Hi all,
I just picked myself up a Mediatrix FXO SIP gateway to play around with
and hook into Asterisk but have no documentation.
Are there default passwords or IP's that I need to know if I do a
factory reset?
Or better still, would anyone have a User Manual they could send my
way? Any
On Thu, 2 Sep 2004, Michael George wrote:
I've a question about the bandwidth consumed by IAX2/GSM.
According to the wiki page, the GSM codec should run about 13 kilo-bits/sec
for a voice encoding.
However, watching gkrellm when I initiate a call to Digium, it looks like the
channel is
Hi,
I've purchased two x100p clones, and when I try accessing a line
from asterisk with something like this:
exten = _1NXXNXX,1,Dial(Zap/2/{$EXTEN})
(is that only supposed to put you on channel 2 or actually dial the #
for you?)
but I first hear noise, then a dial tone, but as soon as
hi,
i'm under the impression that this feature is not available in asterisk,
consider this scenario:
- you are the operator. you answer a call from outside and you want to
transfer it to one of the extensions. after you transfer, if the person
you transferred the call to, doesn't pick up or if
Have you contacted digitnetworks for support? This list is owned and
maintained by Digium, who already gave you Asterisk for free. Probably
not the best forum to ask for support for a competitive product here.
-Original Message-
From: Imran Akbar [mailto:[EMAIL PROTECTED]
Sent:
Imran Akbar wrote:
Hi,
I've purchased two x100p clones, and when I try accessing a line
from asterisk with something like this:
. . . .
Any suggestions?
Throw them away and get Digium cards.
B.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I have the same hardware (x2)
/etc/zaptel.conf file
fxsks=1-2
loadzone=au
defaultzone=au
/etc/asterisk/zapata.conf file
[channels]
language=en
context=inbound
group=1
musiconhold=default
; need these much shorter than defaults
flash=90
signalling=fxs_ks
threewaycalling=yes
transfer=yes
I have a couple of questions on the zapbarge:
1) zapbarge asks for a channel - how would a manager know what channel to
enter ? Is there any way of being able to enter an extension number instead
? I know that you can get the information from the manager interface, but I
wouldn't want to give my
Does it mean that we cannot talk about Cisco or other FXS products since
IAXy is released??
I hope this list for every member who uses asterisk not Digium's products
users alone.
- Original Message -
From: Jay Milk [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial
Digitnetworks is profiting off the cards so they should support them.
If it wasn't for Digium there wouldn't be Asterisk anyway. So doesn't
that make it better to support the primary company for software that
many of you use every day at home and work?
On Fri, 3 Sep 2004 08:40:59 +0100, Kannaiyan
Didn't want to start a flamewar here... but anyway, could the issue be
that both fxo cards are on IRQ 11? How do I even change that?
Thanks
William Suffill wrote:
Digitnetworks is profiting off the cards so they should support them.
If it wasn't for Digium there wouldn't be Asterisk anyway. So
Good day all
I'm interested in video on asterisk using SIP and windows clients
Now I did my research on http://www.voip-info.org/wiki-Asterisk+video
I have a few question:
*On the page they say you need the H.261 H.263? codecs,are these compiled in
by default or do I need to do something special
If you could learn from the previous mails around here, as far i have seen
the issues were discussed based on the use of asterisk with and without
devices, not just supporting digium alone. You can see mails from
broadvoice, voicepulse, iconnecthere. Do they support Digium? never mind
about
Hello,
I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2
installed but failed. I applied the patch to the required OpenH323 library
according to the instructions, and set the proper directories in the Makefile.
Here is what I receive after I issue make:
On Fri, 2004-09-03 at 10:56, Altus Snyman wrote:
Good day all
I'm interested in video on asterisk using SIP and windows clients
Now I did my research on http://www.voip-info.org/wiki-Asterisk+video
I have a few question:
*On the page they say you need the H.261 H.263? codecs,are these
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi All,
I've just finished my upgrade to asterisk RC2.
I need to have H323 support, and in the last months i've been using
the chan-oh323 with good results.
My question is: anyone in the list have made tests with both chans
(oh323 and h323), which is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I Vlasis,
I'm using those versions (Fedora COre 1) and it compiled without
problems, but when i try to initialize asterisk i get the folowwing error:
ERROR [-1084337504]: chanoh323.c:4636 load_module: H.323 listener
creation failed.
