On Tue, May 03, 2011 at 04:30:46PM -0400, Richard Kenner wrote:
Please create a mantis issue describing this problem.
Pardon my ignorance, but what does mantis refer to?
Useless trivia:
A mantis eats bugs. But sadly the bug tracker is now called issue
tracker. http://issues.asterisk.org/ ,
On 2011-05-03 16:32, Dean Hoover wrote:
I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an
HP ML110 G6 using Ubuntu Linux 10.04 LTS.
I have two Digium TE121 single T1 port cards and a Digium AEX800
8-port FXS card. All PCI Express cards.
Co-workers are hearing hissing
Hello List,
We are running two asterisk machines in virtual IP as primary and secondary
server.
Initially virtual IP will be active in primary server; during the failure of
primary secondary will get the virtual IP.
Is there any way to retrieve pending queue calls from primary to secondary, in
Hi Rajib,
I think It is not possible with asterisk , as primary server goes down it
will stop asterisk services so once asterisk service down i think all
connected calls to queue will hangup automatically, and you cannot retrive
those calls as they all are disconnected .
I think you need to
hi
you can add this in extenssion.conf
exten = 223,1,Answer()
exten = 223,2,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten = 223,3,Dial(SIP/223)
exten = 223,4,Hangup()
i can record without any issue in /var/spool/asterisk/monitor
2011/5/4 Bruce B bruceb...@gmail.com
Thanks for the
I am attempting to install Dahdi on a virtual machine running Centos 5.5 and
having various problems.
yum install kernel-devel gcc make gcc-c++ libxml2-devel
Loaded plugins: fastestmirror
Loading mirror speeds from cached hostfile
* base: mirror.optus.net
* extras: mirror.optus.net
* rpmforge:
10%
De: Matt Riddell li...@venturevoip.com
Para: asterisk-users@lists.digium.com
Enviadas: Quarta-feira, 4 de Maio de 2011 0:32:28
Assunto: Re: [asterisk-users] Fading voice problem
On 3/05/11 10:16 PM, Eduardo Leones wrote:
Guys,
I'm having problems in the
Look like codec mismatch issue.
--
Sent from my iPhone
On May 3, 2011, at 9:55 PM, Jerry Geis ge...@pagestation.com wrote:
Under 1.4.35 I get this message printed MANY times
[May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame
type 4, while native formats is 0x1000
Here is the bug report
https://issues.asterisk.org/view.php?id=19171
Please add a comment to the bug indicating that you are also experiencing the
issue with asterisk 1.4.35 to 1.4.41
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I've given up on trying T38 because there is no universal support for it...
Can someone recommend another way of faxing without using T38?
On Tue, May 3, 2011 at 5:13 PM, satish patel satish...@hotmail.com wrote:
Enable debug and verbose on CLI ?
Did you enable and also at logger.conf
full
Thanks for the input. I think that works as my other recordings work. I will
test that again regardless.
Is there no real other way to know why MixMonitor fails or look more into
it?
Regards,
Bruce
On Wed, May 4, 2011 at 5:03 AM, salaheddine elharit
salah.elharit...@gmail.com wrote:
hi
you
Did you try digim fax ?
Also you can record you incoming fax via mxmonitor and analize it.
--
Sent from my iPhone
On May 4, 2011, at 8:50 AM, vip killa vipki...@gmail.com wrote:
I've given up on trying T38 because there is no universal support
for it... Can someone recommend another way of
Hi everybody!
I have asterisk 1.8.3.3 with pickup deadlock avoidance patch applied in
production.
I'm use next dialplan construction
exten = *,1,PickUP(queue-number)
exten = *,2,PickUP()
Can anyone tell me how can I stop dialplan execution on the first
priority if that priority's pickup was
doesn't digium fax cost money?
On Wed, May 4, 2011 at 9:21 AM, Satish Patel satish...@hotmail.com wrote:
Did you try digim fax ?
Also you can record you incoming fax via mxmonitor and analize it.
--
Sent from my iPhone
On May 4, 2011, at 8:50 AM, vip killa vipki...@gmail.com wrote:
Digium has a 1-port Free Fax for Asterisk - FFA
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, May 04, 2011 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Single channel license is free.
--
Sent from my iPhone
On May 4, 2011, at 9:44 AM, vip killa vipki...@gmail.com wrote:
doesn't digium fax cost money?
On Wed, May 4, 2011 at 9:21 AM, Satish Patel satish...@hotmail.com
wrote:
Did you try digim fax ?
