Samy Antoun wrote:
--- Bill Maidment [EMAIL PROTECTED] wrote:
Hi
I've just tried to compile the zaptel-1.2.9 release and I get the
following error:
Same here, using CentOS 4.4 kernel 2.6.9-42.0.2.ELsmp, got these errors when
compiling zap:
make[3]:
There are several reasons why your attempts to set CNAM might fail:
(1) your upstream PRI provider does not allow you to set it (although
normally in this case you will find that you cannot set the CID either);
(2) someone between you and the called party will set the CNAM using a
directory
Rich Adamson wrote:
Have you tried:
cd /usr/src/zaptel
make update
make install
From a tarball? I don't think so!
That would only work for SVN and as we know the SVN version is OK.
--
Bill Maidment
Maidment Enterprises Pty Ltd
www.maidment.com.au
si hoc non legere potes tu asinus es
Thank you for your help, it works now
Michael
2006/9/6, Mojo with Horan Company, LLC [EMAIL PROTECTED]:
Try breaking up the contexts. Contexts are what you call 'submenus'.
For example:
[MetarMain]
exten = 1,1,answer
exten = 1,n,Background(Met_welcome)
exten =
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dijkstra, Roelof wrote:
Hello,
We currenty have an asterisk cluster running, with a quad PRI and a quad BRI.
This all works pretty well.
What i was wondering:
If i do a
show sip peers
I see all the ip addresses of the phones that
Hi,
Is there a way to do call notification to a desktop when a call is
connected from a queue to an agent ? I have seen the call notification
page in wiki, but they do not deal with queues.
raj
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Title: Satellite link-IAX Jitter Buffer.
I would like to setup asterisk via a dedicated satellite link with latencies of 700ms -1000ms I am having problems adjusting the jitter buffer and would like to find out if anyone has iax configs for a similar setting I am using g729 codec .I can
Dear folks,
In my senario I receive a call on a Zap channel and bridge it to a SIP
extension. On the sip client i should get what is the Zap partner of this
call. I though i should do it through the manager but i really dont know
how. I just couldnt even find what is the SIP channel that the
I would like to setup asterisk via a dedicated satellite link with
latencies
of 700ms -1000ms I am having problems adjusting the jitter buffer and
would
like to find out if anyone has iax configs for a similar setting I am
using
g729 codec .I can communicate but just having issues with
C F wrote:
send callerid to a ringing phone they don't look at what you set for
name, they look up the name for the phone number you set using a CNAM
query.
Thanks for the info.
Doug
-- Ben Franklin quote: Those who would give up Essential Liberty to
purchase a little Temporary Safety,
Michael Graves wrote:
Polycom Aastra are both great in this manner.
Even Cheapo PA168S phones will remotely update their configurations from
a simple handful of files from a tftp, ftp or http url. (which is
admittedly only simple to do with 30 phones simultaneously if you use PoE).
I'm very
I prefer Linksys ones. Spa 9xx series, are great, and provisioning from
Sipura/Linksys is much better than PA1628 (Unencrypted).
Supports https,tftp and http. With Encryption. Vonage use it.
Regards
Thomas Kenyon escribió:
Michael Graves wrote:
Polycom Aastra are both great in this
Alberto Sagredo wrote:
I prefer Linksys ones. Spa 9xx series, are great, and provisioning from
Sipura/Linksys is much better than PA1628 (Unencrypted).
Supports https,tftp and http. With Encryption. Vonage use it.
I'm sure they are much better, they should be they cost a lot more. I
was
The POE idea explains a lot. I was wondering how one forces a reboot on the phones in order to direct uptake of new settings.
Since I only have a handfull of extensions I use the phones built-in web management interface to reboot each phone. If it were more of an issue I'm reasonably
On Sun, 2006-09-10 at 09:07 -0500, Michael Graves wrote:
The POE idea explains a lot. I was wondering how one forces a reboot
on the phones in order to direct uptake of new settings.
In the case of Aastra phones (and others, I only use Aastras) use
sip_notify.
I don't have much details on your set-up, but I assume that since
quintums had performance troubles with SIP (about 2 years ago) your best
bet is to get them to work with h323. For that your first step willl be
to install h323 support on your asterisk box. I may be a little rusty
on this, so
Ok, this is the 3rd RMA replacement I've gotten from Cisco. None of
the phones were bad, but I have been trying to upgrade them to SIP
firmware from Skinny and I've always managed to turn them into
doorstops because I've factory reset them and afterwards realize I
can't do anything with
At 07:07 AM 9/10/2006, you wrote:
The web management interface on the Polycom phones in nowhere near
as good as the one in the Aastra phones. Aastra also supports
encrypted config files using freely downloadable software tools.
