First off, you should post a new message rather than replying to an
existing message.
On Tue, Jul 28, 2009 at 08:18:42PM -0400, John F. Ervin wrote:
> Never having actually rolled an Asterisk (Trixbox in my case) system
> into production. I was wondering if in most peoples opinion if given
>
On Mon, Jul 27, 2009 at 01:28:59PM -0300, Gustavo A Gonzalez wrote:
> Hello all! Im running asterisk 1.4.23 and sometimes it crashes. Because I
> need to look for what asterisk crashes I run asterisk with option '-g' for
> debugging purpose. When I search for core files in filesystem nothing
> hap
Hello,
I just installed asterisk 1.6.1.1, asterisk-addons 1.6.1.1 and Asterisk-GUI
2.0 on Ubuntu 9.04 from *source.
*Asterisk-GUI's web interface doesn't appear from this URL
http://localhost:8088/asterisk/static/config/index.html*. *It says
Not Found
The requested URL was not found on this serv
Dear All
Regarding our opened case , can you please confirm if our attached
extensions.conf file can fullfil the needs of sending the subs dialed digits
one-by-one instead of sending it as an whole packet ?
Regards
H.Motamedi
extensions.conf
Description: Binary data
__
Hello,
Your smtp server is on?
Best regards
Carlos Rojas
On Mon, Jul 6, 2009 at 7:30 PM, Steve Anness wrote:
> Today I discovered that voicemail attachments are not working on our
> latest asterisk server (version 1.4.24.1). I have two other asterisk
> servers that I maintain but I didn’t
On Tue, Jul 28, 2009 at 9:24 PM, Tseveendorj wrote:
> Hi David,
>
> Thank you for your response.
>> Ummm, what kind of peer?
> I mean peer = VoIP providers or SIP working equipment. for example Cisco
> AS5350.
>
> If I want to connect VoIP provider anywhere in the World. I think most
> VoIP provide
On Tue, Jul 28, 2009 at 8:59 PM, David Backeberg wrote:
> On Tue, Jul 28, 2009 at 8:18 PM, John F. Ervin wrote:
> > Never having actually rolled an Asterisk (Trixbox in my case) system into
> > production. I was wondering if in most peoples opinion if given the
> choice
> > would rather have a st
On Tue, Jul 28, 2009 at 9:13 PM, Miguel Molina wrote:
> John F. Ervin escribió:
> > Never having actually rolled an Asterisk (Trixbox in my case) system
> > into production. I was wondering if in most peoples opinion if given
> > the choice would rather have a straight VOIP/SIP system or would
>
On Tue, Jul 28, 2009 at 9:13 PM, Miguel Molina wrote:
> John F. Ervin escribió:
> I'd go VoIP without thinking twice. We are on the 21st century!
Sailboats are ancient, but they're still a reliable way to get across
a body of water. Some people pay a very large amount of money for a
sailboat, and
Hi David,
Thank you for your response.
> Ummm, what kind of peer?
I mean peer = VoIP providers or SIP working equipment. for example Cisco
AS5350.
If I want to connect VoIP provider anywhere in the World. I think most
VoIP providers didn't use username and secret for trunk between partners.
If
John F. Ervin escribió:
> Never having actually rolled an Asterisk (Trixbox in my case) system
> into production. I was wondering if in most peoples opinion if given
> the choice would rather have a straight VOIP/SIP system or would
> rather have a system with normal POTS/analog types lines and
On Tue, Jul 28, 2009 at 8:53 PM, Tseveendorj wrote:
> Hello,
>
> I'm novice on the SIP protocol also on Asterisk. Could someone explain
> me why the Asterisk is using username and secret config on peer connection?