Hope someone can
Hi,
we're testing Asterisk 1 RC 2 behind ordinary router and NAT. Since we're
sharing network with web server it seems like voip packets are not coming
through fast enough (Digium demo dies after few seconds...). It's the same
if I make direct calls (passing Asterisk) so we conclude it's network
Hi all,
Im using asterisk. I have one doubt.
Im running asterisk in one machine(RedHat9.0)
running firefly softphone in 3 windows machine
I hv 3 users in sip.conf like 1001, 2001 3001
appropriate entry for those users are also include in
extensions.conf like
I have my x-lite connected to the server but messanger does not want to log in
It does not even show its trying on the server
I went and seclected sip and adduse the server and username.no
[EMAIL PROTECTED]
Is there anything special
Thanks Altus
On Friday 03 September 2004 10:38, Vladyslav
Joa~o Amaro wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I Vlasis,
I'm using those versions (Fedora COre 1) and it compiled without
problems, but when i try to initialize asterisk i get the folowwing error:
ERROR [-1084337504]: chanoh323.c:4636 load_module: H.323 listener
creation failed.
It works fine for me on a Slack9.1 laptop.
Michael.
Vlasis Chatzistayrou wrote:
Hello,
I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2
installed but failed. I applied the patch to the required OpenH323 library
according to the instructions, and set the proper
Hello ml,
i need some help on my zaptel configuration. My E100P only shows some
YELLOW / RED alarm when I load the wct1xxp module and do a
cat /proc/zaptel/1
Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS YELLOW RED
...
..
.
My /etc/zaptel.conf is:
span=1,1,0,ccs,hdb3
Im using asterisk. I have one doubt
Question, not doubt. I wonder why all people from India have doubts and
not questions :-)
I guess that because of the hindu language characters you use HTML e-mail?
However, for english mailing lists it's better to not use HTML, but pure
text. Then people
hi all
Attachd is a PRI DEBUG dumped while dialling out to a busy number among
with zap(ata|tel).conf. asterisk did not flag busy, and I got a busy
indicator going mep-meep-mep-meep-mep-meep (never heard
this before)
Can someone help me out here?
thanks
roy
zapata.conf
On Fri, 2004-09-03 at 05:31, Jan Goericke wrote:
Hello ml,
i need some help on my zaptel configuration. My E100P only shows some
YELLOW / RED alarm when I load the wct1xxp module and do a
cat /proc/zaptel/1
Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS YELLOW RED
...
Hello,
Thanks for replying. On a Slackware 9.1 it may compile, but on a RH9 it
doesn't and I don't think we can install another distro on that machine...
:-)
I guess I'll have to wait for the new version of OH323 in order to try
cimpiling again...
Best regards thanks,
Vlasis.
Look at the wrt54g or wrt54gs with sveasoft firmware and wondershaper, allows you to
QOS VoIP data.
Google for sveasoft forums to find the right forum to search.
P
-Original Message-
From: Robert Rozman [mailto:[EMAIL PROTECTED]
Sent: Friday, September 03, 2004, 2:32 AM
To:
Hi,
Having no digium hardware in my box and two cpus and a ohci usb bus im
forced to use zaprtc.
I have recompiled the kernel and removed enhanced rtc support.
When I attempt to compile zaprtc I get the following error.
zaprtc.c:1077: warning: implicit declaration of function `barrier'
Well thanks for trying to help, mod=0 didn't fix that problem.
I'll check out the sequential problem later, didn't notice that.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Baker
Sent: Thursday, September 02, 2004 11:51 PM
To: Asterisk Users
On Tue, Aug 10, 2004 at 10:00:58AM +0200, Stefan Tichy wrote:
Using active AVM cards in connection with kernel 2.6 seems to be a
bad idea.
http://listserv.isdn4linux.de/pipermail/i4ldeveloper/2004-August/000630.html
This patch should be interesting if you are using AVM B1 cards and
kernel
On Fri, Sep 03, 2004 at 08:26:28AM +0200, [EMAIL PROTECTED] wrote:
On Thu, 2 Sep 2004, Michael George wrote:
I've a question about the bandwidth consumed by IAX2/GSM.
According to the wiki page, the GSM codec should run about 13 kilo-bits/sec
for a voice encoding.
However, watching
Hello Clayton,
Is there chances that you share your work with the list :)
I am planning to create an Asterisk testing tool,
- Generate call to an other Asterisk Box
- Check if the Asterisk answer correctly
- Check if the application is well played, etc...
I guess your application would be a
On Thu, Sep 02, 2004 at 01:30:05PM +0300, Tzafrir Cohen wrote:
Another beginner's question:
Can I gain root if I have write access to asterisk's config files?
