Thank you Alec,
I needed some confirmation that it wasn't something I was doing. I can
live without pickupsound, and the bug is already reported - so it's all
good.
Sebastian
On 21/06/11 00:29, Alec Davis wrote:
This has been fixed only last month, see
Dear Alex,
Thanks for the answer. Will it work in the trunk failover mode. I mean if
trunk1 fails, the call will go through trunk2. Also I have another problem that
my provider needs P-Preffered-Identity Header to be added for each trunk. So
how can I add this header to each appropriate trunk
I'm thinking of implementing some easier to remember key sequences in
features.conf. Something like:
pickupexten = ##
atxfer = **
Can anybody think why this might be a bad idea, compared to the defaults
*8 and *2? I have made sure that no other feature uses single * or #, to
avoid matching
On Tue, Jun 21, 2011 at 5:47 AM, Alex Balashov
abalas...@evaristesys.com wrote:
I nominate this for most imaginative use of Asterisk-users of 2011.
It's already qualified to win in the grammar and spelling categories.
/r
--
_
Dear Dan;
I have to do something in the compilation to have chan_sccp? Because, I do not
have this channel and I have only chan_skinny.
Even in the /usr/lib/asterisk/module/, I did not find chan_sccp.
Maybe that is the reason why I do not have the sccp.conf file?
So, using the sccp channel,
Hello!
Im trying to setup the xorcom asterisk patch to be able of sending rtp
setrema to an oreka voip recording server but I get error messages.
[2011-06-20 15:43:07] VERBOSE[22529] logger.c: [2011-06-20 15:43:07]
-- Reloading module 'res_monitor.so' (Call Monitoring Resource)
thanks a lot now it's work correctly :)
2011/6/20 Warren Selby wcse...@selbytech.com
On Mon, Jun 20, 2011 at 12:17 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten =
Hi
When using an extension to my android gingerbread nexus one,
calls drop after a n minutes of call due as per the following
[Jun 21 09:34:37] == Begin MixMonitor Recording SIP/nexusone-0a39
[Jun 21 09:34:37] -- Executing [00310@default:4] Dial( ...
[Jun 21 09:34:37] --
On 06/01/2011 06:28 PM, Steve Davies wrote:
On 1 June 2011 15:10, randall rand...@songshu.org wrote:
On 06/01/2011 03:55 PM, randall wrote:
On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
Hi all,
After running fine for a few months now
On 06/01/2011 01:12 PM, Karsten Wemheuer wrote:
Hi randall,
Am Mittwoch, den 01.06.2011, 10:00 +0200 schrieb randall:
i get the following errors:
pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel
of span 2
Your telco provider
We have a web server that connects to our asterisk server via the AMI
for simple polling of data.
This was all working fine yesterday but now, possibly after an asterisk
restart yesterday, the Command type command are not yielding any
results.
Here's an example, which is from a log file of the
Any update ?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Tuesday, June 21, 2011 12:40 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users]
Hi all,
We have a project involving voice recognition, and will need a vocabulary of
10,000 words (actually names).
Can anyone recommend a product that works with Asterisk?
Thanks,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
I set verbose to 10. I will let you know if capture it.
I would like to elaborate as well.
We use this MTE to test polycoms and routers and allow clients to demo the
hardware.
It's the polycoms that are displaying the asterisk caller ID.
There is also no inbound route to the
Hello list,
I am unable to call *2*999... because my phone automatically sends the
number after I press *. So my IP-phone calls *2.
Now this is a Cisco, but that's not my question. Does anyone know what
setting I need to adjust so my phone (but actually any IP-phone) accepts
an * in the
Hi All,
Reposting this since since i was not a member of the list when i first sent
it
I'm trying to configure an asterisk serrver with a NTT T1 line, i get the
error 'no D-Channel found', i searched the net and seems that NTT Japan uses
a special switch type called NTT (
Ok, for the variables, I can retrieve some of them like the caller number
and so on (I would assume that all the variables that last for duration of
call are there), but I still think that I sould not use the h extension to
continue after ReceiveFAX use, it's like not a lot of people use FFA,
Hello
I have created the menu below, with this menu when I call 520460XXX I can
hear the welcome message [home] context and when I press the # I can go to
the [menu] context and hear the menu message
When I press 1 in order to go to the [call] I can hear the call message
Now I have 2
If memory serves isn't that support contract include broken phones / parts
too?