Because with any authentication, any system that can connect to your
system can init
On Tue, Jul 28, 2009 at 8:18 PM, John F. Ervin wrote:
> Never having actually rolled an Asterisk (Trixbox in my case) system into
> production. I was wondering if in most peoples opinion if given the choice
> would rather have a straight VOIP/SIP system or would rather have a system
> with normal
On Tue, Jul 28, 2009 at 8:18 PM, John F. Ervin wrote:
> Never having actually rolled an Asterisk (Trixbox in my case) system into
> production. I was wondering if in most peoples opinion if given the choice
> would rather have a straight VOIP/SIP system or would rather have a system
> with norma
Hello,
I'm novice on the SIP protocol also on Asterisk. Could someone explain
me why the Asterisk is using username and secret config on peer connection?
Does Asterisk can send call to peer without username and secret
configuration ?
Sincerely,
Tseveen.
___
Never having actually rolled an Asterisk (Trixbox in my case) system
into production. I was wondering if in most peoples opinion if given
the choice would rather have a straight VOIP/SIP system or would rather
have a system with normal POTS/analog types lines and something like a
digium card?
I have a carrier who tells me he will be sending me traffic from a wide
range of IP addresses.
so I set up a realtime peer as follows:
[peer]
defaultip=xxx.xxx.xxx.xxx
host=xxx.xxx.xxx.xxx
deny=0.0.0.0/0.0.0.0
allow=xxx.xxx.xxx.0/255.255.255.0
insecure=port,invite
Yes, he's really claiming to o
Benny Amorsen wrote:
> Doug Lytle writes:
>
>> I don't suppose anybody has an updated CalledID patch for Asterisk 1.4.26:
>>
>> https://issues.asterisk.org/view.php?id=8824
>>
>> I've been running 1.4.21.1 for a while and have the need to update one
>> of our Asterisk Conference/Faxserver/switch
On Tuesday 28 July 2009 16:12:00 Jared Smith wrote:
> On Tue, 2009-07-28 at 15:32 -0500, Karl Fife wrote:
> > UNEXPECTED:
> > Channels 1 is unexpectedly assigned to the context outbound-pri
> > Channels 2-23 are 'properly' assigned to the context inbound-pri
> > Channels 25-47 are 'properly' assign
Dear All,
I'm trying to configure my new phone Cisco 7960 to work with asterisk.
I followed
http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html
and I got into the point where I can see on the the display line indication
showing
"55 " so it looks like the phone is not re
On Tue, 2009-07-28 at 15:32 -0500, Karl Fife wrote:
> My config works fine but I must be missing a concept because a small change
> gives an unexpected result. Can someone help me understand the
> chan_dahdi.conf parser that would explain this?
I'll do my best.
> Based on the config below, Chan
Doug Lytle writes:
> I don't suppose anybody has an updated CalledID patch for Asterisk 1.4.26:
>
> https://issues.asterisk.org/view.php?id=8824
>
> I've been running 1.4.21.1 for a while and have the need to update one
> of our Asterisk Conference/Faxserver/switches and wanted to see if the
>
My config works fine but I must be missing a concept because a small change
gives an unexpected result. Can someone help me understand the
chan_dahdi.conf parser that would explain this?
Based on the config below, Channels 1-23 are assigned to the context
inbound-pri, and Channels 25-47 are ass
John A. Sullivan III wrote:
> On Tue, 2009-07-28 at 13:33 -0500, Tom Poe wrote:
>
>> Any vitelity customers with pbxinaflash boxes? I'm able to call
>> in-house, but failing to make outbound calls. My assigned server at
>> vitelity is not reachable. I can ping to my ISP OK.
>> Any help appr
I don't suppose anybody has an updated CalledID patch for Asterisk 1.4.26:
https://issues.asterisk.org/view.php?id=8824
I've been running 1.4.21.1 for a while and have the need to update one
of our Asterisk Conference/Faxserver/switches and wanted to see if the
patch available would apply. It
Hello
I'm trying to use an AGI that returns the queues status (numbers of
available agents, etc ), but I'm having some problems with it (it's
still very buggy).
Is there any AGI repository with source code samples?
Had anyone used an AGI to check queues and agents status?
