On Friday 25 January 2008 05:25:57 Lyle Giese wrote:
You need to do a 'make' before the 'make install'.
make install will do all that is necessary to install a program including
making any files necessary.
--
Dave Cotton
___
-- Bandwidth and
Hi Jarga,
What type of connection you are using is it VoIP or ISDN PRI, if it is
VoIP check your dtmfmode in sip.conf if it is PRI check zapata.conf
On Jan 25, 2008 12:13 AM, Jarga Jallow [EMAIL PROTECTED] wrote:
Hi,
I am having trouble making a selection when I call a number and need
Hi Pandey,
What type of OS you are using, is it redhat or fedora. and install with
latest version.
On Jan 25, 2008 9:55 AM, Lyle Giese [EMAIL PROTECTED] wrote:
You need to do a 'make' before the 'make install'.
Lyle
[EMAIL PROTECTED] wrote:
Hi all,
Please help me in installing
Hi,
I am using Fedora and I tried to install Libpri-1.4.3 but now I am getting
error AVC access denied. Its saying I need to disable SELinux protection. I
do not know what to do. Please help me out.
Thanking you,
Preeta
-Original Message-
From: [EMAIL PROTECTED] on behalf of Gopal
Has anyone experience with (or an educated guess of) the largest paging
group that can be supported by the Page() command?
We have an installation coming up with 110 phones -- any hope to page
this entire facility?
--
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102
Hi Dave,
I did make clean and then make. But then when I am giving make install its
giving error AVC access denied.
I am using Fedora.
What may be the problem?
Help me..
Thanking you,
Preeta Pandey
-Original Message-
From: [EMAIL PROTECTED] on behalf of Dave Cotton
Sent: Fri
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Good morning,
Is it possible with asterisk to allow to share the same account on 2 different
devices, for example I want both my fix phone and my wifi phone to ring
in the same time.
I want to do it without making ringroups...
Any idea how to do it?
Hi all,
i need sample xml configuration files for linksys pap2, linksys pap-2t,
sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are
linksys/sipura products. So if anyone has these sample files then plz share.
--
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
Hi,
I hope someone in the mailing list has a good experience in server's
configuration requirement since I was not able to make a large scale
test to know Asterisk's configuration requirements for my application.
So, I'd like to know what kind of configuration the following
application
Hello,
I'm checking some Billing Software for Asterisk. In opensource I only
know (the name, I haven't used) AstBill. What other software should I
check with similar capabilities?
Thank you!
--
Carles Pina i EstanyGPG id: 0x8CBDAE64
http://pinux.info Manresa -
Hi,
I encountered the following problem:
My asterisk works as a gateway between two sip networks external (public
internet) and internal (local lan)
From public side asterisk is registered UA in external network.
Internal sip UAs are registered in local SIP Proxy.
When there is an incoming call
Carles,
You can find more info about the Open Source billing alternatives
in the voip-info wiki:
http://www.voip-info.org/wiki/view/Asterisk+billing
Regards,
Ariel
On Fri, 2008-01-25 at 12:26 +0100, Carles Pina i Estany wrote:
Hello,
I'm checking some Billing Software for Asterisk. In
2008/1/25, Carles Pina i Estany [EMAIL PROTECTED]:
Hello,
I'm checking some Billing Software for Asterisk. In opensource I only
know (the name, I haven't used) AstBill. What other software should I
check with similar capabilities?
Thank you!
--
Carles Pina i EstanyGPG id:
Hi Everyone,
I am having some issues getting my 7961 working with Trixbox. I have loaded
the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an
unprovisioned state. A status message shows up and says ³Error Verifying
Config Info².
I have read quite a bit on this topic (getting
Hello there,
I have set a simple environment to test some functionalities of
asterisk's new jitterbuffer.
The environment is composed of a sip softphone registering in asterisk
1.4 and calling a pstn phone connected to asterisk through a fxs
board.
