Hi !
I have the same problem with the latest asterisk and FFA.
Does anybody know how to eliminate the situations when this problem occurs?
I looked in the Digium download section and found just old
res_fax_digium version 1.2.0.
Is there any chance to get driver update?
Thanks!
--
*Ilmars*
On Tue, 13 Jul 2010, Paul Belanger wrote:
On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellens jonas.kell...@telenet.be
wrote:
I have no licenses and I want to avoid transcoding all together.
For terminating a call into Asterisk, you need g729 licenses. It is
that simple.
The sounds package is
On 07/14/2010 08:55 AM, Gordon Henderson wrote:
On Tue, 13 Jul 2010, Paul Belanger wrote:
On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellensjonas.kell...@telenet.be
wrote:
I have no licenses and I want to avoid transcoding all together.
For terminating a call into Asterisk,
Thermal Wetland wrote:
I have a virtual server with godaddy but can not compile DAHDI as it
complains that I do not have the correct kernel source.
The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686:
Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and
latest
Hi.
The best easy way is:
copy kernel-devel-2.6.18-028stab064.7.rpm to /usr/src
then run rpm -ivh kernel-devel-2.6.18-028stab064.7.rpm
2010/7/14 Gareth Blades list-aster...@skycomuk.com
Thermal Wetland wrote:
I have a virtual server with godaddy but can not compile DAHDI as it
complains
Hi
Check your kernel version using *uname -r *and then try to download tar.gz
setup for that version.
And extract it into /usr/src/kernels directory , then try to compile.
--
Regards,
Chandrakant Solanki
On Wed, Jul 14, 2010 at 1:46 PM, Gareth Blades
list-aster...@skycomuk.comwrote:
Hi,
probably a misconfiguration or you havent plugged the cable in yet.
2010/7/14 C F shma...@gmail.com
It has nothing to do with the D-channel, however you will never know
if the B-channels work if the D-channel is down. D-channel is what
allows the B-channels to work, and is the first place
Hi
If you install rpm from any location it goes to its default location.
You just go for above steps. For kernel you can go for http://kernel.org
--
Regards,
Chandrakant Solanki
On Wed, Jul 14, 2010 at 2:06 PM, liuxin nyliuxin...@gmail.com wrote:
Hi.
The best easy way is:
copy
Thank you for your response Doug,
Please move the thread as you think appropriate (but please tell me
how/where to join the mailling list (as this is the only one I have
subscribed).
I have 2 physical Fax machines connected to the GXW and people will be
sending faxes old fation from thembut
No one have, at least, an idea ?
On 07/12/2010 05:36 PM, Xavier wrote:
Hi guys,
I've got a question about chanspy and meetme.
I'd like to transfer all the persons involved in a chanspy (the guy
spying, the guy that is spied and the guy that is speaking to the
spied one - total: 3) in a
hello
I found silence RTP packet from Asterisk in early dialog.
I want to know reason and how to solve.
RTP packet
80 00 40 22 00 0c 74 58 06 98 eb 44 ff ff ff ff ..@..tX...D
0010 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff
0020 ff ff ff ff ff ff ff ff ff ff ff ff
hello
I want to know how to pass through 100rel header.
and I hope that asterisk PRACK to UAS.(RFC3262 behavior)
_
_
Hello Asterisk community,
I'm trying to use BLF with Asterisk Realtime, i've been searching for
some info but nothing seems to be clear, can anyone help me eith some
ideas to make this work ok?
I'va my dialplan with Realtime
Thanks in advance
--
Saludos
Danny Dias
SkypeID: danny.dias1
--
On Wed, 14 Jul 2010, Jonas Kellens wrote:
On 07/14/2010 08:55 AM, Gordon Henderson wrote:
On Tue, 13 Jul 2010, Paul Belanger wrote:
On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellensjonas.kell...@telenet.be
wrote:
I have no licenses and I want to avoid transcoding all together.
For
Frank Church voi...@googlemail.com writes:
Is there a database of MAC address prefixes used the common VoIP
devices. I see the Linksys Sipura devices state with 00:0E.
Does the same apply to other Linksys VoIP equipment?
Is there some way VoIP equipment allow themselves to be identified by
On 07/14/2010 01:39 PM, Gordon Henderson wrote:
And it's nice to have a choice of vendors to buy G729 from now too.
Doesn't help on weedy hardware though.
Gordon
I thought you could only buy licenses from Digium ? Can you install
other G729-licenses on Asterisk ?
I need the
On 07/14/2010 05:15 AM, kawanobe tomohito wrote:
hello
I want to know how to pass through 100rel header.
and I hope that asterisk PRACK to UAS.(RFC3262 behavior)
Asterisk is not a proxy; it does not 'pass through' headers, or any
other portion of SIP requests and responses. Asterisk is a
On 14/07/10 12:17, Danny Dias wrote:
Hello Asterisk community,
I'm trying to use BLF with Asterisk Realtime, i've been searching for
some info but nothing seems to be clear, can anyone help me eith some
ideas to make this work ok?