I thought I read that if my phone Is broken - it is covered
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Monday, June 20, 2011
From: khalid touati khalidtou...@gmail.com
Sent: Tuesday, June 21, 2011 9:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Problem with ReceiveFAX app from FFA
Ok, for the
check this out:
http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/258425
From: kche...@xplorium.com
To: asterisk-users@lists.digium.com
Date: Tue, 21 Jun 2011 13:25:39 +0300
Subject: Re: [asterisk-users] Asterisk call limitation
Any update ?
-Original Message-
On 06/21/2011 09:12 PM, khalid touati wrote:
Ok, for the variables, I can retrieve some of them like the caller
number and so on (I would assume that all the variables that last for
duration of call are there), but I still think that I sould not use
the h extension to continue after ReceiveFAX
@ Bryant: thanks so much for the interesting figure of use.
Why do so may people think their problems are unique. Many people use FFA
and spandsp. They all come across this. The issue is widely known, well
understood, and not at all strange once you think about it.
Steve
@ Steve: don't
I have discovered that if I enable pickupsound = beep in
features.conf,
if I try to do a pickup with *8, the calling channel keeps on ringing,
while the phone where I pick-up from shows that the call has been
answered (I don't know where though). Also, it seems to completely
bugger up my
You do not need sccp.conf if you are not using chan_sccp.
It has different features(bugs) than chan_skinny, but yes
it would also reset the phones (if it supports reload, and
I have no idea if it does).
Also if the phone is in a call it will not reset until after
the user hangs up. Reloading
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 06/21/2011 08:37 AM, Jonas Kellens wrote:
At the moment, I don't really know what I'm looking for. So if anyone
knows how to do it in a Cisco, Grandstream, Yealink or Snom IP-phone I
can find out myself what settings to look for in other
On Tue, Jun 21, 2011 at 4:12 AM, randulo rand...@randulo.com wrote:
On Tue, Jun 21, 2011 at 5:47 AM, Alex Balashov
abalas...@evaristesys.com wrote:
I nominate this for most imaginative use of Asterisk-users of 2011.
It's already qualified to win in the grammar and spelling categories.
/r
we've been getting complaints that DTMF is not working, i checked full log
for a call that they claimed DTMF didnt work, I noticed this:
DTMF begin '7' received
DTMF begin ignored
DTMF end '7' received
DTMF end passthrough '7'
why is the DTMF begin ignored called?
--
Hi I wanted to help out with dial plan, but it's not obvious what you want
to achieve, also I do recommend to read the chapter that talks about
contexts and dialplan from future of asterisk book. but if you're in rush
just try to make clear how you want your system to behave and i'll be glad
to
I'm not Asterisk expert but I had some DTMF problems in the recent past...
This message is called on main/channel.c everytime a DTMF is received, and,
afaik, this is not an error, it Asterisk ignores the first milisenconds of the
DTMF to distinguish between a real DTMF and any sound in the same
On Mon, Jun 20, 2011 at 2:48 PM, Warren Selby wcse...@selbytech.com wrote:
On Mon, Jun 20, 2011 at 3:52 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
Two requests, not from me but the community.
1. Don't top post
*cough*
2. When you find your solution, reply to this thread
Hahahah Baltimore and SE DC. How about Philly too J
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Tuesday, June 21, 2011 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Personally, I have been shot at on top the Iraqi Government building
in the IZ from the Red Zone. I was setting up and troubleshooting the
Motorola Canopy WiFi system. Just a few 7.62x39 rounds, nothing I
would call heavy fire.
