> The source code found heere http://www.holgerschurig.de/destar.html
> is in an unsupported TAR format.
It isn't. It's tarred and bzip2'd. If your tar can't do this, then you can
resort to this:
bzip2 -d *.tar.bz2 | tar xv
But tar nowaday has the "j" option for bzip2 and the "z" option to
> Is there an application I could use to test this? I.E. like the echo
> test, but doesn't send anything back...
app_record.so ?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
Clive,
Freshtel who provide the Firefly IAX softphone have some IAX hardware
based phones coming out in the next few months.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, 21 July 2004 4:01 PM
To: [EMAIL P
Hi
Out of interest, (this may be not possible) but I think it
would be an excellent idea to modify firmware to handle the
IAX2 protocol. Especially since its a linux based phone.
Thoughts?
Regards
Clive
On Mon, 19 Jul 2004 21:54:59 +
Joshua Colp <[EMAIL PROTECTED]> wrote:
> Hello Yian
Dear Asterisk Group.
I have two Asterisk servers serving two data/help desk
centers, both centers have a near identical setup.
However, when connected to one of my data centers, I call a
user, I can see on the CLI that the phone is ringing, but I hear no ringing on
my SIP soft phone
Mr. Siler -
I respond in kind...
> I am using the latest firmware from the Wiki. 2.4.2 I believe.
Oops. The latest firmware version is 1.2.0
Try http://www.freedomphones.net/polycom/files/ for the latest firmware.
If you don't show the latest version, (try pressing the right buttons
on your Pol
Hello friends,
I got one page from net "http://www.voip-info.org/wiki-CLASS";
In that page I saw lot of *xx codes for asterisk feautres.
I don't know how to use these codes.
If anyone used these codes can you teach me.
Thanks in advance
Regards
Mur
On 21 Jul 2004 at 10:20, Adam Goryachev wrote:
> On Wed, 2004-07-21 at 02:03, Deon Rodden wrote:
> > I installed a server in Australia with a Wildcard X100P in it. From
> > my server in the U.S, I pushed a call via IAX to the server in
> > Australia which then pushed it out that card. Severe echo,
Sorry the misunderstanding.
I am using h323 nufone and faststart is being sent to the gateway, but SIP
times out.
My recent test was to register both ports of the ATA186 as SIP client
directly to * and experiencing the same problem. When trying to call from
one port to another, the call times
Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ
labs softphone, i have the most recent Asterisk version, but when connecting to the
PSTN i have choppy voice problems, not internally just when connecting with my
Mediatrix gateway and ATA, my SJLabs softphone wo
hi!
What's the libr2 status for Asterisk ? I've got R2
E1 delivered to my * box. I have TE410P digium quad card with newest CVS.
How much % is completed with libr2 ?
what's completed ?
& What's missing ?
Thanks,
bit123.
On 20 Jul, 2004, at 21:37, Steven Critchfield wrote:
On Tue, 2004-07-20 at 15:58, Carmi Weinzweig wrote:
Chris -
In the real telephony world, one can buy a DID trunk without buying a
PRI. If one wants more than about 10 trunks (depending on provider),
it
may be cheaper to buy a PRI instead of ind
Install the kernel-source RPM off of the RH9 CD.
-Seth
On Tue, 2004-07-20 at 20:28, Wiley E. Siler wrote:
> The error I receive when I run make
>
> Thanks,
> Wiley
>
>
> -Original Message-
> From: Wiley E. Siler
> Sent: Tuesday, July 20, 2004 4:12 PM
> To: [EMAIL PROTECTED]
> Sub
It would be whatever context you set in zapata.conf, sip.conf, iax.conf,
etc... depending on what protocol you are using.
zapata.conf
signalling=fxo_ks
context=default <- right here
channel=>1
sip.conf
[grandstream1]
type=friend
context=from-sip<- right
On Tue, 20 Jul 2004 11:46:15 -0500, brian <[EMAIL PROTECTED]> wrote:
> Well they fail to realize that ISDN is used for more than data. I just
> wanna scream at them and say "IT DOES VOICE TO YOU NINNY!".. Rates are far
> from reasonable. 167/mth here is what I would have to pay for ISDN-BRI.
