Hi everyone,
THis is my second thread regarding the issue.(before I was having problems
with accessing my email, which slow down my responses, sorry for that)
My setup looks like this
Firewall
|
|
Asterisk---Asterisk (two asterisk servers with the same setup for high
avail)
|
|
phones
Ports
I don't know what is wrong with the Broadvoice, but for me everything
works fine. I used the setup they provided on their website.
It works fine and with no problems.
To make sure that all incoming calls will never miss my box I added those
lines in sip.conf. For me it works fine.
On 21/02/2005 11:41 david said the following:
Hello,every one,
I have recorded the voice files with mandarin (China). Where should
I contrib the files ?
you could host it on a web server, and then modify the wiki page at
http://www.voip-info.org/wiki-Asterisk+sound+files+international to
In article [EMAIL PROTECTED],
Nigel Taylor [EMAIL PROTECTED] wrote:
Can anyone recommend a Digium Reseller in the UK ?
TelAppliant
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
On Sat, 5 Mar 2005, BJ Weschke wrote:
Asterisk has the ability to do agent queueing and some general ACD
functionality. The functionality doesn't come close to the
functionality/flexibility of Avaya's Expert Agent functionality, but *
won't cost you several hundred thousand dollars for
sorry still doesnt help with incoming calls, there is definatley
something more wrong, my config was working fine until today and its
worked fine for months. They have broken something.
On Sun, 2005-03-06 at 02:23 -0600, Bartosz Wegrzyn - asterisk wrote:
[broadvoice-incoming]
type=peer
David J Carter wrote:
Nigel,
Should really be on the biz list for this, but Telappliant sells Digium
hardware.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nigel
Taylor
Sent: 05 March 2005 21:30
To: Asterisk Users Mailing List - Non-Commercial
Thanks. Adding those lines appears to have fixed the problem. I'll
just hold on til the NEXT TIME Broadvoice decides to make a change.
Thanks again.
Bartosz Wegrzyn - asterisk wrote:
I don't know what is wrong with the Broadvoice, but for me everything
works fine. I used the setup they
I guess it is a chan_sccp bug, it's sccp_sched reporting it.
Mar 6 12:06:03 WARNING[352]: sccp_sched.c:65 sccp_sched_keepalive: Dead
SCCP client: SEP
Is chan_sccp still alive, is there a developers list or anything? The
sourceforge page mentions a new release in January 2005 but the last
I am interested in using the M(x) option on the Dial command to run a macro
upon connection of a call.
I am using the lastest stable release. The wiki indicates that improvements
have been made for the 1.1 version (sending parameters delimited with ^).
Does M(x) work at all with the current
On Sat, Mar 05, 2005 at 03:57:07PM -0600, Blake Van Eekeren said:
Fredrik wrote:
I see from my CDR's that some of my callers also have unknown in
their FROM field. I would like to let them through. Only block the
FROM anonymous that the telemarketers use.
Fredrik, I found something on
I try to setup SJphone on my PDA, but it does not register with Asterisk.
I have setup a sip account on asterisk, ...
Can anybody give me a hint?
bye
Ronald
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Debugging lack of registration:
Watch the console (set verbose 255) and see if there are registration
attempts.
If you see failures, the name and secret are probably wrong.
If you don't see attempts, either the phone isn't trying, or there is a
connectivity problem from the phone to the
Ronald,
You will need to give *more* information than that
I have SJphone on my PDA, and have setup a SIP account on *, and it works
fine :-)
I take it you have setup sjphone to register to *.
I take it your PDA has a network connection?
C
-Original Message-
From: [EMAIL
Hold, transfer and flash only!
the conference key is only for model 102-D
Bill Michaelson escreveu:
Is it possible to use the Hold/Transfer/Conference/Flash keys of the
Budgetone-101 (FW 1.0.5.22) with Asterisk?
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C. Tomlinson wrote:
Ronald,
You will need to give *more* information than that
I have SJphone on my PDA, and have setup a SIP account on *, and it works
fine :-)
I take it you have setup sjphone to register to *.
I take it your PDA has a network connection?
I have setup a sip account at
Hello. New to the list. We're in the process of deploying Asterisk.
Actually, we're going live tomorrow, and I just found out that my Zaptel
cards have been mis-configured.