If the asterisk process has root priviledges only root should be
allowed to modify its config files. But root priviledges are not
did you tried it with crc4 as well ?
span=1,1,0,ccs,hdb3,crc4 ?
On Fri, 2004-09-03 at 13:00, Steven Critchfield wrote:
On Fri, 2004-09-03 at 05:31, Jan Goericke wrote:
Hello ml,
i need some help on my zaptel configuration. My E100P only shows some
YELLOW / RED alarm when I load the
Hi, all!
Will asterisk use G729 license if both ends have support for G729 and no
transcoding needed? So, the scheme:
remote phone G729Asterisk with G729 codeclocal phone G729
As I understand, in this situation everything can be passed through, and
is so on default asterisk
Thanks for the hint.
I did it and zap show channels shows me the 31 channel. But when I check
/proc/zaptel/1, i still get the same error as before.
On Fri, 3 Sep 2004, Steven Critchfield wrote:
On Fri, 2004-09-03 at 05:31, Jan Goericke wrote:
Hello ml,
i need some help on my zaptel
I've a question about the bandwidth consumed by IAX2/GSM.
According to the wiki page, the GSM codec should run about 13 kilo-bits/sec
for a voice encoding.
However, watching gkrellm when I initiate a call to Digium, it looks like the
channel is taking a consistent 5-6
Try using Authenticate() to permit zapbarge access to others. With
Zapbarge you may also supply the channel number. You can also implment the
secruity that you want by using the simple features of extensions.conf.
For example:
exten = 100,1,Zapbarge()
- OR -
exten = 100/5002,1,Zapbarge()
the
Hi,
I did this the following way:
-) define a global variable - AGENTS_AVAIL=0
-) when agent logs in increment -
SetGlobalVar(AGENTS_AVAIL=$[${AGENTS_AVAIL} + 1]);
-) when agent logs off decrement -
SetGlobalVar(AGENTS_AVAIL=$[${AGENTS_AVAIL} - 1]);
-) when queue is called evaluate and goto
Hello list,
Is there some parameter on sip.conf to always let the client reachable ?
I'm trying to avoid this situation :
Sep 3 09:49:29 NOTICE[135442432]: chan_sip.c:7653 sip_poke_noanswer:
Peer '1264' is now UNREACHABLE!
Sep 3 09:49:39 NOTICE[135442432]: chan_sip.c:6408 handle_response:
I just picked myself up a Mediatrix FXO SIP gateway to play around with
and hook into Asterisk but have no documentation.
I spent a substantial amount of time evaluating the 1204 box back in
the January timeframe, and then returned it to the reseller. I can
answer some of your questions but
Jefferson Carvalho wrote:
Hello list,
Is there some parameter on sip.conf to always let the client reachable ?
I'm trying to avoid this situation :
Sep 3 09:49:29 NOTICE[135442432]: chan_sip.c:7653 sip_poke_noanswer:
Peer '1264' is now UNREACHABLE!
Sep 3 09:49:39 NOTICE[135442432]:
I have the user manual, I'll send it to your email tonight when I'll be in
my home.
I have an APA III-4FXO too, until today I can't put it to work with
asterisk.
Kind regards,
Miguel
Date: Fri, 03 Sep 2004 16:07:59 +1000
From: Jamie Carl [EMAIL PROTECTED]
Subject: [Asterisk-Users] Mediatrix
Yes I tried this too. But the problem is the same.
On Fri, 3 Sep 2004, Michael Bielicki wrote:
did you tried it with crc4 as well ?
span=1,1,0,ccs,hdb3,crc4 ?
On Fri, 2004-09-03 at 13:00, Steven Critchfield wrote:
On Fri, 2004-09-03 at 05:31, Jan Goericke wrote:
Hello ml,
i
Hi all, did not find much info in lists about subj.
I have ztdummy working properly because I can use conferences without
any errors.
But when I try to use trunk=yes, I get the following:
Sep 2 21:20:51 WARNING[1137720112]: chan_iax2.c:6422 build_user:
Unable to support trunking on user home'
Is there any reason why there should ever be more than 1 instance of mpg123
running on a * server?
I just did an 'uptime' and noticed all 3 of my loads where over 3.00.
'top' showed 8 mpg123 processes all processing the same 3 songs (our
background music).
I tried to kill one of them but
Absolutely the IRQ issue is probably the root cause.
How do you change that? Move the cards around on the PCI slots until they
are on seperate and unique IRQ's.