>
>
What phones and Interface are you using?
-Shaun
- Original Message -
From: Stephen Hon <[EMAIL PROTECTED]>
Date: Tue, 20 Jul 2004 12:30:29 -1000
Subject: [Asterisk-Users] hold then transfer...
To: [EMAIL PROTECTED]
Hi..
Has anybody been experiencing any problems with transfers us
I am trying to setup a couple of virtual pbx's off of my one may
asterisk box. So far I have been able to segment most everything via
the Dial plan. My only question/problem has to do with the # Transfer
function. I had set up # Transfers prior to segmenting the dial plan,
and I cannot remember
Thanks Jason.
I have spoken to ZyXEL support and they have also confirmed that these
advertised features are not found in the phone. They claim they will
have a new firmware out in the next two or three weeks that will allow
the phone to hold & transfer.
Andrew
_
Andrew
> after plugging it in it says "Configuring IP" - I unlocked it and
> entered the "Network Configuration". I can see the edit-buttons but when
> I trie to press then it says "That key is not active here"
before you hit "network configuration" hit **# to unlock the
config
you may want to read
On Tue, 2004-07-20 at 15:58, Carmi Weinzweig wrote:
> Chris -
> In the real telephony world, one can buy a DID trunk without buying a
> PRI. If one wants more than about 10 trunks (depending on provider), it
> may be cheaper to buy a PRI instead of individual trunks.
>
> Having said that,
Reid A. Forrest wrote (on Jul 20):
> I am having the same problem with voicepulse, since their change announcement
> the other day.
>
> [voicepulse-in]
> context = voicepulse-in
> type = user
> host = gw5.voicepulse.com
> allow=ulaw
Try adding the secret and auth type in this section.
Chris.
Thanks I change it that way and now it's working. Those are not real
secrets. :)
Thanks Again
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Luke
Sent: Tuesday, July 20, 2004 4:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Voice Pulse
On Tuesday 20 July 2004 18:18, George Pajari wrote:
> In spite of what my learned colleague implies above, there is more to
> Canada than Ontario (Bell's territory).
Please retract your statement that I implied anything of the sort; I never
even mentioned the province I was in, nor do I harbour a
CVS 7/16/04 (the latest one I have b4 today) seems to have this problem
too...
Anyone else having problems with the current CVS ignoring calls after about
5 minutes of being up?
I've also noticed that no matter what I set default_expiry to in sip.conf,
it starts at that number and then jumps to 4
According to the Festival website
(http://www.cstr.ed.ac.uk/projects/festival/), a new version of festival
version 2.0 should be showing up soon (end of July). There is even a
Beta version (Festival 1.95) available for download. Supposedly, the new
version has many bug fixes as well as a new HTS an
TN is one of the BEST states for BRI's.. Bellsouth messed up and had to
make some concessions to the PUC a long time ago.. You can get BRI
anywhere, and it's a flat fee.. typically $80-$90 per month Biz rate,
$35-45 Residential..
Thanks, Billy
+---
On Jul 20, 2004, at 5:27 PM, Jay Milk wrote:
I wouldn't mind buying an entire exchange for $0.01/number. What's
that, like $10/month? Heck, I'd even pay $30-$40/month for this.
Where
can I sign up?
Since I'll probably be PRI and DID shopping in the next month or two,
roughly what *do* DIDs run
On Wed, 2004-07-21 at 02:03, Deon Rodden wrote:
> I installed a server in Australia with a Wildcard X100P in it. From my
> server in the U.S, I pushed a call via IAX to the server in Australia which
> then pushed it out that card. Severe echo, only I could hear it though. The
> remote side heard no
I wouldn't mind buying an entire exchange for $0.01/number. What's
that, like $10/month? Heck, I'd even pay $30-$40/month for this. Where
can I sign up?