I'll preface this by saying that I have looked in the wiki, read through
the samples, and attempted to call Digium
Here is a sample of our zaptel.conf config as it was handed to me (I
inherited this Asterisk project, btw). These configs are likely a train
wreck, so if anybody could possible either generate a config that would
work, or explain a somewhat laymens terms how I can go about making a
good
Hi Chris,
No such luck. When I was cut pasting the config files into the email,
I accidentally deleted the hashmark that was before fxoks=1-12 so that
option was never loading.
I am at my wits end now! :)
Chris Modesitt wrote:
Remove fxoks=1-24, In the setup you described you want to use the
Ben Ruset wrote:
I have two Digium cards. One is a TE405P quad T1 card. The other is a
TDM40B (I believe) quad analog POTS card.
Our provider has been telling us that they are only seeing one D channel
active. This would make sense if somehow only the first T1 in the 405P
was activated.
maybe
In the last email I sent, I did not mean to insult anyone, but I have
tested the instructions thoroughly I provided. If you were using the
instructions I provided originally, you would not be able to make outbound
calls. Here are the instructions that have been known to work;
Please read line
Where do you have setup for incoming calls to go?
To an extension ?
To [context] s extension?
sip show registry does it show that you are registered with BV?
The first 3 questions have to do with Asterisk.
If you have them to go to an extension and that extension has DND or it is not
Hi Frank,
I've changed the timing on both spans. That unfortunately has not solved
the problem.
I have also removed all of the entries in zaptel.conf and zapata.conf
for the analog card, as well as prevented the module from being loaded
at boot. So now, as far as the machine knows, it only has
I'm all so using SJphone on my x50v, works surprisingly well :).
Is voip.elmit.com also in the 192.168.1.X NAT space that your PDA is in?
Do you have host=dynamic in your * sip.conf entry for 701 ? Actually might
help to toss your sip.conf entry out here for 701 without the secret.
Do you see
Ben Ruset wrote:
I have two Digium cards. One is a TE405P quad T1 card. The other is a
TDM40B (I believe) quad analog POTS card.
We have two T1's. Both of them are split in half (half voice, half data.
- Don't ask me, that's how I inherited them.) Voice traffic flows on the
back 12 channels of
Hi all,
i am new to this list and i dot not know, if anybody had already the
same problem. I have two cisco 7960 which i want to upgrade to sip. Has
somebody already taken the upgrade-process for special hints and
suggestions? I have already visited the cisco-page and i have read the
proposal
Read the wiki, if you can't located enter the following in the google
search box:
cisco site:voip-info.org
This should bring something up.
On Sun, 06 Mar 2005 20:03:52 +0100, Thomas Trepper
[EMAIL PROTECTED] wrote:
Hi all,
i am new to this list and i dot not know, if anybody had already the
George Pajari wrote:
It appears
that every so often the Sayson does not send out another REGISTER
message after the registration has expired resulting in the reverse
mapping being closed and the phone made unreachable.
Even behind regular home Linksys router that doesn't close the mapping
the
Sounds like you are having a codec issue with 2 of your providers. Make
sure you find out what codecs are supported and that your config is set up
accordingly.
Thanks :)
I don't think that is it though as I have tried with other codecs initially
and inbound calls work fine regardless.
My
I am trying to run a macro at the beginning of call and after the call is
terminated.
exten = 33,1,Macro(makeOnJS,${EXTEN},${CALLERIDNUM},${DATETIME})
exten = 33,2,Dial(SIP/33,15,tg)
exten = 33,3,NoOp(makeOffJS*${EXTEN}*${CALLERIDNUM}*${DATETIME})
exten =
On Asterisk 1.0 with a 4-port Digium FXO card, voicemails from the PSTN
have volume so low they often can't be heard. Worse, callers sometimes get
cut off in the middle of leaving a message. It is extremely frustrating to
hear ...and my number is...END OF MESSAGE
A search of the archives shows
Hi, I'm a newbie on * trying to setup an HFC card.
I'm locked for many days getting the all-circuits-busy. And no idea what
else to look for/how to diagnose.
I'm in Spain, I've tried changing many parameters on zapata/zaptelcong
with no luck, also NT TE modes (honsetly, I've no idea what is).
Hello All ~
Having problems sending and receiving faxes with SpanDSP. I am testing on a
simple 2 analog POTS to 2x X100p set up, connecting one line to a Konica 720
fax machine to test, or with other remote fax machines. Voice calls are
working pretty well now. Platform is P3/800MHz/256MB/FC1.