Just be careful using both the web interface and config files
I'm a big fan of these phones too for all of the reasons you pointed out.
They really stand out now that their firmware is mature.
I especially like their commitment to the open source community and
associated firmware/support policies.
-Original Message-
From: Tim St. Pierre
Firmware 1.1.1.9? Where's that? most recent on Grandstream's site that I see is 1.1.0.16, which I have loaded.
On Fri, 2006-09-08 at 20:20 -0500, Lacy Moore - Aspendora wrote:
You also have to make sure that on the web config for Grandstream that you allow it to receive auto-answer (or
Hi everyone,
While this isn't a true asterisk question, I know a lot of people here
use Polycom phones. Anyway, I have two Polycom 601 phones that share the
same voicemail box. Now it is intermittent, but sometimes both phones
will have a notification there is a voice mail, but then sometimes
--- Rich Adamson [EMAIL PROTECTED] wrote:
Have you tried:
cd /usr/src/zaptel
make update
make install
No, but I tried the SVN version and it compiled just fine.
Thanks
__
Do You Yahoo!?
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Jason Lixfeld wrote:
Ok, this is the 3rd RMA replacement I've gotten from Cisco. None of
the phones were bad, but I have been trying to upgrade them to SIP
firmware from Skinny and I've always managed to turn them into
doorstops because I've factory reset them and afterwards realize I
can't
Bill Maidment wrote:
Rich Adamson wrote:
Have you tried:
cd /usr/src/zaptel
make update
make install
I've never used the tarball, however if the tarball is installed and the
resulting code can't be compiled, obviously there is a Makefile
present. Part of the Makefile includes update,
Hi Friends,1) I am new to Trixbox. 2) First I explain my network architecture. I have a public IP and got internet connection from a ISP. I have connected the internet cable which is coming from ISP to a router. Now, I have connected to Trixbox server to the router. 3) I have assigned static IP
I have to agress with this. I am a US citizen and
what the US is doing is wrong. If they get away with this then A)They will do it
in the future and B) Who knows where else they will try to use usch
"law".
- Original Message -
From:
Dean
Collins
To: Asterisk Users
I'm trying to install Asterisk billing server and when I put in su - postgres I
get this response instead of the password response.
-bash-3.00$
-bash-3.00$
-bash-3.00$
Anyone seen this before? I'm using Fedora core 4 and have the same on a local
machine that works fine.
Has anyone been succesful at configuring tenor with asterisk.
I have a few analog fxs-fxo boxes and would like to put them to use
With asterisk, but I need help
Thanks.
On 9/10/06 11:54 AM, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Send asterisk-users mailing list submissions to
can someone please explain the differnces to me??? I have an asterisk system im setting up for a small office (4 or 5 phones)and as im looking for a voip provider, i find that voip providers generally have unlimited plans, and those that offer sip origination and termination get charged for
Rich Adamson wrote:
Bill Maidment wrote:
Rich Adamson wrote:
Have you tried:
cd /usr/src/zaptel
make update
make install
I've never used the tarball, however if the tarball is installed and
the resulting code can't be compiled, obviously there is a Makefile
present. Part of the
the thing to remember is that these terms are from the point of view of the PSTN.
So, SIP origination is Direct Inbound Dial (DID, or DDI in european parlance) which allows callers on the PSTN to originate calls that end up at your SIP server.
SIP termination allows calls which originate on your
Kevin Smith wrote:
Hi everyone,
While this isn't a true asterisk question, I know a lot of people here
use Polycom phones. Anyway, I have two Polycom 601 phones that share the
same voicemail box. Now it is intermittent, but sometimes both phones
will have a notification there is a voice
i thought so! this helps alot. thanks so much! Yair Hakak [EMAIL PROTECTED] wrote:the thing to remember is that these terms are from the point of view of the PSTN. So, SIP origination is Direct Inbound Dial (DID, or DDI in european parlance) which allows callers on the PSTN to originate
The Quintums work great with SIP (TenorAX at least). Don't bother with
H232 until you absolutely have to, and I bet you will not.
My problem with getting the box working was that is shipped with G729 as
the default codec and I use nothing but ulaw on my lan.
Can you get any debugging info
FRANCISCO PEREZ-LANDAETA wrote:
Has anyone been succesful at configuring tenor with asterisk.
I have a few analog fxs-fxo boxes and would like to put them to use
With asterisk, but I need help
Thanks.
On 9/10/06 11:54 AM, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I will be more than
offers unlimited calls, in and out in the US asterisk support no setup fee and support 729 codec? and of course is reliable and clearthanks alot.___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
Christopher Corn wrote:
can someone please explain the differnces to me???
