h and chan_h323.h but no
On Wed, 2003-08-20 at 11:09, Ian Blenke wrote:
Brian West wrote:
I would use the latest CVS for one. And try again.
Unfortunately, I've tried numerous times to get a current CVS trunk
snapshot to talk to *anything*, to no avail. Even getting my Grandstream
phones to register with it was
On Wed, Aug 20, 2003 at 12:13:07PM -0500, Dave Weis wrote:
On Wed, 20 Aug 2003, Mike Ciholas wrote:
I am facing a move in two months to newly renovated space. I
have to decide *this week* between:
A) Pull LAN and phone cables, prepare to move and expand our
traditional PBX (Panasonic
My switch doesn't let me set up a loop but I am confident that
everything is OK at the T1 layer. I can connect via robbed bit to the
Asterisk box with no problem. Also I can use my T1/PRI tester towards
either systems and it works fine with PRI and I can place and receive
calls.
There seems to
If you want to be able to use G723 from a legal standpoint you will have
to license the codec from the current patent holders. The patent
holder's price list can be found at
http://www.dspg.com/technology/LicensePricing.html
If you obtain a license to use G723 then Digium or the Asterisk user
Just idly watching * in console mode and saw that someone from
50.49.54.102 tried to register with my *.
whois gives:-
OrgName:Internet Assigned Numbers Authority
OrgID: IANA
Address:4676 Admiralty Way, Suite 330
City: Marina del Rey
StateProv: CA
PostalCode: 90292-6695
As for cables. Pull ONLY Cat5 or Cat5e as they can be
used for either Ethernet OR voice. You can then
use a plug pannel in the phone closet to route a
spicif cable to either a voice or data switch.
Is Asterick ready??
I'd say the software is but ONLY IF
1) You or someone you can depend on
I would also like to see a more structured release
program. It's kind
of scary to tell people that they should just use
the latest CVS code.
That's where consultants earn their money. They
should be preforming some kind of quality control.
You build the code, get it to work, test it and
This may have nothing to do with it but have you verified your timing?
Make sure one end of the T1 is using an internal clock and the other
end is using timing off of the T1.
Don Pobanz
On Wednesday, August 20, 2003 12:37 PM, Barry Porch
[SMTP:[EMAIL PROTECTED] wrote:
My switch doesn't let
Thanks, that's the answer I was looking for.
Do we know if VAD will ever be supported? I know some people don't like VAD
and in my testing, how well VAD works depends on how well it was coded (and
the hardware I suspect). I have seen very good and very bad implementations
of VAD.
I have a real
Title: Message
Hi
all,
I am experiencing a
problem with the quality of the voice communication between an IAX based
softphone (WinIAX) and an outside line through a FXO port or even with a regular
analog phone connected to a FXS port. The party using the IAX softphone hears
his own echo a
I have been using Slackware 9 with the sources for ptlib_unix, pwlib_unix,
openh323, asterisk, zaptel and libpri in /root/src
1) load all 6 packages (+ ohphone if you need an h323 client) from cvs to
/root/src
2) /root/src/pwlib: configure, make, make install, ldconfig (not all that
sure why, but
Chris Albertson wrote:
I would also like to see a more structured release
program. It's kind
of scary to tell people that they should just use
the latest CVS code.
For testing and development, this isn't a bad thing - as long as the
trunk codebase generally *compiles* and *runs* more often than
Comfort Noise and VAD are diffrent things.
bkw
On Wed, 20 Aug 2003, Eduardo Goncalves wrote:
When I turn on VAD on cisco ATA186, asterisk shows:
Aug 20 11:12:20 NOTICE[618513]: File rtp.c, Line 239 (process_rfc3389):
RFC3389 support incomplete. Turn off on client if possible
Great! Thanks for the recommendation. I'll beat on Redhat a little bit
longer, then try to load slackware and give that a whirl.
Thanks again.
Sean
On Wed, 20 Aug 2003, John Fortman wrote:
I have been using Slackware 9 with the sources for ptlib_unix, pwlib_unix,
openh323, asterisk, zaptel
I sent this to the asterisk-dev by accident...
Original Message Follows
Hi all,
I have a couple questions about RTP header compression with Asterisk:
1) Has this been implemented before or is it included in the Asterisk
package?
