we also use the 220 but with the additional button panel with great
results!
On Tue, 2004-12-14 at 07:25 -0600, Gerald J. Puhl wrote:
Does this phone have LEDs showing lines in-use?
Thanx!
Gary P.
Tracy R Reed wrote:
On Mon, Dec 13, 2004 at 12:50:54PM -0600, Gerald J. Puhl spake thusly:
Nicolas,
Thank you for your response. I had tried that before and it didn't work. I
am trying to look up the route for a dialed number, so its a full E.164
number. Please see my query below when I try to look up the route for a USA
number;
mysql SELECT * FROM routes WHERE ^13237309880 RLIKE
On Tue, 2004-12-14 at 18:44 +, Jean-Michel Hiver wrote:
To the list: Am I right understanding that Fritz + BRI line = no echo
issues?
I have two systems using AVM cards, one uses a C2 and the other 2 Fritz
cards neither have any problems of echo.
--
Dave Cotton [EMAIL PROTECTED]
Oh, friend... I have realised just yesterday that's impossible to use
regular modems (say hayes/v90 33.6 or 56k) to plug asterisk to the world. I
can't figure out why. But they simply don't support it.
If you want to use your isdn modem to plug * to the world, it's OK.
Else... Only just one
Hi
Im having trouble getting the caller id displayed on the GS BT101
phone. I am aware that it cant display alpha characters. In my set up all the
calls are being going thru * except media (ie canreinvite=yes). Here is the
trace of the INVITE message that is received by the phone and I am
Hi
Try the Voicetronix Openline 4, we found these to be the best
so far!
Cheers
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dorn Hetzel
Sent: Tuesday, December 14, 2004 5:48 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] least sucky FXO interface?
Hi List!
Does anyone have experience setting up this beast with Asterisk using
Skiny/SCCP?
I know that somebody on the list managed to get it working with H32H but
in this setup it's not possible to transfer calls, quite annoying.
I tried to set it up using the Cisco protol. With Skinny the
I'm thinking in sending a mail for asking WHY THE HELL they can't support
bare modems, even if they have voice support
Smart move, on their part:
1. Digium exists to sell hardware. Without hardware sales, formal Asterisk
development would stall, and the project as we know it would fragment.
Fabrício Zimmerer Murta wrote:
Oh, friend... I have realised just yesterday that's impossible to use
regular modems (say hayes/v90 33.6 or 56k) to plug asterisk to the world. I
can't figure out why. But they simply don't support it.
If you want to use your isdn modem to plug * to the world, it's
I'm thinking in sending a mail for asking WHY THE HELL they can't support
bare modems, even if they have voice support
As someone else mentioned, modems suck for voice. It seems that the
people that want to use modems with Asterisk are the same people that do
not have the training and
As far as I can remember I only opened sip and tftp ports for the phone.
For some reason (didn't look into it too much) the call stays with sip
and doesn't use RTP.
The problem you describe (the call doesn't even ring on the other side)
is something I had and was solved by upgrading the
It would be nice to see a WIKI page on ISDN/PSTN-CISCO-Asterisk.
drool!
i am having voice quality issues when sipping out over a 7650 fxo
to pstn. i sounds a bit like too much silence suppression.
randy
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Hello all,
I have a uip200 for testing and I can't seem to get the phone to
register to my * server. I have configured the unidencomm.txt and the
unidenMACOFPHONE.txt files and the phone tries to register but * comes
back with a 403 Forbidden message in sip debug, the phone simply
Hello!
I have a special problem with a zaptel PRI card (E100P) with Asterisk
STABLE CVS (zaptel and libpri also from stable branch). I use iaxComm
softphone for outgoing calls (ulaw currently). Calls are going out
through PRI. After about 5-6 calls, when the 7th call arrives, the
voice volume on
Does anyone know how to use the cisco router with an fxo wic with
Asterisk? I don't have enough space on this device to support an IOS
that supports sip or h323. Currently the only one signaling in there
says Cisco. I assume this is the skinny protocol.
www.kingston.com has inexpensive DRAM and
Dorn Hetzel wrote:
Would anyone care to offer opinions as to the FXO interface which sucks
the least :)
So far, for me, using VoIP - PSTN termination provider has been the
solution which sucked the least.