> -Original Message-
> From: George Pajari [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 20, 2004 5:12 PM
> To: [EMAIL PROTECTED]
>
Tim,
good point, and BRI's steal precious Cu pairs off the aging trunk lines.
PRI costs in our area (northeast USA) were around $600/month a few
years back.
BTW, I found this interesting. Not much about a VoIP service though.
Verizon Announces FTTP Rollout
Verizon has announced that it will be
I have tried both a nul and the following...
Subscribe = 8
callbackmode = contact
Callback = 8
This retrieves my mail through the menu system but not directly.
I am using the latest firmware from the Wiki. 2.4.2 I believe.
I edit my XML docs in notepad only.
Voicemail answers on extension 8.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Steven Critchfield
> Sent: Tuesday, July 20, 2004 4:27 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] BRI dead in USA?
>
> On Tue, 2004-07-20 at 14:15, Scott Stingel wrote:
> > Brian-
>
Could this have to do with the fact that I do not have a copy of the
redhat source code in the palce specified immediately at the top of
Makefile? The writer makes reference to Redhat breaking stuff and that
the headers... Here is is...
# Okay, the people at RedHat have to break everything they
On 01:58 PM 7/20/2004, Carmi Weinzweig wrote:
>Chris -
>In the real telephony world, one can buy a DID trunk without buying a
>PRI. If one wants more than about 10 trunks (depending on provider), it
>may be cheaper to buy a PRI instead of individual trunks.
>
>Having said that, most of these Vo
I am having the same problem with voicepulse, since their change announcement
the other day.
This is what I get on the * console:
Jul 20 18:59:46 NOTICE[1156066224]: chan_iax2.c:5183 socket_read: Rejected
connect attempt from 66.234.228.132
Here are the relevant part of my iax.conf, set accord
I have tried this repeatedly and I get errors and no output. I tried
with the CVS version and the download rfom ftp.digium.com. I have the
output of the make command but it is 109k in text file. Can I post an
email with a zip file or is that not allowed?
Wiley
s
-Original Message-
Fr
Why do you have a non-null msg.mwi.1.subscribe? You're sending a
SUBSCRIBE request to asterisk at extension '8' upon bootup. Is that
what you want?
Did you upgrade the phone with the latest firmware?
Did you use an XML editor to mess with the configuration? I messed up
mine once using a text e
On Jul 20, 2004, at 1:58 PM, Carmi Weinzweig wrote:
They assign me a phone number (a value of $0.01 and $0.10) and let me
receive as many simultaneous calls as my bandwidth allows (using these
numbers every call absorbs a channel that costs between $4.35 and
$43.48).
What I would like is to be
Hi..
Has anybody been experiencing any problems with transfers using
# after holding?
Transfers using the # and music on hold work fine by
themselves. However, when we place somebody on hold we can no longer use the #
to transfer. This is a problem since we use the # button to park
On Tue, 20 Jul 2004, Brian D'Arcy wrote:
> Anyone ever seen anything like this before?
Yes, with a Grandstream over an ADSL (routed) line. I disabled the check,
but the problem stil occured. And since this friends line was almost the
lowest ping in the complete ISP network something else must be
> Here in Canada (Bell Canada) you can get fractional PRI; depending on your
> pricing tier you pay about $53/B channel/mo and then I think $355/mo for
the
> PRI service itself. Considering regular POTS lines are $55/mo there isn't
> really any cost savings at all.
In spite of what my learned col
When I run make I get all kinds of errors. So far I
ahve yet to get past that problem and when I look for /etc/zaptel.conf and
/etc/asterisk/zaptel.com these fiels do not exist.
W
From: Celedonio Albarran
[mailto:[EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 12:57
PMTo: [EMAIL PROTEC
> As an example, they purchase a PRI from either an ILEC or a CLEC for
> between $100 and $1000 (depending on distance and market) giving them
> 23 voice channels and as many numbers as they want (again, numbers cost
> them at most between $0.01 and $0.10).
