HI ,
Is there Any way is there so that I can store my recordings directly to
a database rather storing the same to a file .
Thanks in advance .
Regards
Mahesh
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.
Hi..
We are facing a problem that is making the channel to be stuck. we are using
asterisk 1.4.22.1 version, and we have a direct sip trunk...We have 2 queues
and one has 2 agents and the other 5 agents, from last week the second
queue's channel is getting stuck, it happened 3 times till now and t
Hello,
This is what what I see after a Yum install asterisk16 asterisk16-config
freepbx:
Use of uninitialized value in string ne at
/var/www/html/panel/op_server.plline 4997.
Use of uninitialized value in substitution (s///) at /var/www/html/panel/
op_server.pl line 5439.
Use of uninitialized val
Calls are not going outside of the network. I had to setup up the subnet of
the other side (openvpn client) as the localnet of the Asterisk server for
Asterisk to not handle it with NAT or hand shake it with external IP.
Thanks,
-Bruce
On Wed, Sep 22, 2010 at 1:58 PM, Paul Belanger wrote:
> On
22.09.2010 16:08, Philipp von Klitzing пишет:
> Hi Dmitry!
>
>
>
Hello!
> And the third hit in my google result is this:
>
> http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html
>
> Since I mentioned in my previous message that you will find the answer in
> the archive of this
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Cunningham
Sent: Wednesday, September 22, 2010 4:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Recording maximum time and stop on silence
All,
Two questions:
1. Is there a limit on how long a call can be recorded for? For example is 4
hours a problem?
2. Can recording be stopped after a configured period of silence?
Thanks in advance,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of amit salunkhe
Sent: Wednesday, September 22, 2010 3:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk- speech to text(Voicemail to text
message)
Dear All
Ca
Dear All
Can you let me know is this possible to if we are using Asterisk version 1.4
or 1.6 for incoming voicemail we can send as email in text formta. Means
voice mesage converted into text message & send it to resp. email ids. is
this possible.
If yes. we can do the same with help of Asterisk
Hi!
> I need the system to be resilient to any network partition, so that
> anyone can send announces from any mic to all the reachable clients.
> I'd need also to page a subset of all the speakers.
Most of the major phone vendors (that are employed by the users of this
list) have support for m
With a proper setup and asynchronous dialing, this can be done in a
relatively seamless (although not as simple as this indicates) fashion.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asteri
On Wed, 22 Sep 2010, Matteo Fortini wrote:
> I'm building a paging system composed of roughly 10 switches in daisy
> chain, with an embedded box with a speaker and a microphone for each
> switch. The embedded box runs my software.
>
> I need the system to be resilient to any network partition, so
Hello
I recently heard this should be possible. Has anyone experience with this?
Thanks!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every T
On Wed, Sep 22, 2010 at 1:46 PM, bruce bruce wrote:
> Thanks, but Carlos Chavez was right on point. This fixed the problem:
> externip=123.123.123.123
> localnet=192.168.100.0/255.255.255.0
> nat=no in each extension.
>
So now I am confused, If you have a VPN setup between sites, why are
calls goi
Thanks, but Carlos Chavez was right on point. This fixed the problem:
externip=123.123.123.123
localnet=192.168.100.0/255.255.255.0
nat=no in each extension.
Maybe combination of both or only the localnet just fixed it.
Thanks,
Bruce
On Wed, Sep 22, 2010 at 1:35 PM, Steve Edwards wrote:
> Un-
Un-top-posting...
> On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce
> wrote:
> > Any feed back is appreciated.
> On Wed, Sep 22, 2010 at 9:49 AM, Paul Belanger
> wrote:
> Then configure you endpoints to use the 192.168.100.0/24 network. This
> is not an Asterisk issue, since your A
On Wed, Sep 22, 2010 at 09:50:00AM -0700, Steve Edwards wrote:
>
> Still, for scripting and portability, I'd recommend specifying the
> "decompressor" and using the long option form:
>
> tar\
> --list\
> --[un]gzip\
> --file\
>
Thanks for that Carlos. I am playing with that right now. What do you
suggest localnet should say?
Server A = OpenVPN Server:
localnet=127.0.01
localnet=192.168.100.0/255.255.255.0
Where 192.168.100.0 is the DHCPd subnet of Server B (the openvpn client)
Server A doesn't have any localnet other t
I don't think it's an endpoint issue. I think the SIP packet headers get
over-written by the tunnel (openvpn) protocol.