My FXO card doesn't seem to work so well. Never tried my SIPURA as an
FXO device though...
My users are reporting that some voicemail messages are being cut off in
the middle of being played back. The recordings are OK (they play fine
when forwarded to e-mail, and they can often be accessed OK during a
later call to voicemail). I found nothing in the archives on this --
ideas anyone?
Hi,
I'm currently using a ISDN-BRI with a Fritz ISDN card and the
chan-capi. The problem is that the fax detection is not executed, here is a
snippet of the extensions.conf:
[from-capi]
exten = h,1,Macro(record-cleanup)
exten = s,1,Wait(2)
exten = s,2,ResponseTimeout(15)
exten =
Why not try a show uptime to see if it restarted.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Asterisk
Sent: Tuesday, December 14, 2004 9:54 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
On Tue, Dec 14, 2004 at 11:20:08AM -0500, VoIPCarib wrote:
mysql SELECT * FROM routes WHERE ^13237309880 RLIKE pattern ORDER BY
LENGTH(pattern) DESC;
It shows a few other routes before the USA, but ASTCC is written to use the
first row in the result table, which in this case wiould be
I've configured an analog/single line off my mitel sx-2000 to a x100p
card in my *, however when the remote caller drops the line I get a
dialtone back from my pbx and the x100p doesn't detect the end of the
call. Any way I can tell * to drop? I've tried call progress but it
doesn't seem to
The hardware I currently have is:
TDM400P with 3 FXO ports, and 1 FXS port
4 Cisco 7960 Phones (only 1 is currently configured for testing purposes)
Asterisk on slack 10
I can dial out just fine via the Cisco phone, but when I try to dail
in I get the following output when I load asterisk up
Christopher Dobbs [EMAIL PROTECTED] wrote:
My company has started development on a Ethernet based channel bank.
Here are the (current) spec's
- 10/100 Ethernet Port
- Up to 96 FXS/FXO ports (Thats 4 DS1's for the math impaired)
- Serial Console
- TDMoE
- IAX2
-
I am shopping around for an IAX provider that provides both
outbound minutes and an inbound DID for a flat fee. I have been looking at TELIAX.
http://teliax.com/
Does anyone have any experience with them? What codecs do
they support? Inband or 2833 dtmf? Voice quality and
And T1s too. If you can supply a seperate piece of hardware that can handle
all the T1 crap then pass calls to asterisk as SIP/IAX, that would be
awesome in our situation.
Right now our only solution is 8 T1s into a 5300, then SIP to asterisk.
-Matthew
- Original Message -
From: Marc
On Tue, Dec 14, 2004 at 05:20:02PM +0100, Dave Cotton wrote:
On Tue, 2004-12-14 at 18:44 +, Jean-Michel Hiver wrote:
To the list: Am I right understanding that Fritz + BRI line = no echo
issues?
I have two systems using AVM cards, one uses a C2 and the other 2 Fritz
cards neither
Hello,
Thank you for your response. I had tried that before and it didn't work. I
am trying to look up the route for a dialed number, so its a full E.164
number. Please see my query below when I try to look up the route for a USA
number;
mysql SELECT * FROM routes WHERE ^13237309880 RLIKE
Asterisk wrote:
Asterisk -r connects straight back to the cli.
Is there a new timeout (I'm running CVS head as of last night) or is there
something more seriously amiss with this ?
I would expect that for some reason, your copy of Asterisk is:
A) Being restarted automatically by a restart script
Hi,
thanks for your help
here is the show ver dump.
GWSCZ01#sh ver
Cisco Internetwork Operating System Software
IOS (tm) 5400 Software (C5400-IS-M), Version 12.2(15)T5, RELEASE
SOFTWARE (fc1)
TAC Support: http://www.cisco.com/tac
Copyright (c) 1986-2003 by cisco Systems, Inc.
Compiled
Holy cow. Option A is not possible. (I control the server myself!) but
hadn't considered option B. Sure enough, * seems to have restarted. I've
included a portion of the log here:
Dec 14 15:10:55 DEBUG[1751]: Set option AUDIO MODE, value: ON(1) on Zap/59-1
Dec 14 15:10:55 DEBUG[1751]: Hangup:
It's been hours since I've seen a post from this list
Must be broken again.
Regards
Greg Cirino
___
Cirelle Enterprises Inc.