Pray tell where one can purchase DIDs in
Ok, maybe a _wee_ bit esoteric, but...
I've setup a developemnt system on a Sun Netra T1 running Aurora Linux
(rh-7.3 for sparc64) -- just a few minor Makefile changes in codecs
necessary. All in all, it's running nice. My only problem is lack of
meetme or comprable features. I can't get zapdum
Hi,
I've just (earlier today) updated from CVS so that I can apply the dtmf caller id
patches. Unfortunately this has had an undesired effect.
I have an intertex ix66 which up until the CVS update allowed me to register my *
server with the ix66 for my local domain (eg sip.mydomain.com). Now it
On Tue, 2004-07-20 at 14:15, Scott Stingel wrote:
> Brian-
>
> Wow - that is high!
> I got quoted only $35/month for BRI (and a hefty installation) - not too
> bad. But the comment about "no CLI" scared me off.
In Nashville, it is $90/month for a BRI without per minute charges for
business. I re
On Tue, 2004-07-20 at 13:01, David Goldfein wrote:
> Thanks Everyone!
>
> I appreciate all the feed back.
>
> Right now I am using a Digium T400P card and my system, although it is fast,
> has a slight load, about 15% due to some mysql activity.
>
> I know that Digium as a new card the TE410P.
Chris> In this case, you want to not pay the T1 fee but still
Chris> pay low per number rates.
That is not what he wrote.
And there is a definite market out there for exactly what he
specified: a fixed number of simultaneous calls for a fixed
MRC, plus some (typically larger) block(s) of DIDs fo
You have to compile and install zaptel *before* asterisk for that to
work. You don't have to change your version, just "make install" in
zaptel source directory and then "make clean" & "make install" in
asterisk source directory.
-Seth
On Tue, 2004-07-20 at 13:54, Wiley E. Siler wrote:
> I attemp
Hi
Jul 20 16:59:12 WARNING[1242768448]: app_meetme.c:924 conf_run: Unable to
writey
== Spawn extension (voicepulse-outgoing, 8000, 6) exited non-zero on
'SIP/241'
-- Executing Hangup("SIP/241-f931", "") in new stack
I was in a conference and it just hung up on me. This is the 5 time it di
On Jul 20, 2004, at 11:31 AM, Marty Mastera wrote:
No need for that!...thanks for the info everyone...I'm going to start
keeping my eyes open for a WRT54G for a good deal somewhere!
CompUSA in Seattle had a sale on WRT54G boxes for $59 yesterday.
Scott
__
Chris -
In the real telephony world, one can buy a DID trunk without buying a
PRI. If one wants more than about 10 trunks (depending on provider), it
may be cheaper to buy a PRI instead of individual trunks.
Having said that, most of these VoIP providers have their pricing model
exactly backwa
I am looking for a provider that will provide a
phone number via IAX, IAX2 or SIP using numbers in Barcelona or Tarragona or
even Other City in Spain. May be a 902 number.
Thanks in advance.
On Tuesday 20 July 2004 15:00, Carmi Weinzweig wrote:
>
> Fractional PRI might be something to consider as well, but it really
> depends on how many lines you need and what SBC's gouge rates are these
> days. I'd also check with the CLECs there.
Here in Canada (Bell Canada) you can get fractional
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 19 July 2004 05:09 pm, Steve wrote:
> This occurs every now and then even though there's no one on the phone.
> I sit there by the desk, and all off a sudden the VM light goes off. I pick
> up the phone and I can make calls. No indication on
Doh! Never mind, a brain far on my part.