I'va my dialplan with Realtime
Thanks in advance
Hi
I
On Wed, 14 Jul 2010, Jonas Kellens wrote:
On 07/14/2010 01:39 PM, Gordon Henderson wrote:
And it's nice to have a choice of vendors to buy G729 from now too.
Doesn't help on weedy hardware though.
I thought you could only buy licenses from Digium ? Can you install
other G729-licenses on
On Wed, Jul 14, 2010 at 10:45:33AM +0800, Malvin Rito wrote:
Thanks for the reply. There is no folder dahdi under /dev folder. I cannot
also find /udev.d on /etc folder.
Under /dev folder I only see /dev/zap/pseudo.
What version of Asterisk is it?
--
Tzafrir Cohen
On 07/14/2010 03:41 PM, Gordon Henderson wrote:
It's the default codec used in DECT phones. I trialled it for a while for
some backhaul applications - the users didn't notice anything different
and CPU overhead seemed very low, but I've since gone back to alaw. It
does save 32Kb/sec per call
On asterisk 1.4 using real-time, subscribecontext field never worked for me
and I have to add the hints in extensions.conf. But once there, they work
just fine.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-14 9:12 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
On 14/07/10 12:17, Danny Dias
I found this requirement very interesting because it is challenging, needs
some serious thinking on how to do it, but it is certainly possible. My idea
would be to record the sip channels which are involved in the spying process
and use a dynamic feature, pressing which would generate a conference
On Wed, 14 Jul 2010, Jonas Kellens wrote:
On 07/14/2010 03:41 PM, Gordon Henderson wrote:
It's the default codec used in DECT phones. I trialled it for a while for
some backhaul applications - the users didn't notice anything different
and CPU overhead seemed very low, but I've since gone
- Original Message -
On 07/12/2010 05:36 PM, Xavier wrote:
I've got a question about chanspy and meetme.
I'd like to transfer all the persons involved in a chanspy (the guy
spying, the guy that is spied and the guy that is speaking to the
spied one - total: 3) in a conference room.
- Original Message -
How do I specify to which parking lot the hints refer to?
For exemple, on 1.4 I use this to see whether a call is parked in 800:
exten = 800,hint,park:8...@parkedcalls
But on 1.6 I have multiple parking lots working apparently
sucessfully. How do I build
I am stuck with the same problem but I have used asterisk yum repository and
it worked by itself without me worrying for kernel stuff.
However, I need to install speex codec and now I am stuck as it doesn't get
picked up by the yum asterisk install somehow. I have lib speex and speex
already
I totally agree with the barge mode but for future evolution, what
about if there is more than 3 people ?
On 07/14/2010 04:36 PM, Russell Bryant wrote:
- Original Message -
On 07/12/2010 05:36 PM, Xavier wrote:
I've got a question about chanspy and meetme.
I'd like to transfer all
Hi Guys,
Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and
6730i, but none of them indicate the voic-email. Where should I look for
trouble to find the root issue for MWI?
Thanks,
--
_
-- Bandwidth
On 07/12/2010 05:36 PM, Xavier wrote:
I've got a question about chanspy and meetme. I'd like to transfer all
the persons involved in a chanspy (the guy spying, the guy that is spied
and the guy that is speaking to the spied one - total: 3) in a
conference room. Is there a way to do it
Asterisk 1.4.32
dahdi-2.3.0.1
Centos 5.5
Digium TE420
CAC channel bank (2)
Cisco RVS4000 router
Cox 50 Mbps/ 5 Mbps cable modem
Flowroute provider
codac G-711
90 % CPU idle
callwaiting=no
When there are 10-15 or more calls up the farend hears a callwaiting
like beep every 3 to 6 sec. the
bruce bruce wrote:
Hi Guys,
Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i,
and 6730i, but none of them indicate the voic-email. Where should I look
for trouble to find the root issue for MWI?
Thanks,
For each extension in sip.conf I have :-
On Wed, Jul 14, 2010 at 10:04 AM, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and
6730i, but none of them indicate the voic-email. Where should I look for
trouble to find the root issue for MWI?
(1) Check from the
On Wed, Jul 14, 2010 at 09:27:29AM -0700, Steve Casto wrote:
Asterisk 1.4.32
dahdi-2.3.0.1
Centos 5.5
Digium TE420
CAC channel bank (2)
Cisco RVS4000 router
Cox 50 Mbps/ 5 Mbps cable modem
Flowroute provider
codac G-711
90 % CPU idle
callwaiting=no
When there are 10-15 or more calls
Thanks for the input guys. I don't use .xml files for Aastra. Everything is
done on the UI.
#voicemail show users:
*ContextMbox User Zone NewMsg*
*|default007 Alex 2*
*default2100 Peter
Hello again!
Just info what we discovered if anybody gets the same problem.
The reason is fax file (.tiff) resolution.
If you try to improve fax quality by raising resolution then * crashes
with core dump.