603-425-2221
www.cirelle.com Web Application Development Design
www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster
www.cedata.com
Hello All,
I am receiving following error message while debugging Asterisk with
gdb.Help Appreciated
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 16384 (LWP 7710)]
0x4018f0f1 in strncpy () from /lib/libc.so.6
(gdb) bt
#0 0x4018f0f1 in strncpy () from /lib/libc.so.6
./iaxyprov 192.168.0.4 myiaxy.conf
what's in the myiaxy.conf file? (obliterate passwrds)
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On Tue, 2004-14-12 at 03:22 -0600, Justin Carlson wrote:
Hello all,
I have a uip200 for testing and I can't seem to get the phone to
register to my * server. I have configured the unidencomm.txt and the
unidenMACOFPHONE.txt files and the phone tries to register but * comes
back with
I had the same Problem with an CVS close to
1.0
Did you start * withoption Verbose (-...)
and console (-c)and closed these Terminal?
I had no Problems after starting * with no option
an then asterisk -r
mfg
Marco
- Original Message -
From:
Goutam Shaw
To: [EMAIL
this is a test
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Greg - Cirelle Enterprises [EMAIL PROTECTED] wrote:
It's been hours since I've seen a post from this list
Must be broken again.
You determined that the mail list was broken and decided to alert
everyone by posting an article to the list? Good thinking. :-)
--
_/ _/ _/_/_/_/ _/
I'm having exactly the same problem. I have sip.conf rows in the sql
table (ast_config), and removed the /etc/asterisk/sip.conf file. Now I
have no sip devices. It's as though realtime is not looking for the
sip.conf rows in the table.
This is my extconfig.conf:
[settings]
; Static
Fabrício Zimmerer Murta wrote:
Oh, friend... I have realised just yesterday that's impossible to use
regular modems (say hayes/v90 33.6 or 56k) to plug asterisk to the world. I
can't figure out why. But they simply don't support it.
If you want to use your isdn modem to plug * to the world, it's
Yes I did. Thanks it worked.
Regards,
Goutam
-Original
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Sent: December 14, 2004 5:09 PM
To: [EMAIL PROTECTED]; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 99%
CPU
I had
Google for asterisk gotoif -- third match down brings you to
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf
-Original Message-
From: VCI Help Desk [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 14, 2004 2:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Thanks a lot Peter, I will check this.
On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
Can you show me the simple example of this in asterisk words?
Make an extension that calls an agi and then starta the MeetMe application
in Asterisk. The agi should create two call files (see the
So in FC3 with udev you aparently need to start the zaptel service twice
to get it to work properly(according to the digium tech I spoke with
today.) However, it looked like it was all working and then I got off
the line and rebooted and things were still screwy, so what I did was
add a sleep
Hi,
We are looking for a regular supplier of Digium hardware in Australia.
any help will be appreciated.
regards
Kavit
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Bruno Hertz wrote:
I'm considering that board as a mail and voip gateway for home use.
In view of all those statements about how little resources asterisk
needs, did anybody already try running asterisk on it?
Thanks, Bruno.
Brunno,
Have a look at my site - http://www.krisk.org/astlinux/ - I have
Norman Zhang wrote:
Hi,
My firewall allows the first SIP packet out from * (running NAT), but
then it follows by dropping it saying SIP Reason: SIP Validator: Out of
State. May I ask how can I solve this?
Regards,
Norman Zhang
Norman,
With not much to go on, I am guessing that you have some
I -think- that the Sipura 841 was PoE... and I'd been anxious to find out.
However, according to Atacomm.com, it's been delayed until mid-January.
*sigh* So: does anyone know of a (decent) phone that meets the following
criteria, and isn't too expensive?
- SIP
- two (or more) lines
- some form
Check out used Lucent MaxTNT equipment there was a thread on the
mailling list about it.
On Tue, 14 Dec 2004 13:14:43 -0600, Matthew Boehm [EMAIL PROTECTED] wrote:
And T1s too. If you can supply a seperate piece of hardware that can handle
all the T1 crap then pass calls to asterisk as
Hello list,
your help will be apreciated in this regard.
i have an asterisk server (working just as SIP-H323 translator) I have
a customer that has a SNOM 4S sip proxy and with 60 sip users in
there, I want to terminate calls for them.