Also you can record you incoming fax via
meaning asterisk can receive only 1 fax at a time?
On Wed, May 4, 2011 at 9:47 AM, Satish Patel satish...@hotmail.com wrote:
Single channel license is free.
--
Sent from my iPhone
On May 4, 2011, at 9:44 AM, vip killa vipki...@gmail.com wrote:
doesn't digium fax cost money?
On Wed, May
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, May 04, 2011 8:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] receive faxes
meaning asterisk can
screw that i just got hylafax to work with IAXMODEM...i refuse to pay
digium a dime... supposed to be open-source right?
On Wed, May 4, 2011 at 9:52 AM, Danny Nicholas da...@debsinc.com wrote:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
Does Hylafax support T.38?
The free fax works just fine with DAHDI. I've never tried to do T.38 with that
since it seems like it would be complicated and not give me much over using
DAHDI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Why use T.38 when you can use ulaw ?
On Wed, May 4, 2011 at 10:12 AM, Eric Wieling ewiel...@nyigc.com wrote:
Does Hylafax support T.38?
The free fax works just fine with DAHDI. I've never tried to do T.38 with
that since it seems like it would be complicated and not give me much over
On Wed, May 4, 2011 at 10:12 AM, Eric Wieling ewiel...@nyigc.com wrote:
Does Hylafax support T.38?
The free fax works just fine with DAHDI. I've never tried to do T.38 with
that since it seems like it would be complicated and not give me much over
using DAHDI.
There is the t38modem[1]
Unless someone has broken something recently, you'll get better results
with spandsp than you get with the Digium FAX package.
Steve
On 05/04/2011 09:21 PM, Satish Patel wrote:
Did you try digim fax ?
Also you can record you incoming fax via mxmonitor and analize it.
--
Sent from my iPhone
or, may be, I should use +101 priority? but how?
WBR
A.Rymkus
04.05.2011 17:42, A.Rymkus пишет:
Hi everybody!
I have asterisk 1.8.3.3 with pickup deadlock avoidance patch applied
in production.
I'm use next dialplan construction
exten = *,1,PickUP(queue-number)
exten = *,2,PickUP()
Can
Read the UPGRADE*.txt fles.
+101 was deprecated in 1.4
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A.Rymkus
Sent: Wednesday, May 04, 2011 10:38 AM
To: asterisk-users@lists.digium.com
Subject: Re:
On Wed, 4 May 2011, vip killa wrote:
screw that i just got hylafax to work with IAXMODEM...i refuse to
pay digium a dime... supposed to be open-source right?
Great attitude. Should be worth about a bazillion bad karma points.
--
Thanks in advance,
Eric Wieling wrote:
Does Hylafax support T.38?
Not at this time, no. And, I have no need for it.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
Steve Edwards wrote:
Should be worth about a bazillion bad karma points
+1
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
Honestly Digium's Asterisk is not a quality project. Though it has lead the
way in innovative open-source VoIP, it's a flawed and chaotic project.
Hence, I refuse to pay Digium. Digium seems to make a bazillion dollars
off of these flaws by selling commercial support/addons anyway... so that
Non-T.38 faxing works well. I assume there are reasons you must use T.38.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, May 04, 2011 11:02 AM
To: Asterisk Users Mailing List -
On 4 May 2011, at 15:01, vip killa wrote:
screw that i just got hylafax to work with IAXMODEM...i refuse to pay
digium a dime... supposed to be open-source right?
There is so much wrong with that sentence, I don't know where to start.
On 4 May 2011, at 16:02, vip killa wrote:
Honestly
Non-T.38 faxing works reasonably well. I have some issues with some things
at Digium as well, but I'm not going to bite the hand that feeds me. I
assume that Digium takes most of the known bugs out of what they charge
folks for. From what I read, if you had to make a living on T.38 faxing,
On 4 May 2011, at 16:02, vip killa wrote:
Honestly Digium's Asterisk is not a quality project. Though it has lead
the way in innovative open-source VoIP, it's a flawed and chaotic project.
Hence, I refuse to pay Digium. Digium seems to make a bazillion dollars
off of these flaws by selling
On Wed, May 4, 2011 at 11:13 AM, Danny Nicholas da...@debsinc.com wrote:
Non-T.38 faxing works reasonably well. I have some issues with some things
at Digium as well, but I'm not going to bite the hand that feeds me. I
assume that Digium takes most of the known bugs out of what they charge
Look like this is what we need
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding
What is conditional and unconditional forwarding ?
