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
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.
How can I
Hi
Anyone knows how to do cross compile asterisk 1.6.2.13 using
mipsel linux.?
Thanks
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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 or
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 archive of
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 nothing.
The
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...
--
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
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
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 absolutely not
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 Dutch,
.slin is not .wav
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http://www.asterisk.org/hello
asterisk-users
-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] Unable to
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] -- SIP/testcorp6- Playing
'vm-youhave.slin' (language 'nl')
[r...@asterisk16 asterisk-1.6.2.10]# ls -l
-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] Unable to
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:
A few corrections!
On Tue, Sep 21, 2010 at 6:32 PM, Steve Murphy m...@parsetree.com 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
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 ro...@firedrake.orgwrote:
On Wed, Sep 22, 2010 at 01:27:07AM -0400, bruce bruce wrote:
I have setup an OpenVPN tunnel between
On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce bruceb...@gmail.com 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
On Wed, Sep 22, 2010 at 5:42 AM, Nikhil d.nik...@cem-solutions.net 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:
On Wed, Sep 22, 2010 at 9:21 AM, IMS ims77@gmail.com 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:
On Wed, Sep 22, 2010 at 7:58 AM, federico cabiddu
federico.cabi...@gmail.com 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
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.
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
--
On Wed, Sep 22, 2010 at 10:05 AM, Gustavo A. Gonzalez
ggonza...@despegar.com 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
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 that out
On Wed, Sep 22, 2010 at 10:00 AM, Adam Moffett a...@plexicomm.net 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
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
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
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
Server
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
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.
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 would be
On Wed, Sep 22, 2010 at 11:45 AM, Klaus Darilion
klaus.mailingli...@pernau.at 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
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 the version
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 called
Is there a documentation about the CEL format?
l.
2010/9/22 Steve Murphy m...@parsetree.com
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
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,
-
Steve
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 and bzip
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
You
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
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 wrote:
Modern
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.
Thus,
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 paul.belan...@polybeacon.com
wrote:
On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce bruceb...@gmail.com wrote:
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
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\
Un-top-posting...
On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce bruceb...@gmail.com
wrote:
Any feed back is appreciated.
On Wed, Sep 22, 2010 at 9:49 AM, Paul Belanger paul.belan...@polybeacon.com
wrote:
Then configure you endpoints to use the 192.168.100.0/24 network. This
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
On Wed, Sep 22, 2010 at 1:46 PM, bruce bruce bruceb...@gmail.com 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,
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
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 that
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
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
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
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
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
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
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 list you
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
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
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
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