I wish to see Asterisk as a sip gateway from the snom part.
I actually had this same problem even with busycount=8 so I had to
remove the busy detect all together.
Jared Armstrong
-Original Message-
From: Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 14, 2004 10:27 AM
To: Asterisk Users Mailing List -
HengWee Chin wrote:
Hi,
I have a problem with the queue cmd.
I am trying to redirect an incoming call to another phone when nobody
in the queue answer it within 18 seconds. Somehow the incoming call
keeps on retrying within the queue. The second part was never executed.
Below is a part of
Hi -
We're moving up in the world to a PRI (Verizon), and I'm having some
problems with it. I'm new to this PRI thing, though, so maybe I've
just screwed up a simple config detail. I've got a TE410P on a Dell
PE1600SC (ServerWorks Chipset). The card itself has a green light for
the PRI, and
Sure thing,
First of all - I don't configure the phone manually. I use tftp config
files to accomplish this for the Cisco phones (I bring some home from
work... and don't want to load the firmware and configs manually
everytime!).
Here's a couple of links to get this done:
Hello list,
your help will be apreciated in this regard.
i have an asterisk server (working just as SIP-H323 translator) I have
a customer that has a SNOM 4S sip proxy and with 60 sip users in
there, I want to terminate calls for them.
I wish to see Asterisk as a sip gateway from the snom part.
Hi Kristian,
Kristian Kielhofner wrote:
Norman Zhang wrote:
My firewall allows the first SIP packet out from * (running NAT), but
then it follows by dropping it saying SIP Reason: SIP Validator: Out
of State. May I ask how can I solve this?
With not much to go on, I am guessing that you have
Hi -
Thanks for the fast response!
Try changing your dial statment to use a group instead such as
Dial(Zap/g2/301212)
Same result:
Spawn extension (from-sip, 000, 1) exited non-zero on 'SIP/2000-8dca'
-- Executing Dial(SIP/2000-a4f1, Zap/g1/19142246402) in new
stack
Dec 14 22:11:09
My firewall allows the first SIP packet out from * (running NAT), but
then it follows by dropping it saying SIP Reason: SIP Validator: Out
of State. May I ask how can I solve this?
With not much to go on, I am guessing that you have some
commercial firewall product - i.e. Checkpoint or
Can someone else confirm that your phone does not recieve MWIs when using
SIP and RealTime?
Is this a problem with SIP or with Voicemail?
-Matthew
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I have to ask
myself: Is the mentality of the press nothing more than a sensible
response to the interests of their audience?
Whoa... I haven't heard stuff this heavy since my Jimi Hendrix days!
I always thought Wired was more of a yuppie lifestyle
magazine than one to attract phrackers
In the phones web browser is an area that describes caller id as name or
number display,
Regards
Michael Hatzis
0421 476 211
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
Estep
Sent: Tuesday, 14 December 2004 5:34 PM
To: [EMAIL PROTECTED]
The hardware I currently have is:
TDM400P with 3 FXO ports, and 1 FXS port
4 Cisco 7960 Phones (only 1 is currently configured for testing purposes)
Asterisk on slack 10
I can dial out just fine via the Cisco phone, but when I try to dail
in I get the following output when I load
Brian Wilkins wrote:
So why not allow qualify=yes under the general heading for sip and iax?
That's not the issue :-)
The issue is that the thread that sends qualify messages has to have a
list of peers to send them to, and it works off of the peer list that
the channel driver keeps in memory.
Some of the others you mentioned, name etc, can be increased. But most of
those options that call for 'Yes', 'No' or NULL can all be 1 char wide.
-Matthew
Thanks Matthew,
greg
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Ken D'Ambrosio wrote:
- SIP
- two (or more) lines
- some form of TCP/IP-based configuration
- 802.3af (power-over-ethernet)
- 100 Mbit passthrough (not required, but would be nice)
- two (or more) lines
- echo cancellation
The Polycom IP300 fits this
Noah Miller wrote:
The Status has me concerned - Provisioned, Down, Active. Is that
Down normal?
Sounds like Verizon has not turned up your PRI yet, even though the loop
is up. They have to tell their switch to actually start talking to yours :-)
___
In the phones web browser is an area that describes caller id as name
or
number display,
Regards
Michael Hatzis
0421 476 211
or as in not both? :(
Thank you.