Thanks,
Bruce
On Wed, Sep 22, 2010 at 9:49 AM, Paul Belanger wrote:
> On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce wrote:
> > Any feed back is appreciated.
> >
> Then configu
On 10-09-22 11:45 AM, Klaus Darilion wrote:
> Hi!
>
> Since some time the download of the newest Asterisk does not contains
> the version number anymore, but is just called "asterisk-1.4-current.tar.gz"
>
> This gives me a tarball where I do not know the version without looking
> into the tarball.
>> On Wed, 22 Sep 2010, Jose P. Espinal wrote:
>>
>>> If you are using a script you could get the version with something like:
>>>
>>> tar -tf asterisk-1.4-current.tar.gz | head -n1
> On 09/22/2010 11:20 AM, Steve Edwards wrote:
>> You need a '-z' in there.
On Wed, 22 Sep 2010, Kevin P. Fleming
I'm building a paging system composed of roughly 10 switches in daisy
chain, with an embedded box with a speaker and a microphone for each
switch. The embedded box runs my software.
I need the system to be resilient to any network partition, so that
anyone can send announces from any mic to all
Oh, my bad.
It my box there might be some defaults predefined, as it did not yield
any errors.
Steve Edwards wrote:
> On Wed, 22 Sep 2010, Jose P. Espinal wrote:
>
>> If you are using a script you could get the version with something like:
>>
>> tar -tf asterisk-1.4-current.tar.gz | head -n1
On 09/22/2010 11:20 AM, Steve Edwards wrote:
> On Wed, 22 Sep 2010, Jose P. Espinal wrote:
>
>> If you are using a script you could get the version with something like:
>>
>> tar -tf asterisk-1.4-current.tar.gz | head -n1
>
> You need a '-z' in there.
Modern versions of 'tar' auto-detect gzip an
On Wed, 22 Sep 2010, Jose P. Espinal wrote:
> If you are using a script you could get the version with something like:
>
> tar -tf asterisk-1.4-current.tar.gz | head -n1
You need a '-z' in there.
--
Thanks in advance,
-
Ste
Is there a documentation about the CEL format?
l.
2010/9/22 Steve Murphy
>
> CEL was my answer, built on the channel event goodness that Russell. It's
> now in 1.8; but it
> lacks a converter to CDRs. You *could* just use the string of events coming
> out of CEL, but...
> I'd love to see your
Hi Klaus,
If you are using a script you could get the version with something like:
tar -tf asterisk-1.4-current.tar.gz | head -n1
Regards,
Klaus Darilion wrote:
> Hi!
>
> Since some time the download of the newest Asterisk does not contains
> the version number anymore, but is just calle
On 09/22/2010 10:55 AM, Steve Howes wrote:
> On 22 Sep 2010, at 16:45, Klaus Darilion wrote:
>> Since some time the download of the newest Asterisk does not contains
>> the version number anymore, but is just called "asterisk-1.4-current.tar.gz"
>>
>> This gives me a tarball where I do not know th
On Wed, Sep 22, 2010 at 11:45 AM, Klaus Darilion
wrote:
> This gives me a tarball where I do not know the version without looking
> into the tarball.
>
Should be simple to do, since
http://www.asterisk.org/downloads/asterisk/releases/asterisk-1.8.0-betaX.tar.gz
currently redirects to the proper
Klaus Darilion wrote:
> Hi!
>
> Since some time the download of the newest Asterisk does not contains
> the version number anymore, but is just called "asterisk-1.4-current.tar.gz"
>
> This gives me a tarball where I do not know the version without looking
> into the tarball.
>
> Thus, IMO it
On 22 Sep 2010, at 16:45, Klaus Darilion wrote:
> Since some time the download of the newest Asterisk does not contains
> the version number anymore, but is just called "asterisk-1.4-current.tar.gz"
>
> This gives me a tarball where I do not know the version without looking
> into the tarball.
>
Hi!
Since some time the download of the newest Asterisk does not contains
the version number anymore, but is just called "asterisk-1.4-current.tar.gz"
This gives me a tarball where I do not know the version without looking
into the tarball.
Thus, IMO it would be very useful to switch back to o
Do you have a localnet statement in your sip.conf? That and using
nat=no will make sure Asterisk does not replace the IP address in the
Invite.