I have an asterisk system im setting up for a small office (4 or 5
phones) and as im looking for a voip provider, i find that voip
providers generally have unlimited plans, and those that offer sip
origination and
Bill Maidment wrote:
Rich Adamson wrote:
Bill Maidment wrote:
Rich Adamson wrote:
Have you tried:
cd /usr/src/zaptel
make update
make install
I've never used the tarball, however if the tarball is installed and
the resulting code can't be compiled, obviously there is a Makefile
In article [EMAIL PROTECTED],
Nigel Godfrey [EMAIL PROTECTED] wrote:
On a new set up Centos 4.4, kernel 2.6.9-42.0.2.EL, yum updated, 2
BRI-HFC cards, no digium hardware.
modprobe zaptel and modprobe ztdummy are both in rc.local, and lsmod gives:
[EMAIL PROTECTED] ~]# lsmod
Module
thanks for the verbose explanation!Rich Adamson [EMAIL PROTECTED] wrote: Christopher Corn wrote: can someone please explain the differnces to me??? I have an asterisk system im setting up for a small office (4 or 5 phones) and as im looking for a voip provider, i find that voip providers
anyone has any success in using music
onhold.
even if we have ztdummy installed we still
got choppy music. buy our old asterisk 1.0.9 is working, why is
that?
thanks for any help in
advance.
Best Regards
matt
___
--Bandwidth and Colocation
actually Rich, not to be picky or anything, but your first paragraph is backwards.
There are some providers that allow you to originate calls to the US/World pstn network via their facilities, but do not provide any way for the US/World to call you from the pstn network. (eg, Origination
only
Yair Hakak wrote:
actually Rich, not to be picky or anything, but your first paragraph is
backwards.
There are some providers that allow you to originate calls to the
US/World pstn network via their facilities, but do not provide any way
for the US/World to call you from the pstn network.
ok maybe thats asking for too much. how about a voip provider that provides 729 codec support ? :)Christopher Corn [EMAIL PROTECTED] wrote:offers unlimited calls, in and out in the US asterisk support no setup fee and support 729 codec? and of course is reliable and clearthanks
Better to get one provider that does origination and termination and has no minimum requirements. Most companies will require a deposit or minimum usage requirements. Make sure for origination you know the diffference between metered and unmetered DIDs. codec 729 will be pushing it a little bit.
Rich Adamson wrote:
If you look at the sample configs, you'll find:
[EMAIL PROTECTED],[EMAIL PROTECTED] ; Subscribe to status of multiple
mailboxes
in the sip.conf.samples for v1.2 stable. That is the only way that I
know of to turn on the mwi for two different phones (eg,
Hi List
I'm trying to connect a to an idsn data device through an Asterisk box
--Pri-- asterisk --Pri-- Ip Office --bri-- Video Conferencing
unit
I would be using a channel that is configured for both voice and data no
specific DDI as the IPO will route that data call to the BRI,
I have almost 1,000 800 numbers that are routed any number of ways.
Currently I call on fastagi which checks a database and returns the
extension or route that DID is supposed to take.
The DIDs are all over the place as far as sequence, so pattern matching
is out of the question.
My
I spoke to a voip provider today who mentioned that though they offer an unlimited plan, if we use it for a business and it is over-utilized, it will be canceled.is this true for all residential voip plans? i have a small office of about 4 or 5 phones. i tend to chose residential plans because
Looking at a Asterisk server which will not be attached to
the Internet - and the user will be pretty computer illiterate.
Has anyone seen a script or some mechanism to set
the server time by using an extension, and entering the date/time via the
keypad?
Thanks!
Its a trickish business, when they say unlimited and you make more than 2500 minutes they cut you off.
-- Original message -- From: Christopher Corn [EMAIL PROTECTED]
I spoke to a voip provider today who mentioned that though they offer an unlimited plan, if we use it
i see. thanks for the info.[EMAIL PROTECTED] wrote:Its a trickish business, when they say unlimited and you make more than 2500 minutes they cut you off.-- Original message -- From: Christopher Corn [EMAIL PROTECTED] I spoke to a voip provider today who mentioned
I'd just use a service that's being offered to business customers...like Nuvio's nPBX. While they don't support Asterisk directly some of their resellers will support using *. I've used it for about 6 months and its been very reliable. The only annoying thing is that they only support SIP
thanks for the reply. why are residential lines cheaper than businesses? say for unlimited, it always costs more for residential.Michael Graves [EMAIL PROTECTED] wrote: I'd just use a service that's being offered to business customers...like Nuvio's nPBX. While they don't support Asterisk
What player? I found that my system had mpg123 but too new a version and
something was seriously hosed with it. I downgraded to the version listed
in the install help and it started working.