It was because you were setting up the canopy stuff, not
Is anybody using Mitel phones? It appears that when you page a Mitel phone
using asterisk's MeetMe, the paged phone will hang up the call its on to
take the page. Thanks in advance.
--
_
-- Bandwidth and Colocation Provided by
On Tue, Jun 21, 2011 at 1:07 PM, Steve Totaro
stot...@asteriskhelpdesk.comwrote:
LOL at the haters.
Ahh, how often I forget that the subtleties of sarcasm are usually lost in
email...
snip
It would also be a whole lot easier for someone to physically feed me so my
hands could be free to
I have a 5224 and 5220's, I will try it tonight when I get home.
From: vip killa
Sent: Tue 6/21/2011 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call paging interrupts call when using Mitel 5224
Is anybody using Mitel phones? It appears that
I am getting events using asterisk 1.4.41. However, when I place a call
on hold I do not get that event.
Some of the events I am getting I show below. I wish to monitor when
channels are placed on hold
and taken off hold.
How do I that?
Thanks,
jerry
-
I am getting events using asterisk 1.4.41. However, when I place a
call
on hold I do not get that event.
Some of the events I am getting I show below. I wish to monitor when
channels are placed on hold
and taken off hold.
Set callevents=yes in sip.conf to enable Hold/Unhold AMI call events
Am 21.06.2011 18:28, schrieb Mark Deneen:
On Tue, Jun 21, 2011 at 4:12 AM, randulorand...@randulo.com wrote:
On Tue, Jun 21, 2011 at 5:47 AM, Alex Balashov
abalas...@evaristesys.com wrote:
I nominate this for most imaginative use of Asterisk-users of 2011.
It's already qualified to win in
--
I know we've come a long way,
We're changing day to day,
But tell me, where do the children play?
Yusuf Islam, Nov. 1970
AKA CAT STEVENS.
--
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Dear Warren;
It look like u have a good experience in 791x series and in selecting SIP
formware, so I am sure you might be able to help in the following:
As u know, there are SIP firmware for Cisco phones to be used with Call Manager
and other firmware to be used with Generic SIP Server (other
I know Asterisk 1.8 can send out texts via SMS()
Can I send Asterisk a text via a DID and it do something?
So far I am reading that it cannot but I do not know if there have been
updates.
Thanks,
-E
--
_
-- Bandwidth
Hi
I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i can't
show the callerid name in the way Asterisk == Siemens. I realized that
Asterisk send calleridname in format namePresentationAllowedSimple to
Siemens e Siemens send calleridname in format
namePresentationAllowedExtended.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Tuesday, June 21, 2011 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call paging interrupts call when using Mitel
5224
Is
On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Dear Warren;
Please, keep all discussions to the list. There's no need to email me
personally about this.
snip
cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP
Phone load) and
wow I think someone needs to just spend some time reading and playing. Getting
these phones working is not rocket science and there are similarities with how
to do firmware / config pushes.
Not to sound mean but RTFM
Sent from my iPhone
On Jun 21, 2011, at 7:45 PM, Warren Selby
I dropped my DC customers for much safer Bethesda customers :)
On Tue, Jun 21, 2011 at 2:11 PM, Robert Huddleston rhuddles...@gmail.comwrote:
Hahahah Baltimore and SE DC… How about Philly too J
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
I am using asterisk 1.4.41 and polycom phones.
When I dial long distance I hear a pause on the last 2 digits.
This is the dialout context that matches.
[ Context 'smvoice-dialout' created by 'pbx_config' ]
'_71XX' = 1. Set(CALLERID(number)=3175551212)
[pbx_config]
Hi,
I have small asterisk pbx. I was made a dialplan like whenever out side
world is dialing to asterisk pbx DID's it was goes to dial his extension
number. if this extension is not pick the call after 30 sec it will dial
his(user) mobile number. i need to create dialplan only office timings(9:00
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