Using the fixed buffer implementation the call
BJ,
Yeah, that's what I figured from the code. But I still can't get my
hard coded #'s to work.
The line:
number = 201XXX,5
(201XXX is a US based phone # I want dialed that I've obscured
with X's - in case that wasn't originally clear)
This line reacts the same way as the AstDB
Hi asterisk-users@lists.digium.com,
Add me as a friend on Last.fm so we can share our music taste :)
Check out what I'm listening to:
http://www.last.fm/user/shina01/?invitedby=shina01tp=ff_tp_b
I also sent you a personal note:
boo!
Signing up is free and takes less than a minute.
Classy.
On Jan 25, 2008 2:37 PM, Sina Owolabi [EMAIL PROTECTED] wrote:
Hi asterisk-users@lists.digium.com,
Add me as a friend on Last.fm so we can share our music taste :)
Check out what I'm listening to.
A personal note from me:
boo!
Signing up is free and takes less than a
Is there a minimum zaptel and libpri version for use with 1.6-beta1?
Thanks,
MC
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On Friday 25 January 2008 20:03:00 Michael Collins wrote:
Is there a minimum zaptel and libpri version for use with 1.6-beta1?
zaptel will remain at version 1.4 for the time being, but there is a 1.6-beta1
release of libpri.
--
Tilghman
___
--
Hi,
I tried to install Libpri-1.4.3 after downloading from sites- www.asterisk.com
and www.downloads.digium.com.
But in both the case the problem is coming AVC access denied. I am using
Fedora core 8. I asked this problem earlier and got advice to disable SELinux.
But many people adviced not
My magic orb is on the fritz. Can you give some more info? What
extension is ringing? What are you dialing to pick up? What does
your conf files look like?
I think I might know what the problem is, but I need a little more
info. Read core show application Pickup carefully, and then re-read
I just registered an Avaya 9620 set to my Astlinux system (0.47 - Asterisk
1.2.22), using Avaya SIP Firmware version 2.0.1.34.
Set [EMAIL PROTECTED] in the sip.conf
Found MWI worked immediately. Turned off as expected.
Have Fun!
Tom
___
-- Bandwidth
Hi,
I'm having problems with Directed PickUn and Asterisk 1.4.
Directed call pickup **EXT works ok with internal calls which are in the
same CONTEXT but,, with calls in which are from other context or
incoming calls from IVR this function doesn't work as is pointed in
I just checked the SIP debug when my 7960 registers and it looks like NAT is
enabled and working properly.
Does anyone have a 7961 on Asterisk that is going through NAT successfully?
-- SIP read from HOME IP ADDRESS:5061:
REGISTER sip:TB IP ADDRESS SIP/2.0
Via: SIP/2.0/UDP HOME IP
I have emailed Linksys about this, and they have not answered. I have
figured out much of how to do this, despite Linksys not being any
help. My only remaining issue is how to configure the PSTN line on a
SPA3102.
Try a file like this (example info included);
flat-profile
Thanks Chad. This config seemed to have worked a bit. I don't get the
Unprovisioned or Error Verifying Config Info messages anymore. However,
the phone sits at Registering and will never register.
I took a look at the sip debug and I see the below messages. Do I need to
enable NAT in the
Is there a setting that can add additional volume to
the Console/Dsp output?
Jerry
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On Fri, Jan 25, 2008 at 06:48:21PM +, Deepak Naidu wrote:
Before installing ensure selinux is disabled.
This should not be needed. If it is, it is a bug.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
Tzafrir Cohen wrote:
On Fri, Jan 25, 2008 at 02:08:02PM +0530, [EMAIL PROTECTED] wrote:
Hi,
I am using Fedora and I tried to install Libpri-1.4.3 but now I am getting
error AVC access denied. Its saying I need to disable SELinux protection.
I do not know what to do. Please help me out.