Date: Tue, 3 May 2011 16:41:30 -0700
From: cwall...@lodgingcompany.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk
On 03/05/11 09:09, Robles Román, José Miguel wrote:
Perhaps using one-way hash functions
(http://en.wikipedia.org/wiki/Cryptographic_hash_function) like MD5 or SHA-x,
even if you get the file with passwords and the code that checks them, it would
be difficult to find a collision (a password
On Wednesday 04 May 2011, vip killa wrote:
Honestly Digium's Asterisk is not a quality project. Though it has lead the
way in innovative open-source VoIP, it's a flawed and chaotic project.
Hence, I refuse to pay Digium.
Don't worry. You can always get your money refunded if it breaks -- and
Hello,
I have been hired to fix a large and complicated installation using several
Kamailio and Asterisk servers.
I found that I require some extra modules on some of the Asterisk servers. I
was hoping to be able to compile only the modules needed and copy them to where
they should be.
snip
(For my part, I'm actually surprised that nobody came up with a proper
protocol for encapsulating the stream of zeros and ones that make up a fax
transmission but rely on the precise timing inherent with a circuit-
switched
network, into something more suitable for sending over a
On Wed, May 4, 2011 at 12:00 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
...
(For my part, I'm actually surprised that nobody came up with a proper
protocol for encapsulating the stream of zeros and ones that make up a fax
transmission but rely on the precise timing inherent with a
doug,
why are you shaking!?!?... do you have a better recommendation?
daveC
Doug Lytle wrote:
C F wrote:
model name : AMD-K6(tm) 3D processor
*shudder*
Doug
--
SJREIA South Jersey Real Estate Investors Association
Want to invest in Real Estate?
come out and join over 450 real
On 04/05/11 17:10, || dave cantera Mobile wrote:
doug,
why are you shaking!?!?... do you have a better recommendation?
daveC
AMD K6 CPU brings back some pretty bad memories from me too.
Doug Lytle wrote:
C F wrote:
model name : AMD-K6(tm) 3D processor
*shudder*
Doug
--
On Wed, 4 May 2011, Andrew Latham wrote:
Faxing is considered a legal method of
doing business in many areas. Maybe lobbing for more effective
digital signatures would help get faxing removed from our everyday
lives.
From dictionary.com:
lob1
[lob] Show IPA
verb, lobbed, lob·bing, noun
|| dave cantera Mobile wrote:
why are you shaking!?!?
The AMD K6 and the AMD K6-2 were (At that time) cheaper then what Intel
had to offer, I built many systems based on both.
Sorry, but that memory, along with the memories of running a BBS, just
made me shudder.
Doug
--
Ben
On Wed, May 4, 2011 at 12:00 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
(For my part, I'm actually surprised that nobody came up with a proper
protocol for encapsulating the stream of zeros and ones that make up a fax
transmission but rely on the precise timing inherent with a
On Wed, May 04, 2011 at 09:56:40PM +1200, CB wrote:
I am attempting to install Dahdi on a virtual machine running Centos 5.5 and
having various problems.
yum install kernel-devel gcc make gcc-c++ libxml2-devel
Loaded plugins: fastestmirror
Loading mirror speeds from cached hostfile
* base:
Dear folks,
We have recently installed A400D card with 12 FXO modules, the serer is HP
DL180 G6, cards works fine but after a while all the calls get an awful
noise, you can not get what each side says. The noise cleares as soon as we
restart wanrouter but not asterisk (i mean asterisk restart
Un-top-posting,
On Wed, May 04, 2011 at 10:01:37AM -0400, vip killa wrote:
On Wed, May 4, 2011 at 9:52 AM, Danny Nicholas da...@debsinc.com wrote:
*You are “Running before you learn to walk”! You can’t make T.38 work
(that’s ok, most other folks can’t either) but you want a free faxing
Relatively new to Asterisk and SIP and am trying to run a proof of
concept using Asterisk to make an outbound call through an Audiocodes
gateway via SIP using Asterisk version 1.6.1.12. The specific
requirements of the gateway in the configuration I am trying to use
specify that the Name part
Hey All
;satish testing
exten = 7778,1,Verbose(System crash when no extension specified in dial)
exten = 7778,2,Dial(SIP/)
*CLI == Using SIP RTP CoS mark 5
-- Executing [7778@from-sip:1] Verbose(SIP/7527-0003, System crash
when no extension specified in dial) in new stack
System
Just a follow-up in case somebody else sees this: I upgraded the Polycom phone
to the latest firmware, that did it. I had been on the same version for almost
a year without problems, so I don`t know if it`s the firmware version that was
the issue or simply formatting the phone to factory
paul, doug,
I had several AMD athlons 64bit... no problems running centos, suse.
they seem solid on 1.4.xx... had a few intel celerons and P4s. they
were good as well. guess I was Lucky back then!
thanks for supporting the list!
daveC
Paul Hayes wrote:
On 04/05/11 17:10, || dave cantera
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de
Paul Hayes
Enviado el: miércoles, 04 de mayo de 2011 17:55
Para: asterisk-users@lists.digium.com
Asunto: Re: [asterisk-users] Password to be ecrypted?