David
David Filion wrote:
The advanced voice mail options in RC1 is not providing any
functionality except "4 - place outgoing call" and "* - return to mail
menu". Below is a sample line from voice mail.conf:
210 => 234,test1,[EMAIL PROTECTED],,|attach=n
> > they really didn't want to install BRI's. Their
> > comments were "well, BRI is getting quite antiquated"
If they consider ISDN BRI antiquated, what do they think of POTS?
g.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digiu
The advanced voice mail options in RC1 is not providing any
functionality except "4 - place outgoing call" and "* - return to mail
menu". Below is a sample line from voice mail.conf:
210 => 234,test1,[EMAIL PROTECTED],,|attach=no|cid=yes|review=yes
The is a rew install from the tgz file. No pa
You probably did not want to include your secrets in your email!
But
Celedonio Albarran wrote (on Jul 20):
> I have asterisk setup to register with voice pulse. But when I dial
> the DID I get this error message on asterisk:
>
> [voicepulse]
>
> [vpconnect-t01]
You do this twice - an empty
I now have a batch mode installer for GSConfigure which works on at
least 2 of the machines here. If you are having trouble getting the VB
installer version to work, give this a try. It's at
http://www.acsu.buffalo.edu/~sbesch
Stephen R. Besch
___
Ast
Compile zaptel
Edit /etc/zaptel.conf and
/etc/asterisk/zaptel.conf
modprobe zaptel
modprobe wcfxo
ztcfg
start asterisk
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Wiley E. Siler
Sent: Tuesday, July 20, 2004 1:55
PM
To: [EMAIL PROTECTED]
Subject: [As
Hello all,
One of my remote employees is using a 7960 we sent him, on a public IP
address at his home office.
I've run pings and traceroutes both from the server to his phone, and
from the cable modem to our server, there's never a high ping time, or a
dropped packet, however about every 30 minut
I have asterisk setup to register with
voice pulse. But when I dial the DID I get this error message on asterisk:
Jul 20 15:17:54 NOTICE[-276984912]:
chan_iax2.c:5183 socket_read: Rejected connect attempt from 66.234.228.132
This is my iax.conf
disallow=all
allow=G729
What problems are you having? Did you modprobe zaptel,
then wcfxo? Is it configured correctly in /etc/zaptel.conf and
/etc/asterisk/zapata.conf ?
You should not have to scrap your configuration or even
recompile Asterisk. Just load the right modules, run ztcfg and Asterisk should
take it.
Michael Wang wrote:
>Hello,
>
>I have a one-way audio problem. If any one can give me a clue on how to
>solve it, I'd highly appreciate.
>
>My configuration is:
>
>Both Asterisk server and a SIP phone run within a LAN. Asterisk:
>CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build
You want Asterisk to take a call in via SIP and pass it
to the PSTN? Via what hardware?
I may be able to help. E-Mail me via drodden at webunited
dot net
550 Fairway DriveSuite 210Deerfield Beach, FL
33441Online: www.webunited.net
Deon Rodden T
Brian-
Wow - that is high!
I got quoted only $35/month for BRI (and a hefty installation) - not too
bad. But the comment about "no CLI" scared me off.
Regards
Scott
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
URL:www.evtmedia.com
SBC never was overly enthusiastic about BRI circuits; when it was a
popular technolgy 7-10 years ago they priced it too high, and it
languished in limbo since it was too expensive for residential, but
too cheap to interest the corporate sales force, who didn't want to
undercut T1 sales. I helped a
On Tue, 2004-07-20 at 11:16 -0400, Kanuri, Seshu wrote:
> The source code found heere http://www.holgerschurig.de/destar.html is in an
> unsupported TAR format.
>
tar jxvf is all you need.
--
Dave Cotton <[EMAIL PROTECTED]>
___
Asterisk-Users mail
They are definitely giving you the run-around. Call one of the surviving
CLECs. With good negotiation you can get a PRI for <$600/mo. BRIs are
harder to get, they require special hardware on the line side of the switch
and many shelfs don't support them or the advanced features. (They were
sort
I'm having the same problem here. Any solution to this problem?