Lyle
- Original Message -
From: Imran Akbar [EMAIL PROTECTED]
To: William Suffill [EMAIL PROTECTED]; Asterisk Users Mailing
check your musiconhold.conf, for each one you define you'l get an instance.
-Original Message-
From: Matthew Boehm
To: [EMAIL PROTECTED]
Sent: 03/09/04 15:04
Subject: [Asterisk-Users] mpg123 - multiple instances, taxing CPU
Is there any reason why there should ever be more than 1
This is all that is in that file.
musiconhold.conf
-
;
; Music on hold class definitions
;
[classes]
default = mp3:/var/lib/asterisk/mohmp3
There are 4 mp3 files inside that dir. Any ideas?
Matthew
- Original Message -
From: Steve Hanselman [EMAIL PROTECTED]
To:
Actually, I got almost the same issue (i´m not having such load), but I got
defines 4 different moh and got 10 process (I check every time I restart * to kill all
the mpg123 processes also.
LTenorio
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf
Use the 's' extension...
On Thu, 02 Sep 2004 19:42:13 -0700, Kevin P. Fleming
[EMAIL PROTECTED] wrote:
I've got a need to do something like the following:
[foo-context]
exten = _.,1,SetCIDNum(123)
exten = _.,2,SetCIDName(XYZ)
include = local
include = tollfree
But of course, this
The difference is that digitnetworks specifically targets Digium as
competition. Cisco, Sipura, etc, don't directly compete with IAXy
because they have different feature sets and were around long before
IAXy was released. Digium was first on the market with the X100P and
digitnetworks cloned
I frequently get this error message, it repeats itself hundred/thousands
of times and never stops.
chan_sip.c:7467 sipsock_read: Failed to grab lock, trying again...
During this period, I can make no SIP calls what-so-ever. The only way
I've been able to stop it is to killall -9 asterisk. Doing
Is there a way for a natted client with
a dynamic ip address to receive call from the asterisk box ?
I can call from the natted phone using
tasterisk but I can't receive call in the natted phone because *
does not know the ip address of the phone
I have enabled the registration but
when I
Rob Fugina wrote:
Use the 's' extension...
Uhh, no. That doesn't work at all.
The s extension is only used if the channel coming into this context
doesn't have any target extension to look for. If it does, the s
extension is never used. If you have a context for SIP phones, and one
of them
This means either that:
- you do not have nat=yes in the sip.conf for that device,
- or you don't have a STUN server ip in the device settings
- or the device has not properly logged in to * (various reasons).
Turn on sip debugging and see if you see any error messages like 404 Not
Authorized
Hello,
Just wondering if anyone has tried connecting the Dlink
Video Phone (DVC-1000) to Asterisk. It would be cool if you could use Asterisk
as an MCU.
~Ken
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.745
Ah, well... Never tried it with SIP phones. I thought I had used
that before for inbound calls on a Zap channel, and with local Zap
extensions, too...
On Fri, 03 Sep 2004 08:11:09 -0700, Kevin P. Fleming
[EMAIL PROTECTED] wrote:
Rob Fugina wrote:
Use the 's' extension...
Uhh, no. That
You need to a method other than 'include =', which effectively concatenates
the target of the include with the current context. Consider this approach
instead:
[foo-context]
; This needs to match the criteria for tollfree, say a 91800 prefix
exten = _91800.,1,SetCIDNum(123)
exten =
If 'immediate=yes' then the target exten in the context for the zap line
will always be 's', where you would implement digit collection or whatever.
If 'immediate=no' then the simple switch code will collect the digits and
dive in to the context with something to match against, thereby ignoring
To top this off, I also get PRI errors
Sep 3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got
event: 6 on Primary D-channel of span 1
Sep 3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got
event: 6 on Primary D-channel of span 1
Sep 3 10:56:52 NOTICE[196620]:
Hi,
Were looking at options for logging agents into the system
programmatically via Perl/PHP and I was wondering if anyone else is doing
this and if so, how. We're using AgentCallbackLogin now but would like to
set up a web interface instead. I've been looking at Asterisk::Manager
and didn't
Hi Angel-
Had trouble getting Dell's in Portugal, however customer can get HP Proliant
DL320's. I had one shipped to me here, and ran it through some load tests.
Seems fine.
Thanks for responding!
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London
Hi: i have a problem.