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Im having trouble with the Realtime setup. Ive
followed the instructions on voip-info using odbc but I get this message during
asterisk boot:
Parsing '/etc/asterisk/sip.conf': Not found (No such file or
directory)
Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config:
Unable to
Kevin P. Fleming wrote:
Ken D'Ambrosio wrote:
- SIP
- two (or more) lines
- some form of TCP/IP-based configuration
- 802.3af (power-over-ethernet)
- 100 Mbit passthrough (not required, but would be nice)
- two (or more) lines
- echo cancellation
The
I'm considering that board as a mail and voip gateway for home use.
In view of all those statements about how little resources asterisk
needs, did anybody already try running asterisk on it?
Thanks, Bruno.
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On Tue, 2004-12-14 at 14:14 +0100, HBK wrote:
Hi
After modprobe hisax type=35 (Billion HFC PCI) on a Xorcom Rapid ISO I get:
HISAX
Dec 12 16:25:35 localhost kernel: HiSax: Linux Driver for passive ISDN cards
Dec 12 16:25:35 localhost kernel: HiSax: Version 3.5 (module)
Dec 12 16:25:35
Michael Welter wrote:
Ryan Stark wrote:
Hello, I'm running Fedora Core 3 with udev, and asterisk/zaptel/libpri
from cvs. I have followed the README.udev instructions replacing
insmod with modprobe and rmmod with modprobe -r and adding the
60-zaptel.rules file, yet no matter what I do I still
qualify= and mailbox= do not work with the realtime
configuration engine. It doesn't matter if you specify
them in the database, the thread that handles them
will never look at the peers you have defined in the
database, only the ones defined in iax.conf.
---
Thank you.
I am having some problems getting TelIax service to
work with *. Outbound calls work just fine. When I try an inbound call the
phone rings and there is no audio. Upon further investigation iax2
show channels indicates that the codec is unknown The
provider confirmed that they are set for
I ran into the same problem (compiled perl with USE='ithreads' too.) I did an:
export USE='ithreads';emerge libperl
and it worked.
On Thu, 9 Dec 2004 09:45:42 -0500, Steve Woolley
[EMAIL PROTECTED] wrote:
On a new * asterisk install onto new install Gentoo 2003.4 upon startup
of asterisk:
HI
I got the same problem that only started lately. I have to do a
stop start to get the phones registered again. One site out of 12
with the same spec.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Race
Vanderdecken
Sent: Tuesday, December
Wow if that phone is Poe, and 80/90 bucks. Thats a steal right
thereawesome.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
D'Ambrosio
Sent: Tuesday, December 14, 2004 5:14 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sipura 841 delayed:
I'm trying to get two Windows Messenger clients to communicate with
video and audio though asterisk. I'm running into one of two problems.
I get garbled audio under the current config. I had another config
where I could get a voice call to work but using video would cause the
caller to get
Slow busy tells me that the telco has busied all his channels out --
likely
waiting for a call from him to finish provisioning... Unless his
dialplan has
_.,1,Busy or something. :-)
His pri show span said Down, that's likely the cause.
I guess I'll have to get Verizon on the case. Thanks for
Austech Partnerships (www.atp.org.au) I believe are the A-tick holders for
digium hardware in Australia. They have told me previously that digium
hardware not supplied by them is not approved for connection in Australia.
Even then the only approved hardware currently are the quad-port PRI cards.
On 15/12/2004 09:14 Kavit Munshi said the following:
Hi,
We are looking for a regular supplier of Digium hardware in Australia.
any help will be appreciated.
take a look at Australian Technology Partners in Melbourne. we've purchased
numerous TDM and TE410P cards from them and are quite pleased
Noah Miller wrote:
Here are my configs for the TE410P:
zaptel.conf:
span=1,1,0,esf,b8zs
Well, you say you have a green light, but this should be:
span=1,0,0,esf,b8zs
because you want your card to derive the clock from the Verizon-supplied
end of the circuit.
bchan=1-23
dchan=24
zapata.conf:
In article [EMAIL PROTECTED],
Matthew Boehm [EMAIL PROTECTED] wrote:
OK. I just downloaded asterisk-1.0.3.tar.gz and did a 'cvs co -r v1-0
asterisk' into 2 seperate directories.