On 03/05/11 09:09, Robles Román,
On Wed, May 04, 2011 at 09:56:40PM +1200, CB wrote:
I am attempting to install Dahdi on a virtual machine running Centos
5.5 and
having various problems.
yum install kernel-devel gcc make gcc-c++ libxml2-devel
Loaded plugins: fastestmirror
Loading mirror speeds from cached hostfile
Issue created: https://issues.asterisk.org/view.php?id=19228
is there anybody could your please try this ??
-S
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Wed, 4 May 2011 17:12:29 +
Subject: [asterisk-users] asterisk-1.8 crash if no extension specified in
4 maj 2011 kl. 19.44 skrev Robles Román, José Miguel:
By the way, I like the implementation in iax.conf (auth=md5 ...
secret=x), it seems more flexible, and it enables the use of other hash
functions or other security algorithms.
The SIP protocol does not support any other hash
|| dave cantera Mobile wrote:
I had several AMD athlons 64bit
My Myth server is this and I have an AMD X2 for my desktop. Todays'
chips from AMD are great. It's just those K6 K6-2s
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
David Backeberg wrote:
On Wed, May 4, 2011 at 12:00 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
(For my part, I'm actually surprised that nobody came up with a proper
protocol for encapsulating the stream of zeros and ones that make up a fax
transmission but rely on the precise
I'm going to upgrade the BIOS and update dahdi to the latest and greatest first.
I did look at the link you sent when I first started this mission. It
was the basis of me looking at using ACPI to get the IRQs to change.
My maintenance window is tomorrow, so I'll let anyone who's interested
know
A J Stiles asterisk_l...@earthshod.co.uk writes:
(For my part, I'm actually surprised that nobody came up with a proper
protocol for encapsulating the stream of zeros and ones that make up a fax
transmission but rely on the precise timing inherent with a circuit-switched
network, into
On Wed, May 4, 2011 at 12:10 PM, John Hablitzel jjblitz...@gmail.comwrote:
Relatively new to Asterisk and SIP and am trying to run a proof of concept
using Asterisk to make an outbound call through an Audiocodes gateway via
SIP using Asterisk version 1.6.1.12. The specific requirements of the
Hey!
I tried your statement but its not working but if i insert manually it works
exten = *72,10,Set(DB(CFIM/${fromext})=${toext})
at CLI
- Executing [*72@from-sip:9] Wait(SIP/7102-0004, 1) in new stack
-- Executing [*72@from-sip:10] Set(SIP/7102-0004,
DB(CFIM/7102=7207)) in
Oops!! missed your )
Sorry, It has been fixed
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Wed, 4 May 2011 20:43:16 +
Subject: Re: [asterisk-users] asterisk call forwarding
Hey!
I tried your statement but its not working but if i insert manually it
On 5/4/2011 4:04 PM, Warren Selby wrote:
On Wed, May 4, 2011 at 12:10 PM, John Hablitzel jjblitz...@gmail.com
mailto:jjblitz...@gmail.com wrote:
Relatively new to Asterisk and SIP and am trying to run a proof of
concept using Asterisk to make an outbound call through an
Audiocodes
On 5/4/11 7:10 PM, John Hablitzel wrote:
exten = xxx,n,Set(CALLERID(name)=)
I'd either leave the name alone or do te following (haven't had the need
for removing it):
exten = xxx,n,Set(CALLERID(name)=)
--
Andreas Sikkema
--
On 3/05/11 4:01 AM, Hans Witvliet wrote:
Just a thought
If Digium / the community realy want an objective way of deciding
whether can/should migrate to any other version, you realy need a
feature-matrix (pethaps starting from version 1.2.*)
And for every and each version a statement if it is:
-
Are you trunking the calls via IAX2 or something?
Are you using a jitter buffer?
Are you sure about the direction?
Do you get the same problem if you use something like sipp to create 30
LAN calls and one Internet call?