Mi extensions.conf:
exten = _N.,1,Setvar(VOICEMAILREQ=${EXTEN})
exten = _N.,2,SetAccount(${customer})
exten = _N.,3,SetCDRUserField(${VOICEMAILREQ:1})
exten = _N.,4,ResponseTimeout(5)
exten = _N.,5,Background(ifyou)
exten = _N.,6,Background(silence/1)
Not that it makes any significant difference, but the x100p was an
off-the-shelf card that digium integrated into * and spent the time
writing the drivers, etc. The TDM card is a digium copyright design.
The difference is that digitnetworks specifically targets Digium as
- Original Message -
Subject: [Asterisk-Users] Sorry, Newbie here
To: [EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
I never heard of Asterisk before today, but from what i'm looking at
on the website and hearing, it sounds pretty
I think one of the greatest things about * is that not only do you get the
most flexible PBX I've ever worked with, but it also can act as a IP gateway
for much less than traditional hardware IP gateways (a. la.
Cisco/Mediatrix/etc...). You can use it to extend an existing PBX and save
thousands
Kannaiyan Natesan [EMAIL PROTECTED] wrote:
If you could learn from the previous mails around here, as far i have seen
the issues were discussed based on the use of asterisk with and without
devices, not just supporting digium alone. You can see mails from
broadvoice, voicepulse, iconnecthere.
Title: Message
TRUNKBP=Zap/g2
This is E1 trunk to Ericsson BusinessPhone
PBX.
The channel is not answered in that moment. First ring goes
to all phones, and after that only first phone continues ringing and only this
one can be answered.
From: [EMAIL PROTECTED]
[mailto:[EMAIL
The new one, it was upgraded few days ago
CVS-HEAD-08/29/04-13:17:08
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Kevin Walsh
Sent: Wednesday, September 01, 2004 5:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Do these two events coincide? If so, I'd suspect memory problems.
If they don't coincide, I'd still suspect memory, but I'd also look at
IRQ sharing issues.
On Fri, 2004-09-03 at 09:16, Daniel Jimenez wrote:
To top this off, I also get PRI errors
Sep 3 10:56:52 NOTICE[196620]:
Quick questionI have queues setup, when an agent parks a customer
and the park times out, it goes back to the queue. Is there any way to
get it to go back to the extension of the agent that parked them without
using the ParkAndAnnounce cmd?
Thanks,
-Ronan
Jay Milk [EMAIL PROTECTED] lazily top-posted:
The difference is that digitnetworks specifically targets Digium as
competition.
Competition is a good thing, in my view.
I didn't find out about the non-Digium X100P cards until after I'd
bought mine (for use at home). If I'd known then I
The T400P (and E400P) are clones of the Zapata Tormenta II, and anyone
can download the artwork to build and sell their own version. If the
owners of the Zapata Telephony project didn't want people to use their
designs then they would not have released them under the GPL and
published them
On Friday, September 03, 2004 8:45 AM William Suffill wrote:
Digitnetworks is profiting off the cards so they should support them.
I think that it wasn't so much an issue of Digitnetworks vs. Digium
supporting them, but rather Asterisk supporting them.
If it wasn't for Digium there wouldn't be
On September 01, 2004 12:06 PM, Scott Laird wrote:
This brings up an interesting point--disconnect supervision *mostly* works
for me with a X100P in the US. The exception is when calls go to
voicemail; I frequently end up with ~90 seconds of dialtone instead of a
message or a clean disconnect.
Hello All:
We have latest cvs version running on FC2 with one digium
card for PSTN.
When we call the asterisk server the demo greeting answer
but we hear a unintelligible voice with a robotic or like underwater voice. Any
ideas on this issue will be appreciated.
Thanks
Cele
The Mepis Debian distro is pre-configured for *, www.mepis.org They spent a
lot of time making Mepis work with * out of the box.
Everyone has their own very strong opinions on which distro is better. I'm
not about to get into that. All I can say is Mepis is probably your fastest
easiest way to
I can send sms messages just fine via a calling file, however, I cannot send
messages that have more than one line. How do I encode the message to
This is line 1
This is line 2
This is line 3
* complains about 2 syntax errors (I presume because the calling file has
three lines for the message),
Kris Boutilier wrote:
[foo-context]
; This needs to match the criteria for tollfree, say a 91800 prefix
exten = _91800.,1,SetCIDNum(123)
exten = _91800.,2,SetCIDName(XYZ)
exten = _91800.,3,Goto(tollfree,${EXTEN},1)
This is the direction I started going; however, I need to implement this
for
Lol... This never clicked before... It's called Zapata Tormenta (Shoe
Storm)... Like a bunch of women at a shoe sale I guess...
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
Two days ago, I was talking on the phone from the FXO, to a SIP phone.