*
I have FXO (DIGIUM) with Asterisk (PBX). How can I use SNMP in Asterisk
to access FXO?
I need to known if FXO has the LINE with PSTN free to new phone call. Is
this possible? How?
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Hello,
I don't know if your zaptel.conf and zapata.conf setup regarding your
isdn is correct, but if you use the default AMP setup, you need to
assign your channels to group 0 for dialing out, and assign it to
context from-pstn if you want to receive calls.
group = 0
context=from-pstn
channel =
On Asterisk 1.0 with a 4-port Digium FXO card, voicemails
from the PSTN have volume so low they often can't be heard.
Worse, callers sometimes get cut off in the middle of leaving
a message. It is extremely frustrating to hear ...and my
number is...END OF MESSAGE
A search of the
Hey Julian, thanks! It really make a difference. Thanks for pointing me
to this. Stupid newbie mistake.
Yes, I'm using AMP, it was bundled with [EMAIL PROTECTED]
Now I'm not longer getting the all-the-circuits-are-busy-now, but still
doesn't dial out, now I'm getting the congestion tone.
Maybe
On Sun, 6 Mar 2005, Marty Mastera wrote:
The full text of the bug you reference above indicates that pstnVMgain
was (or is) part of an ongoing feature request/bug report and has not
been implemented for use at this time (and may never be).
Right. So -- what can I do to boost volume of PSTN - *
The full text of the bug you reference above indicates that
pstnVMgain
was (or is) part of an ongoing feature request/bug report
and has not
been implemented for use at this time (and may never be).
Right. So -- what can I do to boost volume of PSTN - * voicemail?
thanks
dn
Thomas,
The definitive guide of what versions can be upgraded to what is at:
http://www.cisco.com/en/US/products/sw/voicesw/ps4967/products_upgrade_guides09186a008022a968.html
In particular, look at tables 2 and 3.
Alistair Cunningham,
Integrics Ltd,
Telephony, Database, Unix consulting worldwide
Im new to Astererisk. I compiled the latest CVS and setup the server. It
looks like things are working. I'm running kphone, x-lite and sjphone to
test things out. The kphone (local to the asterisk server) can call and
receive calls from any of the 2 windows machines. The first windows phone
I
i am new to this list and i dot not know, if anybody had already the
same problem. I have two cisco 7960 which i want to upgrade to sip. Has
somebody already taken the upgrade-process for special hints and
suggestions? I have already visited the cisco-page and i have read the
It would be helpful if you pasted the relevant sections of sip.conf and
extensions.conf
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Im new to Astererisk. I compiled the latest CVS and setup the server. It
looks like things are working. I'm running
Hi,
Whilst I agree with Joe, has anybody actually been able to sucessfuly get the
Cisco 7940's/7960's to register into *?
We have just about tried everything that was suggested to us without luck.
Cheers,
Sahil
Quoting Joe Greco [EMAIL PROTECTED]:
i am new to this list and i dot not know, if
Whilst I agree with Joe, has anybody actually been able to
sucessfuly get the Cisco 7940's/7960's to register into *?
Yeah, I've been using a 7960 with * since November.
Nabeel
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Nabeel Jafferali wrote:
Whilst I agree with Joe, has anybody actually been able to
sucessfuly get the Cisco 7940's/7960's to register into *?
Yeah, I've been using a 7960 with * since November.
Indeed, I have 7905/7940/7960's all working with *.
___
Hi,
Whilst I agree with Joe, has anybody actually been able to sucessfuly get the
Cisco 7940's/7960's to register into *?
We have just about tried everything that was suggested to us without luck.
Um, well, really, that's never been a problem here. I've had more problems
trying to get them
On Mon, 2005-03-07 at 09:02, David Newman wrote:
On Sun, 6 Mar 2005, Marty Mastera wrote:
The full text of the bug you reference above indicates that pstnVMgain
was (or is) part of an ongoing feature request/bug report and has not
been implemented for use at this time (and may never be).
Any assistance on gettting bi-directional calling going would be great...
We got it working in SIP but it won't register hence calls going to the phones
don't even start..
Quoting Nabeel Jafferali [EMAIL PROTECTED]:
Whilst I agree with Joe, has anybody actually been able to
sucessfuly get
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
[EMAIL PROTECTED] wrote:
| Hi,
| Whilst I agree with Joe, has anybody actually been able to sucessfuly
get the
| Cisco 7940's/7960's to register into *?
|
| We have just about tried everything that was suggested to us without luck.
|
Yes, working
Thanks as always to everyone who provided feedback. It was most helpful!