Before installing ensure selinux is disabled. Check the link below to
understand Selinux in Redhat/Fedora.
http://www.redhat.com/docs/manuals/enterprise/RHEL-5-manual/Deployment_Guide-en-US/ch-selinux.html
Check below link to disable selinux in Fedora, or google around for ur version
of
On Fri, Jan 25, 2008 at 02:08:02PM +0530, [EMAIL PROTECTED] wrote:
Hi,
I am using Fedora and I tried to install Libpri-1.4.3 but now I am getting
error AVC access denied. Its saying I need to disable SELinux protection. I
do not know what to do. Please help me out.
Again, what distrbution
Hi,
Try this
http://www.kcip.com/support/pap2uk.html
On Jan 25, 2008 4:18 PM, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
i need sample xml configuration files for linksys pap2, linksys pap-2t,
sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are
linksys/sipura products. So
I have found many neat scripts for my home asterisk on the wiki and
elsewhere - and we really like it. But there are a couple of things I'd
still like to find.
And if anyone has some favorites that think think are great for a home
with 2 adults and 3 kids (4 phones) 2 cell phones I'd like to
It appears as though SELinux is preventing you from moving forward.
Perform the following to disable SELinux.
cd /etc/selinux
vi config
change enabled to disabled
write your changes
reboot
Bryan M. Johns
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
Support: [EMAIL PROTECTED]
Hello List,
Not sure if this will be helpful but I made changes to the original
Cisco directory.php.txt script and applied them for use on the Polycom
phones. This will create an extension directory and alphabetize it
based on the sip registrations you have setup in sip.conf. Note that
this
I have a call coming in from Asterisk-A going to Asterisk-B where it's
determined that the called party is in fact yet another number in Asterisk-A
so a new call is created from B to A and the two calls bridged (by Asterisk)
at Asterisk-B.
Originating Caller == Asterisk-A == Asterisk-B ==
I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed
the WIKI page on setting it up but I can't seem to get it to work.
Here is my Asterisk version:
pbx1*CLI core show version
Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on
2008-01-10
12:08:48 UTC
Here is a
Well it would seem that Cisco chose not to make their SEP_MAC.cnf.xml format
standard across all of their phone models. I have had similar issues getting
various models to work. It has made it a challenge, for no obvious good
reason IMO. I have pasted a SEP_MAC.cnf.xml that I use for a 7941G-GE.
From: Raj Jain - Friday, January 25, 2008 10:07 AM
I'm trying to implement a Voice Drop service within Asterisk
dial-plan. The service is supposed to work as following:
1. A initiates a call to B
2. The call is answered by B's answering machine
3. A hears the answering machine's greeting
Try this configuration file...
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configu
ration+files+for+SIP#Cisco7961with833SR2ConfigurationExamples
Chad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory
Wong
Sent: Friday,
I have done a page with at least a hundred phones before. It took
about a full second for the mysql script to run and all the phones to
join the conference, but worked fine. We typically only page 60
phones at once. In the coming months, I will be attempting a page
with 250 phones.
I can confirm that. 110 phones should not be a problem. We've done paging
groups at that size. The only noticeable issue is the delay. When the initiator
starts to speak, there may be up to a 1 second delay for all the phones to
receive the audio. However, you probably wouldn't notice that
Hi,
I'm trying to implement a Voice Drop service within Asterisk
dial-plan. The service is supposed to work as following:
1. A initiates a call to B
2. The call is answered by B's answering machine
3. A hears the answering machine's greeting and the recording beep
4. A speaks a few words into
Mike Coakley wrote:
I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed
the WIKI page on setting it up but I can't seem to get it to work.
Here is my Asterisk version:
pbx1*CLI core show version
Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on
You are usinfg sip or iax ? Its possible to prevent in both cases for sip
under peer definition you can put canreinvite=no and in iax2 you can put
transfer=no for asterisk 1.4 or notranser=yes in asterisk 1.2 .. Search for
this on voip-info.org wiki for more info .
On Jan 25, 2008 7:03 PM, [EMAIL
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