After some time (like 1h30m), all of a sudden, there's a huge noise,
like a buzz... Really loud. So I hungup, and called my asterisk box
again... All I could hear was that sound. Someone called me from the
internet, and as
I just ran across the * site. Looks great. I do not need
a PBX at this time, but DO need to replace an old voice mail
system. I'll do my homework and figure out the specifics,
but before I dive into it all and spend a bunch of time only
to find out I didn't understand, is it reasonable to think
Are the test versions configured for * out of the box?
Mike C.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler
Sent: Friday, September 03, 2004 1:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
On Sep 3, 2004, at 10:12 AM, Kevin Walsh wrote:
Competition is a good thing, in my view.
I didn't find out about the non-Digium X100P cards until after I'd
bought mine (for use at home). If I'd known then I probably would have
avoided the massive markup and bought one of the clones. These days,
Marconi,
I don't know if this is will help you, but I had problems with some
TDM400p cards. They worked fine, but after about 10 minutes in use there
was a very loud static, humming noise. The cards where brand new, rev.
G. I spoke with Digium about the problem, and they suggested that I
On Fri, 3 Sep 2004, Bill Andersen wrote:
I just ran across the * site. Looks great. I do not need a PBX at this
time, but DO need to replace an old voice mail system. I'll do my
homework and figure out the specifics, but before I dive into it all and
spend a bunch of time only to find out
I am new to Asterisk and VoIP. I have been given the task of setting up a telephone
network in US and India. When customers call the US location, the calls should route
to India (using VoIP) and handle there. The Indian location should be able to call Us
numbers using the Voip to save money.
Tor,
Unfortunately (?), my Asterisk, Zapata, and Zaptel versions are already 1.0-RC2.
I apreciate your help, though. :)
Best regards,
Marconi.
On Fri, 03 Sep 2004 11:44:29 -0700, Tor Roberts [EMAIL PROTECTED] wrote:
Marconi,
I don't know if this is will help you, but I had problems with some
Hi,
I believe what you're looking for is QoS. I didn't mess around with it
yet... But I know you can setup a cheap linux router with it, so your
VoIP traffic will get more priority.
Here's an idea: setup one linux box as a router, with 1 ethernet for
inside voip, another one for the rest, and
Is it a good idea to use this option? Or its not stable and going to be
replaced soon anyways?
I'm looking for a stable solution to provision users from a db. Anything
working well w/ *?
TIA
-jon
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Hello!
Is there a way to use AVM Fritz!PCI as a ZAP interface and have it
configured for ZAP channels?
Thanx in advance!
Roland Zagler
mailto:[EMAIL PROTECTED]
@fog smart partners
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Try to specify the the context, it seems to be using default which may
or may not be right.
exten = s,1,HasNewVoicemail([EMAIL PROTECTED]|NEWMSGCOUNT)
Umar
On Thu, 2004-09-02 at 12:51, Nick Barnes wrote:
Hi all,
Maybe I'm being thick here, but I've had a look through the mailing
Can Anybody help how to reject an incoming call using 7960?
-Kannaiyan
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I'd be more than happy to send you some info off-list on how to do this in
Linux... It's much cheaper and more flexible than a low-end hardware
solution...
-Chris
- Original Message -
From: Robert Rozman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Marconi,
Marconi Rivello wrote:
Two days ago, I was talking on the phone from the FXO, to a SIP phone.
After some time (like 1h30m), all of a sudden, there's a huge noise,
like a buzz... Really loud.
You are not alone. This problem has also been experienced by many with
tdm400p cards.
There is
Hi -
no, you can't use the Fritz card as a Zap interface. Use a card that
has the HFC chipset. e.g. Billion, Asustek, etc. They are around EUR15
if you shop around. This works using the bri-stuff drivers from
www.junghanns.net
Rgds
Tim
Roland Zagler wrote:
Hello!
Is there a way to use AVM
Any advice, pointers to more info ?
MeshBox'll work:
http://www.locustworld.com/modules.php?op=modloadname=Newsfile=articlesid
=52mode=threadorder=0thold=0
SIP prioritization is supposed to happen regardless if the clients are wired
or wireless.
The distro is free:
Chris,
I believe it would be nice to send the info also to the list. So
others would be able to benefit as well. You've got at least 2 people
interested :)
Marconi.
On Fri, 3 Sep 2004 13:41:30 -0700, Chris Shaw [EMAIL PROTECTED] wrote:
I'd be more than happy to send you some info off-list on
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