Regards,
Jim.
--
Jim Van Meggelen
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I would like to start a discussion centred around the various
ways one might serve up configuration files from an Asterisk
server (I know,
Hey guys
Anybody knows how to turn up the volume of a Music on Hold Mp3 file ?
When I play it on my windows box, volume is perfect.. but when I use it
Music on hold.. the volume is very low.
Maybe there is a general setting for asterisk volume ?
Thx for the help
Matt
I've just intalled a system with 25 Cisco 7960s worked perfectly,
please tell what you did wrong, maybe I can help you.
On Mon, 7 Mar 2005 11:41:38 +1300, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
Whilst I agree with Joe, has anybody actually been able to sucessfuly get the
Cisco
Works sweet here with a 7960G too, 7.3 SIP fware.
Mike
On Mon, 7 Mar 2005 11:41:38 +1300, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
Whilst I agree with Joe, has anybody actually been able to sucessfuly get the
Cisco 7940's/7960's to register into *?
We have just about tried
[EMAIL PROTECTED] wrote:
Any assistance on gettting bi-directional calling going would be great...
We got it working in SIP but it won't register hence calls going to the phones
don't even start..
If the phones are behind a NAT, make sure the option on the phone for
NAT is set to yes.
Doug
Makes you wonder how many *really* reliable VoIP providers there are out there?
Who would you trust to handle all your incoming/outgoing business calls?
Mike
On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote:
Anyone having problems with LiveVoIP lately? I am seeing failed outgoing
Hello all! I googled lists.digium.com and ser mailing list, but did
not find any working configuration of asterisk used as voicemail for
SER. This is my config
if (uri==myself) {
if (method==REGISTER) {
save(location);
log
the kphone is using 214 and the windows 204 and 203. It doesnt matter
though I can have kphone use 203 and windows 214,204 and get the same
issues.
sip.conf:
[214]
type=friend
username=214
secret=214
callerid=test 214
nat=no
canreinvite=yes
disallow=all
allow=gsm
allow=ulaw
allow=alaw
Joe Greco wrote:
The definitive guide of what versions can be upgraded to what is at:
http://www.cisco.com/en/US/products/sw/voicesw/ps4967/products_upgrade_guides09186a008022a968.html
In particular, look at tables 2 and 3.
Horrible answer.
Better:
1) Take ANY Cisco documentation with a ton of
Mike Dent wrote:
Makes you wonder how many *really* reliable VoIP providers there are out there?
Who would you trust to handle all your incoming/outgoing business calls?
None.
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If I use rewritehostport instead of forward, the call does not reach asterisk:
failure_route[1] {
revert_uri();
rewritehostport(69.70.x.x:5060);
t_relay()
break();
SER log:
Your failure route should read:
failure_route[1] {
revert_uri();
Are there any IP Providers that will pass Caller ID? Broadvoice used to
but no they dont.
THX
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To UNSUBSCRIBE or update options
receive calls from any of the 2 windows machines. The first windows phone
I start I can send/receve calls the second one I cannot. I. No matter
which one I start first only the first one works. The linux kphone can
Please take note that each phone need it's account. You can't have 2
phone
Mike,
No they have not. Calls are failing again today. They have offered to
refund my money but that does not solve the problem. My asterisk server
is only 4 to 12 ms away from their network. I have had VERY good luck
with nufone.(40 to 45ms away) Only have 1 or 2% fail rate. Going to be
calling
Date: Sun, 06 Mar 2005 20:03:52 +0100
From: Thomas Trepper [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 7960
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii; format=flowed
Hi all,
i am new to this list and i dot not
I am based in New Zealand and
am experiencing the same problem as referred to in the post "FXO module in
TDM400P (UK, BT) - Hangup detection failing" from 2 November 2004 i.e. Zap/4
(being the FXO module) not detecting hangup on the PSTN line if the call is not
answered on a PABX
Hi,
See comments in line
configured at mem 0xf91c5e00 fifo 0xf7598000(0x37598000) IRQ 5 HZ 100
Mar 6 21:29:13 linux-1 kernel: zaphfc: Card 0 configured for NT mode
Mar 6 21:29:13 linux-1 kernel: zaphfc: 1 hfc-pci card(s) in this box.