Nick
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR
On 10-Sep-06, at 1:58 PM, Chris Jones wrote:
Jason Lixfeld wrote:
Ok, this is the 3rd RMA replacement I've gotten from Cisco. None
of the phones were bad, but I have been trying to upgrade them to
SIP firmware from Skinny and I've always managed to turn them into
doorstops because I've
It's still beta... here is the link:
http://www.grandstream.com/BETATEST/GXP2000_BT200/
bp
On 9/10/06, Barry D. Hassler [EMAIL PROTECTED] wrote:
Firmware 1.1.1.9? Where's that? most recent on Grandstream's site that I see is
1.1.0.16, which I have loaded.
On Fri, 2006-09-08 at 20:20 -0500,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Christopher Corn wrote:
thanks for the reply. why are residential lines cheaper than businesses?
say for unlimited, it always costs more for residential.
*/Michael Graves [EMAIL PROTECTED]/* wrote:
I'd just use a service that's being
OK, help.. Am not sure where this is not configured right. I followed the
voicemail.conf directions, even tried specifying sendmail -t directly.
My sendmail mail log shows:
Sep 10 17:19:26 goonie sSMTP[4439]: Unable to locate mail
Sep 10 17:19:26 goonie sSMTP[4439]: Cannot open mail:25
Nick
John Marvin wrote:
Rich Adamson wrote:
If you look at the sample configs, you'll find:
[EMAIL PROTECTED],[EMAIL PROTECTED] ; Subscribe to status of
multiple mailboxes
in the sip.conf.samples for v1.2 stable. That is the only way that I
know of to turn on the mwi for two different
Christopher Corn wrote:
I spoke to a voip provider today who mentioned that though they offer an
unlimited plan, if we use it for a business and it is over-utilized, it
will be canceled.
is this true for all residential voip plans? i have a small office of
about 4 or 5 phones. i tend to
Not to answer a question with a questionbut why do so many businesses focus so intently on the cost of their voip service? If we presume that a business intends to stay in business, and that phone service is crucial to actually being in business, then I've never seen the wisdom of going to
Rushowr wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Christopher Corn wrote:
thanks for the reply. why are residential lines cheaper than businesses?
say for unlimited, it always costs more for residential.
*/Michael Graves [EMAIL PROTECTED]/* wrote:
I'd just use a service that's
Take this to sendmail list. this is not an asterisk problem. In any
case it looks like it's trying to send email to host mail on port 25
and it's failing. Try doing a telnet mail 25 and see what happens.
On 9/10/06, Nick Ellson [EMAIL PROTECTED] wrote:
OK, help.. Am not sure where this is not
Michael Graves wrote:
Not to answer a question with a questionbut why do so many
businesses focus so intently on the cost of their voip service? If we
presume that a business intends to stay in business, and that phone
service is crucial to actually being in business, then I've never seen
Rich Adamson wrote:
Phones don't monitor mailboxes. One needs to tell asterisk which
phones are to be notified when a voicemail is left, and the sip
statements above are the only ones that I'm aware of to accomplish that.
Yes I am aware of that. Perhaps I chose the wrong wording, but I
Hi, this message is for Steve.
Sorry for replying to the digest. It wasn't my intention.
I would appreciate if you can guide as to how make the tenor asm200 work
with asterisk. I am using asterisk at home. I guess my problem is
configuring the tenor so that it is recognized and can take calls
Hey, that's why i had no idea how to spot the glitch... I added a line in
my /etc/hosts file for mail aimed at my SMTP server, all better now.
Thanks!
Nick
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Sun, 10 Sep 2006, C F
On Sun, Sep 10, 2006 at 07:02:12PM +, [EMAIL PROTECTED] wrote:
I'm trying to install Asterisk billing server and when I put in su - postgres
I get this response instead of the password response.
-bash-3.00$
-bash-3.00$
-bash-3.00$
Run 'id -a' there and you'll see that you are actually
Hi I'm getting really bad static on forwarded calls to the point of
not being able to hear the person at the other end. I'm running an E1
line in and everything else is fine. I'm also getting this error:
Sep 11 14:56:56 WARNING[18295]: chan_sip.c:2561 sip_write: Asked to
transmit frame type
At 06:49 PM 9/10/2006, you wrote:
thanks for the reply. why are residential lines cheaper than businesses?
Because residences tend to use the phones less and even less during peak hours.
Ira
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If you want to answer directly to him, try Reply to all, and delete
[EMAIL PROTECTED] email address.
It is not so had to do.
FRANCISCO PEREZ-LANDAETA escribió:
Hi, this message is for Steve.
Sorry for replying to the digest. It wasn't my intention.
I would appreciate if you can guide as to
Hey Joel,
That's a nasty one.. Jen and I experienced that for a while on our
office machine.
The nasty part being that I can't tell you what it is, as we rebuilt the
box with a newer version and the problem went away.
(And at the time, we just breathed a sigh of relief and didn't question
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