--
Cheers,
Matt Riddell
On 5/05/11 3:02 AM, vip killa wrote:
Honestly Digium's Asterisk is not a quality project. Though it has lead
the way in innovative open-source VoIP, it's a flawed and chaotic
project. Hence, I refuse to pay Digium.
So why do you use it?
--
Cheers,
Matt Riddell
I have a situation where we have an asterisk box that is extending several
Mitel PBX extensions to
some cordless SIP phones (Cisco WIP310). Everything works great, except
when the cordless
phone walks out of range of one access point and into range of another
(cisco 1100 series APs).
I've been
I have a situation where we have an asterisk box that is extending several
Mitel PBX extensions to
some cordless SIP phones (Cisco WIP310). Everything works great, except
when the cordless
phone walks out of range of one access point and into range of another
(cisco 1100 series APs).
I have
On 5/05/11 10:21 AM, Shawn L wrote:
I have a situation where we have an asterisk box that is extending
several Mitel PBX extensions to
some cordless SIP phones (Cisco WIP310). Everything works great,
except when the cordless
phone walks out of range of one access point and into range of
At 03:21 PM 5/4/2011, you wrote:
Barring that, if the cordless phone becomes un-reachable is there a
way to automatically put the active call
on hold, or park it? That's not the preferred solution, but it
would work great until I figure something else
out.
Not that it applies but I recently
ChanIsAvail + dialplan routing to call parking lot
On Wed, May 4, 2011 at 6:02 PM, Ira i...@extrasensory.com wrote:
At 03:21 PM 5/4/2011, you wrote:
Barring that, if the cordless phone becomes un-reachable is there a way to
automatically put the active call
on hold, or park it? That's not
On 5/05/11 11:40 AM, Sherwood McGowan wrote:
ChanIsAvail + dialplan routing to call parking lot
Problem is, I think he's talking about mid call - so ChanIsAvail will
have returned success - oh unless you can run it in the h exten?
--
Cheers,
Matt Riddell
On Mon, May 2, 2011 at 7:00 PM, Jim Dickenson dicken...@cfmc.com wrote:
On May 1, 2011, at 9:07 PM, Kaushal Shriyan wrote:
Hi Jim,
Thanks for the explanation, I have couple of questions here.
1) Does the xorcom box support *8 Port PRI E1 Interface*. ?
2) Also the Primary and Secondary
On 05/05/2011 03:29 AM, Lee Howard wrote:
David Backeberg wrote:
On Wed, May 4, 2011 at 12:00 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
(For my part, I'm actually surprised that nobody came up with a proper
protocol for encapsulating the stream of zeros and ones that make up
a fax
Yes - the USB connection carries the data. Keep in mind that the HA aspect
of this product just means you can connect to two asterisk servers. There is
not data replication, detection of asterisk failure, etc. (without buying more
xorcom products). Be sure to do your homework. But they do
On 05/05/2011 01:07 AM, Tzafrir Cohen wrote:
Un-top-posting,
On Wed, May 04, 2011 at 10:01:37AM -0400, vip killa wrote:
On Wed, May 4, 2011 at 9:52 AM, Danny Nicholasda...@debsinc.com wrote:
*You are “Running before you learn to walk”! You can’t make T.38 work
(that’s ok, most other folks
My 2 cents. All these problems seem to be lack of focus. Digium,
please stop doing everything to everyone. Too many versions, too many
features, too many code, too many bugs. Following the Pareto's
principle, 80% of the users use only 20% of the code. My suggestion is
to start thinking of Asterisk
I recently tried to update my Aastra 57i to version 3.2 and ran into
a problem. It won't properly register and says contact mismatch.
I added sip contact matching: 2 to aastra.cfg, but that didn't help.
When I look at the SIP trace, but I see is the Aastra sending a
REGISTER and Asterisk
2011/5/5 Richard Kenner ken...@gnat.com
I recently tried to update my Aastra 57i to version 3.2 and ran into
a problem. It won't properly register and says contact mismatch.
I added sip contact matching: 2 to aastra.cfg, but that didn't help.
When I look at the SIP trace, but I see is the
2011/5/5 Flavio Goncalves fla...@asteriskguide.com
snip
but stuffing Asterisk with
many new features on each version does not seem to be contributing to
the stability of the code or the migration to newer versions.
yes but it seems to me that code stability is improving.
Maybe next 1.10.0
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, May 04, 2011 11:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Discussion: Are we ready to leave 1.4
Is asterisk replying differently when firmware 3.2 is used ?
No, but the phone cares with 3.2 and not with 2.6.
--
_
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