-Manuel
(sorry for top-posting, I'm having a stupid mail client here)
-Messaggio originale-
Da: Simon Brown [mailto:[EMAIL PROTECTED]
Inviato: giovedì, 1. luglio 2004 02:05
A: [EMAIL PROTECTED]
Oggetto: [Asterisk-Users] Dia
If one is using BRI primarily for voice, POTS lines while they will
work are not a great replacement for many reasons:
No reliable disconnect
Slower call setup times
Slower and less reliable number delivery (CallerID vs. ANI)
Lower voice quality
While no one seems to actually support it, it shoul
To save your current configuration just make sure you don't run "make
samples". Oh, and always back up your /etc/asterisk directory when you
have a config you are happy with.
In order to roll back run "show version" from the CLI and take note of
the CVS version. It will be something like CVS-HEAD-
Have your dhcp server tell it when it picks up the ip address.
What sort of dhcp server are you running?
P
> -Original Message-
> From: xfastjackx [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 20, 2004, 10:57 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] how to configure my ci
At 11:46 AM 7/20/2004, you wrote:
Well they fail to realize that ISDN is used for more than data. I just
wanna scream at them and say "IT DOES VOICE TO YOU NINNY!".. Rates are far
from reasonable. 167/mth here is what I would have to pay for ISDN-BRI.
SBC is lame.
I don't know where you are but w
But I'm not speaking of BRI in a bandwidth/internet sense of the word. I
want it for voice.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Rich Allen
> Sent: Tuesday, July 20, 2004 11:54 AM
> To: [EMAIL PROTECTED]
> Subject: Re
> Eats humble pie!!
>
> I'd never seen it in the settings and sure enough it's there.
>
> Sorry for misguiding.
>
> P
No need for that!...thanks for the info everyone...I'm going to start
keeping my eyes open for a WRT54G for a good deal somewhere!
Thanks again,
Marty
___
brian wrote:
Well they fail to realize that ISDN is used for more than data. I just
wanna scream at them and say "IT DOES VOICE TO YOU NINNY!".. Rates are far
from reasonable. 167/mth here is what I would have to pay for ISDN-BRI.
SBC is lame.
Back in the day, Pacbell was pretty lame also.
I work
On 10:41 AM 7/20/2004, Carmi Weinzweig wrote:
>I want many phone numbers so that each phone in my facility has its own
>phone number, but I really do not need that many simultaneous calls and
>it would be cost prohibitive to pay several dollars for each phone
>number.
It's a different business plan
It does seem to be on its own irq.
Detected Tormenta 2 Quad T1/PRI or E1/PRA at 0xfcf00800/0xfcf0 irq 20
Nothing else seems to be on irq 20.
I don't know enough about the following line to know if it has any impact:
PCI->APIC IRQ transform: (B1,I8,P0) -> 20
Thanks,
Dave
-Original Message
I attempt to run
make clean:make install and I get the following (cut short for
brevity).
zaptel.c: In function `zt_init':zaptel.c:6123:
warning: implicit declaration of function `register_chrdev'zaptel.c:6124:
`KERN_ERR' undeclared (first use in this function)zaptel.c:6124: parse error
b
Scott,
I have an SBC BRI in California. I would not worry about them going away
anytime soon. SBC just does not like to sell them because they want you
to buy a PRI instead. The main thing to worry about is getting that BRI
working with *. There are only a couple of cards that work the National
Ray Burkholder ([EMAIL PROTECTED]) wrote:
> > yet. The only Wireless SIP phone I would use in a productive environment
> > would be the Cisco 7920.
>
> Does it work in SCCP mode with good results in Asterisk?
>
According to my driver modifications, yes. See the asterisk wiki for
further informa
All,
For the last 10 months, I've been using strategy=ringall. This has worked
fine and did what I wanted, but at this point, I'm needing to implement a
'penalty' or delay for some members of the call queue.
1: remote users(remote flunkies)
2: level-1 support (flunkies)
3: level-2
Thanks Everyone!
I appreciate all the feed back.
Right now I am using a Digium T400P card and my system, although it is fast,
has a slight load, about 15% due to some mysql activity.