On Wed, 2010-09-22 at 01:27 -0400, bruce bruce wrote:
> Hi Everyone,
>
>
> I have setup an OpenVPN tunnel between Server A (running Asterisk) and
> Ser
That's probably what I'm going to have to do. Thanks.
> I suppose that merely removing ATA and asterisk from the middle, and
> plugging a pots line into a fax machine is out of the question.
>
>
--
_
-- Bandwidth and Coloc
Hi all,
i read about the TLS-RENEGOTIATION vulnerability:
http://www.educatedguesswork.org/2009/11/understanding_the_tls_renegoti.html
http://www.sslshopper.com/article-ssl-and-tls-renegotiation-vulnerability-discovered.html
www.phonefactor.com/sslgapdocs/Renegotiating_TLS.pdf
Does the Asterisk
On Wed, Sep 22, 2010 at 10:00 AM, Adam Moffett wrote:
> In the simplest terms I can think of, I'm going to describe what I want to
> do and I want to know if it's possible in the current version of asterisk.
>
> Can I take a T38 call from an ATA, convert that back to analog and have
> asterisk scr
On 09/22/2010 09:00 AM, Adam Moffett wrote:
> In the simplest terms I can think of, I'm going to describe what I want
> to do and I want to know if it's possible in the current version of
> asterisk.
>
> Can I take a T38 call from an ATA, convert that back to analog and have
> asterisk screech tha
On Wed, Sep 22, 2010 at 10:05 AM, Gustavo A. Gonzalez
wrote:
> Hi all! I'm configuring a digium tdm card in Costa Rica, every things
> works well, but calls don't hangup. I've tested setting up progzone=br
> but dont work. Thanks for any help.
>
Does you telco provide a disconnect tone? Most don't
Hi all! I'm configuring a digium tdm card in Costa Rica, every things
works well, but calls don't hangup. I've tested setting up progzone=br
but dont work. Thanks for any help.
Cheers!
--
Gustavo A. González
Dto. Telefonía VoIP
Despegar.com
54 (11) 5032-3500
ext. 3512
--
_
In the simplest terms I can think of, I'm going to describe what I want
to do and I want to know if it's possible in the current version of
asterisk.
Can I take a T38 call from an ATA, convert that back to analog and have
asterisk screech that out on a POTS line to a remote fax machine. Would
On Wed, Sep 22, 2010 at 7:58 AM, federico cabiddu
wrote:
> This did the trick for me but I don't know the implications of such change
> and if it is correct to manage it this way.
>
It might we worth following up with a developer on #asterisk-dev, then
submitting your patch to https://issues.aster
On Wed, Sep 22, 2010 at 9:21 AM, IMS wrote:
> Do you have any ideas of the problem ? config.log don't give me more
> explanations.
>
Attach your config.log so we can see what is going on.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Free
On Wed, Sep 22, 2010 at 5:42 AM, Nikhil wrote:
> Anyone knows how to do cross compile asterisk 1.6.2.13 using
> mipsel linux.?
>
$ ./configure --help
Will output the flags you need to set.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pab
On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce wrote:
> Any feed back is appreciated.
>
Then configure you endpoints to use the 192.168.100.0/24 network.
This is not an Asterisk issue, since your Aastra 55i/2.5.2.1500 is
sending the INVITE message.
--
Paul Belanger | dCAP
Polybeacon | Consultant
J
Thanks for the feedback. I thought about that but it's not an option for me
right now.
Any other ways folks?
Thanks
On Wed, Sep 22, 2010 at 4:06 AM, Roger Burton West wrote:
> On Wed, Sep 22, 2010 at 01:27:07AM -0400, bruce bruce wrote:
> >I have setup an OpenVPN tunnel between Server A (runnin
A few corrections!