2) If the answer to (1) is no, is there an RTP stack
On Wed, 20 Aug 2003 13:28:59 -0500 (CDT)
Brian West [EMAIL PROTECTED] wrote:
Comfort Noise and VAD are diffrent things.
bkw
Yeap. But most devices when uses VAD looks out for gaps in speech and replaces
those gaps with comfort noise. :-)
[ ]'s
Eduardo
On Wed, 20 Aug 2003,
For those of you wanting to salvage your Cisco ATA-186 after
inadvertent locking, or after recovering your devices from a vendor
who has locked them, here is a rainy-day project for you:
http://www.sst.com/downloads/datasheet/S71077.pdf
The above document gives exact specifications on the 4mb
Thanks, that's the answer I was looking for.
Do we know if VAD will ever be supported? I know some people don't like VAD
and in my testing, how well VAD works depends on how well it was coded (and
the hardware I suspect). I have seen very good and very bad implementations
of VAD.
Hard to give an
I've had similar problems with X-Lite to X-lite calls.
Sometimes you get a one second burst of audio. After putting the call on
hold and then resuming the call the audio seemed to then work.
I think there are definitely problems.
-Ed
-Original Message-
From: [EMAIL PROTECTED]
Just idly watching * in console mode and saw that someone from
50.49.54.102 tried to register with my *.
whois gives:-
OrgName:Internet Assigned Numbers Authority
OrgID: IANA
Address:4676 Admiralty Way, Suite 330
City: Marina del Rey
StateProv: CA
PostalCode: 90292-6695
Does http://www.voicepulse.com/ work with *?
On Wed, 20 Aug 2003, John Todd wrote:
At 3:20 PM -0500 8/20/03, Mike Ciholas wrote:
Hi all,
While pondering my choices for local dial tone service via a
bunch of POTS lines or a T1, I began to wonder if perhaps there
is another way.
Are
On Wed, 20 Aug 2003, John Todd wrote:
At 3:20 PM -0500 8/20/03, Mike Ciholas wrote:
Are there VoIP dialtone providers? That is, could I use only
my internet connection for voice calls and not have a separate
T1/POTS bank for that?
First question: Does such a thing exist? Where?
Yes.
Any chance of handouts, transcripts, or video being posted to your website
soon?
On Wed, 20 Aug 2003, John Todd wrote:
For those of you that are in the Portland, Oregon area:
I am giving a talk today on Asterisk at the PLUG Advanced Topics
Meeting. Details below.
JT
From: Zot
Are there VoIP dialtone providers? That is, could I use only my
internet connection for voice calls and not have a separate
T1/POTS bank for that?
I guess I am imagining a company that gateways between the PTSN
and the internet backbone. Calls come in and get VoIP'ed and
sent to me
On Wed, 20 Aug 2003, Brian West wrote:
I think NuFone can do what you need contact [EMAIL PROTECTED]
I have inbound 800 service and outbound ld service with them..
works great.
And for local service, you do what?
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS
I guess my question was a little deeper than that.
Can I simply ditch the PTSN?
911 is the sticking point. Most commercial VoIP services
come with the disclaimer that they are *not* a primary
line replacement, precisely because of the liability
issues associated with providing emergency
pipe my local CO line into my * box with an X100P
bkw
On Wed, 20 Aug 2003, Mike Ciholas wrote:
On Wed, 20 Aug 2003, Brian West wrote:
I think NuFone can do what you need contact [EMAIL PROTECTED]
I have inbound 800 service and outbound ld service with them..
works great.
And for
On Wed, 20 Aug 2003, Adam Roach wrote:
I guess my question was a little deeper than that.
Can I simply ditch the PTSN?
911 is the sticking point.
Ah.
Until this tiny, possibly life-or-death detail gets sorted out,
I'm probably going to have at least one traditional phone line
at all
At 04:48 PM 8/20/2003 -0500, you wrote:
Hmm, okay, so would it be possible to maintain *one* POTS line
that is used if anyone dials 911 on their desk phone (set this
up in * dial plan), then it connects to emergency services
properly, and then use a VoIP dial tone provider for *everything*
else?
Is this the Adtran 624 series channel bank?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bartosz Jozwiak
Sent: Wednesday, August 20, 2003 9:55 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port
Channel
On Wed, 20 Aug 2003, Ernest W. Lessenger wrote:
At 04:48 PM 8/20/2003 -0500, you wrote:
Now, if that is possible, how does the VoIP dial tone provider
get my inbound local and toll calls? I would want my local
phone number to work, of course.
You would need to redirect your local
Here are some options:
Digium X100P x 4
US$100 * 4 = US$400
well supported by asterisk
manufacturer supports asterisk developers
Deployed in lots of places with Asterisk
Voicetronix OpenLine4
-
US$500?
Use 1 PCI card
reported working with chan_vpb
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