I then did 'diff -ur asterisk-cvs/ asterisk-1.0.3/' and there were source
code line differences between the two.
Hi,
I am a newbie to Asterisk so please excuse me if this is a real stupid
question.
I configured Asterisk on Turbo Linux and it seem to work fine with 2
phones. Very simple configuration. There was no voicemail and calls were
made only from X-Lite IP phones on Windows PCs.
I got ambitious
On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
Can you show me the simple example of this in asterisk words?
Make an extension that calls an agi and then starta the MeetMe application
in Asterisk. The agi should create two call files (see the wiki for
details such as moving the
Hi,
I am new to the Asterisk
world. I dont know much about the architecture, but I am involved in
installing and configuring the VoIP system.
My requirement is to build a
VoIP system using the 4 input lines (ISDN up0 telephone lines), it must be
possible to receive calls from outside
Hello all,
Im new in all this and i need your help. I
have some legacy 56k and 33.6K modems and I want to test them to work with
asterisk before purchasing any new hardware. Can anyone provide me instructions
to test them. My hardware is:
-
Pentium 500 MHz with Suse
8.1
-
ISDN BRI 1
Hi,
I have several SIP registrations on my Asterisk box. Sometimes, I try to
call in the inbound number from 1 and find it doesn't work. When I do
sip show registry, it's showing Unregistered (and sometimes there are
several which are showing Unregistered). If I type reload, it registers
and
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting
Erik
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Ian Chilton
Verzonden: dinsdag 14 december 2004 11:33
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] Dial Plan Problems
Hi,
I
On Tue, 2004-12-14 at 21:32, Ian Chilton wrote:
Hi,
I am having a few dial plan problems which I wondered if anyone would be
able to help with.
Firstly, I wanted to send 0800 calls through 1 sip provider and other
08xx calls through another. I have this:
exten =
Hi,
I hope I won't bother too much if I ask you to provide some more info about
your setup, particularly which ports are open and other things (like how
often does Grandstream register, do you use keep alive, etc...).
Having that information I could rule out settings and maybe start searching
on
Hi all !
I try to set up the radius module for Asterisk
(http://appradius.minitelecom.org/) but i don't know
what i can do after the make of app_radius and
cdr_radius.
I would like to recording Asterisk CDR into Radius.
I waiting for your help .
Thanks in advance.
Anthony.
testing
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-Original Message-
From: Simon Ward [mailto:[EMAIL PROTECTED]
Sent: 14 December 2004 12:15
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Softphone features
Hi,
I'm currently looking for a softphone for windows, we have been using
X-Pro but it appears that X-pro doesn't
Well.. subject says it all really.
I have a TDM with 2 FXS modules and 1 FXO and a X100P.
If I load teh zaptel and wctdm drivers. Asterisk sees the TDM ports fine
but not the X100P
I have tried several combinations of port numbering but can some kind
person with a similar setup to
Hi,
What handsets are you using? Could be the firmware!
It's sip providers i'm having the problem with - not phones.
--ian
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A bad hack be to use the URL option in the Dial command. Does this
idea suck?
IMHO, it sucks.
Stuff like variables should be integrated in the protocol and
transmitted seemlessly.
roy
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Hello, Ive just set up SIP with asterisk using
this how-to: http://www.automated.it/guidetoasterisk.htm#_Toc49248757
but when I try calling the number at my SIP provider (Wx3) it doesnt
come thru. I THINK I registered to my SIP provider without any problem, in
sip.conf I do:
register =
Hi List,
I've got * randomly hanging up on inbound or outbound calls on zap
channels. I use a Digitnetworks X100P clone card. Any idea of what might
be happening?
Cheers,
Jean-Michel.
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Hi:
I'm having problems with Unicall channels in a heavily used asterisk. The
box is using 2 E1 lines in mfcr2, with both incoming and outgoing calls,
uning Argentina R2 variant. Besides the already known problem when the
outgoing calls failing and blocking a channel, now I have loop problems.
On Mon, 2004-12-13 at 21:04 -0700, Damon Estep wrote:
That is the difference between a commercial project and an open source
project; you must do your part to add value. Surely you do not expect
glossy ad slicks...
As I stated in my original post... just looking for a starting point.
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