On Tue, Sep 21, 2010 at 6:32 PM, Steve Murphy wrote:
>
>
> On Tue, Sep 7, 2010 at 5:36 PM, Fabiano Carlos Heringer <
> b...@grupoheringer.com.br> wrote:
>
>> Em 07/09/2010 17:15, Miguel Molina escreveu:
>>
>> El 07/09/10 14:49, Fabiano Carlos Heringer escribió:
>>
>> Is there
Hi,
I can cross compile asterisk 1.4.21 on arm (imx27) using ltib
I want to cross compile the new version 1.6.2.13 but there is an error when
I execute the commands :
./configure --build=i686-pc-linux-gnu --host=arm
make menuselect
The configure seems ok, I have the result info :
*configure: Pac
>-Original Message-
>From: asterisk-users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
>Jonas Kellens
>Sent: Wednesday, September 22, 2010 9:04 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [asterisk-users] Unab
On 09/22/2010 02:45 PM, Philipp von Klitzing wrote:
> .slin is not .wav
>
Other files that are also in wav format play without any problem :
[Sep 22 15:02:35] -- Playing
'vm-youhave.slin' (language 'nl')
[r...@asterisk16 asterisk-1.6.2.10]# ls -l /var/lib/asterisk/sounds/nl/
total 388
>-Original Message-
>From: asterisk-users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
>Jonas Kellens
>Sent: Wednesday, September 22, 2010 8:26 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [asterisk-users] Unab
.slin is not .wav
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mail
On 09/22/2010 01:38 PM, Watkins, Bradley wrote:
> This is indicative that you have set the channel's language to something
> that expects there to be a singular and plural version of the 'new' (as
> in 'one new message' versus 'five new messages') sound.
>
> According to the code, that includes Dut
Hi Dmitry!
> > Have you considered using Google (or your favourite search engine)?
>
> Shure, I searched and find nothing.
> > The search terms "C" will surely help you, and in
> > fact point you to the very archive of this mailing list.
Don't know where this quote comes from, but "C" is absolu
Hi,
I'm working with asterisk 1.4.35 and found an issue regarding codecs
negotiation when T38 is enabled (t38pt_udptl=yes).
In particular if the INVITE sdp contains no allowed codec the call is not
rejected with "488 - Not acceptable here" but it goes through and the 200 OK
SDP is as follows:
v=0
This is indicative that you have set the channel's language to something
that expects there to be a singular and plural version of the 'new' (as
in 'one new message' versus 'five new messages') sound.
According to the code, that includes Dutch, Spanish, Portuguese and
Greek.
If you have one of th
22.09.2010 15:12, Andrea Cristofanini пишет:
>> Could you, please, give me link ? :-)
>>
> Google is not difficult to use... BTW
> http://www.voip-info.org/wiki/view/Asterisk+func+shared
>
>
There is no example here!
I already wrote about this...
--
___
> Could you, please, give me link ? :-)
Google is not difficult to use... BTW
http://www.voip-info.org/wiki/view/Asterisk+func+shared
--
---
Andrea Cristofanini Chief Technical Officer
ZeroZero39 srlTel
22.09.2010 14:50, Philipp von Klitzing пишет:
> Hi!
>
>
>> I see. I want to use SHARED function!
>> Do you have example how to
>> "to export them to the local call leg/channel "?
>>
> Have you considered using Google (or your favourite search engine)?
>
Shure, I searched and find not
Hi!
> I see. I want to use SHARED function!
> Do you have example how to
> "to export them to the local call leg/channel "?
Have you considered using Google (or your favourite search engine)?
The search terms "asterisk function shared" will surely help you, and in
fact point you to the very arch
On Wed, Sep 22, 2010 at 12:32:21PM +0200, Jonas Kellens wrote:
>[Sep 22 12:28:51] WARNING[22117]: file.c:650 ast_openstream_full:
>File vm-INBOXs does not exist in any format
>[Sep 22 12:28:51] WARNING[22117]: file.c:953 ast_streamfile: Unable
>to open vm-INBOXs (format 0x8 (alaw)): No such file o
Hello list,
it seems that a sound file is not present on my system, although I have
made a standard install...
[Sep 22 12:28:51] WARNING[22117]: file.c:650 ast_openstream_full: File
vm-INBOXs does not exist in any format
[Sep 22 12:28:51] WARNING[22117]: file.c:953 ast_streamfile: Unable to
Hi
Anyone knows how to do cross compile asterisk 1.6.2.13 using
mipsel linux.?
Thanks
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory we
21.09.2010 18:57, Philipp von Klitzing пишет:
> Hi!
>
>
>> Could somebody tell me how to use SHARED function?
>>
> http://www.voip-info.org/wiki/index.php?page=Asterisk+func+shared
>
>
There are no examples there :-(
>> I want to get RTCP stats from SIP, but current channel is DAHDI.
On Wed, Sep 22, 2010 at 01:27:07AM -0400, bruce bruce wrote:
>I have setup an OpenVPN tunnel between Server A (running Asterisk) and
>Server B suppling it's SIP Phones with DHCP pool of IPs.
Have you considered running Asterisk on Server B as well, and using IAX
to trunk between them? This is work
64 matches
Mail list logo