Best,
--
*Ilmars*
--
Hi
I have a TDM400 and 4 channels of HPEC. I don't use the POTs lines
much so I didn't realize it wasn't working. This morning I was
watching the console and noticed that the echo canceller didn't load
when a call came in. /etc/dahdi/system.conf showed mg2 for all 4
channels. I changed them
On 07/14/2010 01:16 PM, Ilmars Knipšis wrote:
Hello again!
Just info what we discovered if anybody gets the same problem.
The reason is fax file (.tiff) resolution.
If you try to improve fax quality by raising resolution then * crashes
with core dump.
This has already been fixed in
At 11:23 AM 7/14/2010, you wrote:
Is there something I need to do with HPEC to make sure the
dahdi_genconf generates a proper system.conf or is there somewhere
else I show tell asterisk to use HPEC?
Well, Moments later I found /etc/dahdi/genconf_parameters which
seems to solve the problem.
Hi Everyone,
Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones,
how can one receive distinctive ring tones for INTERNAL calls ONLY?
Even though FreePBX Inbound has an option for Alert_INFO but that doesn't
work when the call comes into an IVR or Queue. The calls has to go
Hello list,
using asterisk 1.4.30.
When setting up the MySQL table 'musiconhold' as described in
http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf ,
what is the meaning of the fields :
`*digit*` char(1) NOT NULL default '',
`*sort*` varchar(16) NOT NULL default '',
Hi,
Might be off topic but I thought it would be a good place to ask.. I am
investigating switching to a hosted PBX and dumping my old Asterisk box
thats been running in my office for the last few years.. The few I have
found seem very expensive..
Can anyone point me to any VoIP PBX hosts in the
Using Asterisk 1.6.1.14 and dahdi 2.2.0.2+2.2.0.
We're placing outbound calls over an analog line. Some of these calls
are going to cell phones that play music rather than providing a
standard ring. As a result, the Dial command sometimes returns a
DIALSTATUS of CHANUNAVAIL and sometimes it
At 11:44 AM 7/14/2010, you wrote:
Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra
phones, how can one receive distinctive ring tones for INTERNAL calls ONLY?
It's ugly, but you could give the phone two different SIP IDs and
give those different ringtones.
Ira
--
Thanks for the input but that won't be good because people are not going to
remember two extensions for one person.
The sip header should be able to carry alert_info to internal extensions
really easily. Anyone else got a thought?
Thanks again,
On Wed, Jul 14, 2010 at 5:44 PM, Ira
Is it possible to send a test message to the IP 330 or 550 polycom
phones with asterisk?
Thanks,
Jerry
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Call progress (is only experimental), relies on defined ring tones, coloured
ring (music) messes this up.
in chan_dahdi.conf
callprogress=no
busydetect=yes
busycount=4
and possibly if your incoming analog lines support it.
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
_
At 03:05 PM 7/14/2010, you wrote:
Thanks for the input but that won't be good because people are not
going to remember two extensions for one person.
That's why there's a dialplan. But the piece I'm unsure of is how the
second SIP address handles more than one call.
Ira
--
We do hosted VoIP
www.pack-net.co.uk
Contact me off list for more details if it sounds right for you
Ish
On 14/07/10 22:27, Wipe_Out wrote:
Hi,
Might be off topic but I thought it would be a good place to ask.. I
am investigating switching to a hosted PBX and dumping my old Asterisk
box
On Wed, Jul 14, 2010 at 10:27:13PM +0100, Wipe_Out wrote:
Might be off topic but I thought it would be a good place to ask.. I am
investigating switching to a hosted PBX and dumping my old Asterisk box
thats been running in my office for the last few years.. The few I have
found
On Wed, Jul 14, 2010 at 4:55 AM, bruce bruce bruceb...@gmail.com wrote:
I am stuck with the same problem but I have used asterisk yum repository and
it worked by itself without me worrying for kernel stuff.
However, I need to install speex codec and now I am stuck as it doesn't get
picked
I'm happy to hear it worked out so well with so little. :)
On Wed, Jul 14, 2010 at 12:39 AM, bruce bruce bruceb...@gmail.com wrote:
Thanks for the input guys. For other refrence, a CyberData Voip Amplifier
which supplies 10 Watt to each of the two bogen 30 Watt speakers did the job
for a
On Wed, Jul 14, 2010 at 5:03 AM, liuxin nyliuxin...@gmail.com wrote:
Hi,
probably a misconfiguration or you havent plugged the cable in yet.
OMG you are right, I forgot to plug in the cable. Hey but wait which
cable you talking about?
2010/7/14 C F shma...@gmail.com
It has nothing to do
Hello,
I am using asterisk manager interface (http) for originating calls.
How can I get the name of the channel which is created by originate? I
want to use this channel for other manager commands like Atxfer,
Monitor, Hangup etc.
If I do action=originate, channel=SIP/200 then it creates a
Hi All,
After getting licences for Skype for asterisk a while ago I finally got
around to setting up a server with two channels and setting up a bcp on
the skype end.
The idea behind this is the following:
Users can dial into the PBX, get authenticated and only after
authentication get
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