I know that Digium as a new card the TE410P. Does anyone have any
experience in the new card and is the speed di
Turn off dhcp first. Option 25 in network configuration.
- Original Message -
From: "xfastjackx" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, July 20, 2004 12:34 PM
Subject: *SPAM FOUND* [Asterisk-Users] how to configure my cisco
7960?!
> hi everybody,
>
> just tri
I attempted to
install an X100P card but it was not correctly recognized by my Redhat 9
install. I had a test install running without any cards which was working
great minus the outward dialing since no cards existed. Now that I have a
card, I want to add it to the system. Do I have to sc
I have an Asterisk box with free local termination to area
codes (305) and (786) [Miami area, US]. I want to configure it to accept incomming
VoIP traffic (can’t use IAX) and terminate calls over the PSTN network. I
need help with the configuration and also some incoming traffic for testing
Eats humble pie!!
I'd never seen it in the settings and sure enough it's there.
Sorry for misguiding.
P
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 20, 2004, 7:35 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] 7960 Dynamic D
You are right, there is no SIP firmware for the 7920 - SCCP is currently
the only choice for *.
Ray Burkholder wrote:
yet. The only Wireless SIP phone I would use in a productive environment
would be the Cisco 7920.
I don't see a SIP load for the 7920. Are you sure it is SIP enabled?
Ray.
---
Hello,
I have a one-way audio problem. If any one can give me a clue on how to
solve it, I'd highly appreciate.
My configuration is:
Both Asterisk server and a SIP phone run within a LAN. Asterisk:
CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp
14262. The Linux box tha
On Tuesday 20 July 2004 11:04, Jason A. Pattie wrote:
> I normally wait about a second after I pick up the phone until I hear a
> very small click. I think that might be the end of the training period.
> ~ Then I proceed with my introduction. It seems to work quite well.
I agree and do that myse
Never rely on a telco for correct information, they will very often be
wrong... unless you luck out and actually talk to someone who knows
something...
Both PRI and BRI are capable of ANI (Caller ID) by using their D-Channel to
send/receive this information digitally...
A regular T1 (read non-ISD
Hello,
I am receiving the following repeated Errors and Warnings with
Galaxyvoice. I have placed the sip context below, perhaps someone can
offer suggestions how I could troubleshoot this. Thanks
Kevin
Jul 20 12:35:48 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum
retries exceeded on
I tried this configuration and it still does not work for me. In fact,
now I cannot dial in using the menu system of the message center. Here
is how I have now mine configured and what I get...
The relevent fields being the msg. fields and up.oneTo
i work for a local telco and BRI is avoided due to the amount of
hardware it can take to get to an end user. DSL is simply easier and
cheaper to provide. Not sure why you can't get caller id, i know when
can add that feature to BRI
- hcir
On Jul 20, 2004, at 7:36 AM, Scott Stingel wrote:
Hi-
Be
You can have echo even w/ PRI's, it's just that the echo isn't
INTRODUCED at the PRI demarcation point, it is INTRODUCED somewhere
along the call path, usually where the 4 wire digital signal to 2 wire
analogue signal conversion point exist. We found that moving to a
Compaq DL380 or 6400R and comp
BTW - the title of this was supposed to be "BRI dead in USA?"! (too early!)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Stingel
Sent: Tuesday, July 20, 2004 8:37 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PRI dead in USA?
Hi-
Because
Hi Scott,
Local ISDN BRI service is definitely on it's way out.
We recently have canceled several ISDN BRI accounts
and replaced them with ADSL lines. More bandwidth and
less cost. If you intend on using the lines for voice only, then
FXO is the better option. If you looking to use voice&data the
Well they fail to realize that ISDN is used for more than data. I just
wanna scream at them and say "IT DOES VOICE TO YOU NINNY!".. Rates are far
from reasonable. 167/mth here is what I would have to pay for ISDN-BRI.
SBC is lame.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mai
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