Thanks
regards
Joao Pere
On Tue, 2009-07-28 at 13:33 -0500, Tom Poe wrote:
> Any vitelity customers with pbxinaflash boxes? I'm able to call
> in-house, but failing to make outbound calls. My assigned server at
> vitelity is not reachable. I can ping to my ISP OK.
> Any help appreciated. Such as actually how to make
Any vitelity customers with pbxinaflash boxes? I'm able to call
in-house, but failing to make outbound calls. My assigned server at
vitelity is not reachable. I can ping to my ISP OK.
Any help appreciated. Such as actually how to make email contact with
support at vitelity. They're not resp
I have a mix of both realtime users and static users configured in the file.
Also, I'm using 'sip reload' to fix sip registrations that fail after a
while.
I guess this might be a bit messy.
nevertheless, I thought this is supposed to work based on the bugfix
https://issues.asterisk.org/view.php?i
From my experience sip reload always clears the realtime cache, what
exactly are you doing? Wouldn't doing a 'sip prune realtime peer/user'
for single peers/users be of use to you?
Ish
dan julius wrote:
> Hi,
>
> I'm using Asterisk 1.4.24.1
>
> Is it possible (and recommended) to have realtime
It depends what the carrier is looking for in the SIP header, so some
interop between you and your carrier is necessary in this regard. Last year
for a customer I had to setup *67 to hide caller id, and the carrier asked
me to use "Remote-Party-ID" or "P-Asserted-Identity" in the SIP header. I
did
Hi,
I'm using Asterisk 1.4.24.1
Is it possible (and recommended) to have realtime peers that are not cleared
from memory when 'sip reload' is issued?
According to https://issues.asterisk.org/view.php?id=14196 I thought having
rtcachefriends=yes would be enough, but this does't seem to work.
Than
The trick does seem to make sure you have the right hardware platform
underneath. I have high hopes for the Netgear boxes that should be in my
hands tomorrow. Will report...
j
On Tue, 28 Jul 2009, SIP wrote:
> I've had similar results to you. Packet loss even when not transcoding.
> Overall
That's exactly what I ended up doing:
SIPAddHeader(Remote-Party-ID:
\;privacy=full\;screen=yes)
Note the \ before each ; It wouldn't put anything behind the ; without them.
Thanks.
Original Message
Subject: Re: [asterisk-users] INVITE Privacy Information
From: Philipp Kempge
Yes, I have qualify=yes
Could this be related to various posts regarding DNS issues?
I doubt I have dns issues because the hostname and IP of the other server is
hard-coded in /etc/hosts
Thanks,
Dan
On Tue, Jul 28, 2009 at 3:38 PM, Ishfaq Malik wrote:
> Hi
>
> Have you tried setting qualif
I've had similar results to you. Packet loss even when not transcoding.
Overall poor performance across the board. We considered it a failed
experiment.
N.
Zoa wrote:
> I have played with DD-WRT on linksys wrt54g version 5 last week (2
> different ones, they are the model with less memory so
I have played with DD-WRT on linksys wrt54g version 5 last week (2
different ones, they are the model with less memory so i needed to use
the micro version). I tried to use it as a repeater. (might have
something to do with it)
So far i read reports on great succes everywhere, my experience ar
On Mon, 27 Jul 2009, Jeff LaCoursiere wrote:
>1) The latest 8.09 kamikaze no longer supports the Broadcom radios, so ...
>
Because of closed-source drivers the Broadcom chips only work on the 2.4
series kernels. OpenWRT does make a 2.4 kernel version _and_ a 2.6 kernel
version. Use the 2.4 and the
You just do a make && make install, providing your changes don't produce any
fatal syntax errors.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sriram
Sent: Tuesday, July 28, 2009 7:51 AM
To: asterisk-users@lists.digium.co
Hi List
Might be a very silly question, I want to make some changes in CDR.C of
Asterisk ( i m using trixbox) . I noticed that cdr.c is present inside the main
folder of svn asterisk 1.4 branch. If i make any changes in cdr.c how do i
update hte changes as i dont see a loadable module cdr.so ?