Your card is configured in NT mode this something you do
So you want it 100% perfect and you want it for peanuts.
Makes you wonder how many *really* reliable VoIP providers there are out
there?
Who would you trust to handle all your incoming/outgoing business calls?
Mike
On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote:
Anyone having
On Mon, 7 Mar 2005, Peter Illmayer wrote:
You will need to load the version 3, then 5 and then 7 SIP firmware. I tried
to load the version 7 straight away and of course it wouldnt work.
FWIW, I have also had success doing versions 3, 6, and then 7 in moving
from Skinny to SIP. But it's still
No. When DID go down for a whole day. Do you think thats okay? Ring
busy half time or do nothing at all. Come on! Your DID's are up maybe
50% of the time if that!
Why are calls failing again today?
On Sun, 2005-03-06 at 17:36, The Phone Guys wrote:
So you want it 100% perfect and you want
No, I dont mind paying more for something if I know its going to be reliable.
On Sun, 6 Mar 2005 16:36:16 -0700, The Phone Guys [EMAIL PROTECTED] wrote:
So you want it 100% perfect and you want it for peanuts.
Makes you wonder how many *really* reliable VoIP providers there are out
there?
On Mon, 7 Mar 2005, Howard Lowndes wrote:
On Mon, 2005-03-07 at 09:02, David Newman wrote:
On Sun, 6 Mar 2005, Marty Mastera wrote:
The full text of the bug you reference above indicates that pstnVMgain
was (or is) part of an ongoing feature request/bug report and has not
been implemented for use
The Phone Guys wrote:
So you want it 100% perfect and you want it for peanuts.
OF COURSE!
They all certainly imply and promise that.
Would anyone subscribe if they said we have a second rate service ?
Makes you wonder how many *really* reliable VoIP providers there are
out there?
Who would
What do folks have to say about www.voipjet.com?
(IAX, call termination only)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Novack
Sent: Montag, 7. März 2005 00:58
To: The Phone Guys; Asterisk Users Mailing List - Non-Commercial
Discussion
Looks like if you complain. They terminate your account. Or is this just
another BUG
Mar 6 17:57:56 NOTICE[18108]: chan_iax2.c:6695 socket_read:
Registration of 'wdsmn' rejected: Registration Refused
Mar 6 17:57:56 NOTICE[18108]: chan_iax2.c:6695 socket_read:
Registration of 'tschacher'
Maybe at some stage in the future the big telcos will provide VoIP
termination, DID's etc. They may as well make some money from it, I'm
sure they could get it right?
BT providing IAX2 and SIP termination? Hmmm, maybe one day.
Mike
___
Asterisk-Users
Original Message
Subject:
Re: [Asterisk-Users] BroadVoice configuration changes for
Outbound
Date:
Sun, 06 Mar 2005 19:11:22 -0500
From:
MF Hulber [EMAIL PROTECTED]
To:
Asterisk Users
CClarke wrote:
Hello All ~
Having problems sending and receiving faxes with SpanDSP. I am testing on a
simple 2 analog POTS to 2x X100p set up, connecting one line to a Konica 720
fax machine to test, or with other remote fax machines. Voice calls are
working pretty well now. Platform is
each phone logs in under its own sip
account: 203, 204 and 214. I assume the account is whats in the sip.conf
file.
Time Bandit [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
03/06/2005 06:22 PM
Please respond to
Time Bandit [EMAIL PROTECTED]; Please respond to
Asterisk Users Mailing List
Hello. I'm not sure what's going on with the gentleman who is having
trouble receiving inbound calls as of this weekend, but I can say that
while inbound works for me, calling out through BroadVoice doesn't work at
all. SIP traces show that when I send an invite request out to
Check out musiconhold.conf
you can use loud
On Mon, 7 Mar 2005 00:02:27 +0100, Mateo Meier [EMAIL PROTECTED] wrote:
Hey guys
Anybody knows how to turn up the volume of a Music on Hold Mp3 file ?
When I play it on my windows box, volume is perfect.. but when I use it
Music on hold.. the
All fixed. I just updated from CVS, rebuild and everything works. I did
try restarting astrisks before I tried this so it either didnt pick up a
config right or the new CVS fixed it.
[EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
03/06/2005 07:24 PM
Please respond to
Asterisk Users Mailing
*shrug*. Mine's been working flawlessly since I've had it (~month). The
only 2 issues I have are the ringback problem, and I can only send callerid
number info to them, not name info Guess we'll see how long it
lasts
regards,
Paul
- Original Message -
From: Tim [EMAIL
No, I want it to work 50% of the time and pay half your current pricing.
Or maybe we can make this really easy for you to understand. Make it
work 0% of the time and we pay you nothing.
I think that people expect it to work about 99.99% of the time if they
are going to use it for production
That's surprising; I thought they were one of the larger outfits. I have
tried quite a few for outbound lately and the only one that has reliably
passed Caller ID (using our 406 area code) is simpletelecom.com. They are
also the only ones I've tried that respond to a support ticket in a
I just got off the phone with LiveVoIP. They have address most if not
all of my current issues. These guys are BIG players in the VoIP biz. A
long with that comes big problems. They are working on the issues.
Let's just give them a break this time around.
On Sun, 2005-03-06 at 19:07, Paul
I have about 10 DIDs, I had an issue that lasted a day or so that was Level
3's issue, it took about 12 seconds before the calls would come in. That
was resolved and I haven't had any issues at all. I appreciate the fact
that there are reselling Level 3 DIDs since they seem to be in a lot of
No downtime yet. Also good experience with simpletelecom -- for IAX
termination, there's really no serious cost in using multiple accounts,
except for having to check your balances every so often. Get two or
three, line them up nicely in your dial-plan, and if one fails, go
through the other.
can u send your config and simple description of your network
Bart
sorry still doesnt help with incoming calls, there is definatley
something more wrong, my config was working fine until today and its
worked fine for months. They have broken something.
On Sun, 2005-03-06 at 02:23 -0600,
I also have txlink.net. They have been very solid and very good to work
with. I had a toll free ported that took a long time to do. It wasn't
their issue, it was the original company that was being difficult. I had 3
more ported, took 2 days, again great to work with.
I agree with Jay, it's
You won't be able to send caller-id NAME with any PSTN termination.
That's just not how that works. Each CLEC looks up the name in some
mystical database based on the phone number. How to get that DB, I
don't know, but it sure would be nice to integrate something like this
into *, wouldn't it?
Near 100% for a resonable price that you have set and
the ability for both the provider and consumer to
understand how to work together and make sure that
companies that providers are buying their service from
understands the impact of what out of services means
to the consumers.
I like the idea
Hi all,
How is it possible to do loop with * ?
I want to redirect ALL calls initiate by a SIP channel on itself without
'treatment' by muy * box.
Regards.
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I have used them for 6 months with few issues. Good rates as well.
Michael
On Mon, 7 Mar 2005 01:05:01 +0100, Roman Zhovtulya wrote:
What do folks have to say about www.voipjet.com?
(IAX, call termination only)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Jay Milk wrote:
You won't be able to send caller-id NAME with any PSTN termination.
That's just not how that works. Each CLEC looks up the name in some
mystical database based on the phone number.
And pays the keeper of the database for each lookup.
Also, more than one database exists.
How to
On Sun, 6 Mar 2005 20:22:48 -0500
Steven Frazier [EMAIL PROTECTED] wrote:
I have about 10 DIDs, I had an issue that lasted a day
or so that was Level
3's issue, it took about 12 seconds before the calls
would come in. That
was resolved and I haven't had any issues at all. I
appreciate the
No, I dont mind paying more for something if I know its going to be reliable.
Well, now, that's kind of the problem here, isn't it?
If VoIP pricing isn't more attractive than LEC line pricing, the slam dunk
choice is to go with the traditional LEC service. It's reliable, it's
cheap, and it's
Jay Milk wrote:
You won't be able to send caller-id NAME with any PSTN termination.
That's just not how that works. Each CLEC looks up the name in some
mystical database based on the phone number. How to get that DB, I
don't know, but it sure would be nice to integrate something like this
into
James Pooton wrote:
I'm all so using SJphone on my x50v, works surprisingly well :).
Is voip.elmit.com also in the 192.168.1.X NAT space that your PDA is in?
There might be the problem:
I have the server at two ethernet cards reachable:
Extern with a public IP
Intern with 192.168.250.20
on
I've fought this all weekend.
Friday, they couldn't take an order because the credit card thing on the
website
was broken.
Saturday, I got an account.
Incoming works, put the phonenumber at the end of the register string
and then place that number as an extension in your broadvoice context.
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