Hi
Have you tried setting qualify in the sip.conf?
http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
Ish
dan julius wrote:
> Hi,
>
> I have configure a SIP trunk between two asterisk 1.4.24.1
> After a while, sometimes a day or two, sometimes only a few hours, the
> SIP connection betwee
Miguel Molina wrote:
> Counting that everything works well on the IP portion of the
> communication, you might have something, but the "store and forward"
> process that has to be made twice to emulate a T.38 gateway on both
> asterisks would make it a very slow process to send a single fax, ha
Cyprus VoIP schrieb:
> I ran into this problem: When I change the CALLERID(num and name) to
> anonymous, they are also changed in the RPID line and not only in the From.
OK. I'd try to set sendrpid=no in sip.conf and then add a
Remote-Party-ID header in the dialplan.
SIPAddHeader(Remote-Party-ID:
Hello, all. Alas, another convoluted question. All the simple things
are, well, simple so I suppose we only need to trouble the list with
squirrely problems!
We've noticed a call history problem when using Asterisk where the call
history on the Snom phones (with which we are very pleased) reflec
Hi, sorry to bother u all, i have a trouble
when I call a did number forward to my asterisk server, the server told me:
[Jul 28 19:00:57] WARNING[28080]: chan_sip.c:3806 sip_write: Asked to
transmit frame type 256, while native formats is 0x4 (ulaw)(4) read/write =
0x4 (ulaw)(4)/0x4 (ulaw)(4)
[Ju
Hi,
I have configure a SIP trunk between two asterisk 1.4.24.1
After a while, sometimes a day or two, sometimes only a few hours, the SIP
connection between the two servers is lost.
'sip show peer status' shows the peer is unreachable.
'sip reload' resolves the problem, but I'm wondering if there
Hello,
i´ve a question about the Meetme Options. How could i play a enter and
leave sound but without recording the user name first. I just want
something like "User joined conference" and a "User leaved".
With the i or I Option i have to record the name first.
Is there any way of doing this? As
On Tue, Jul 28, 2009 at 06:05:29AM +0100, hadi motamedi wrote:
> Dear All
> It seems that our Asterisk pbx announcement files are being stored inside
> the "/var/lib/asterisk/sounds" folder . Can you please let us know what is
> the appropriate program to open and hear them on an MS Windows client
Thank you Philipp for your help.
I ran into this problem: When I change the CALLERID(num and name) to
anonymous, they are also changed in the RPID line and not only in the From.
This is the script:
exten => _*67.,1,SIPAddHeader(Privacy: id);
exten => _*67.,2,Set(CALLERPRES()=prohib_passed_screen
Hi,
We have been running asterisk in our telco interconnect box with ss7 and
H323 configured. Everything ran find till now, however, today, it started
crashing with the following messages:
[Jul 28 14:56:55] WARNING[2968]: acl.c:541 ast_ouraddrfor: Cannot create
socket
[Jul 28 14:56:55] ERROR[
Dear All
Can you please let us know how we can modify our outgoing extension routing
such that our subs can dial as "*21" for reaching to IN services . Please
find below our current item for outgoing dialing , as the followings :
"
[line-outgoing]
exten => _X.,1,macro(dialuser,Zap/g1/${EXTEN},${EXT
Hi i got the following script for calling lastcaller.
Script works fine.But on the cdr records calls seem to be made to *1
instead of the number on dial script.
How can i fix this??
exten = *1,1,Answer
exten = *1,2,Macro(user-callerid,)
exten = *1,3,Playback(last-num-to-call)
exten = *1,4,Set(numb
Thanks Philipp it works..
> Oguzhan Kayhan schrieb:
>> Problem is, if the user authenticates, * starts counting as billable
>> seconds even if i hangup the phone before the called party answers..And
>> also
>> as disposition.. it accepts all calls authenticated as 'answered'
>> If i commentout th
53 matches
Mail list logo
