Had something recently on 2 separate sites where there was a lot of audio
lag on a call - and by a lot I'm talking 5 seconds or so.
Two different sites, but the setup is similar:
SIPphone A internet Asterisk IAX/Internet Asterisk - SIPphoneB
If the phone called extensions local to it's
I have some OpenVOX A1200p cards, and driver for them so far works only with
dahdi-2.0.0
Sorry, looks like i don't understand, how to correctly rebuild driver:
rpmbuild --rebuild http://dl.atrpms.net/all/dahdi-linux-2.1.0.3-59.src.rpm
skipped
Wrote:
On Wednesday, February 4, 2009, D Tucny wrote:
I use a slight variant of this...
exten =
s,n,Set(CALLERID(name)=${IF(${ISNULL(${DB(cidname/${CALLERID(num)})})}?Unknown:${DB(cidname/${CALLERID(num)})})})
exten = s,n,NoOp(Caller ID name mapped to ${CALLERID(name)})
Basically the same as
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Hi!
I am going nuts using REGEXP. I just want to verify if a variable
contains a valid +E164 phone number.
These, the the pattern is ^\+[0-9]+
First I tried:
Set(pattern=^\+[0-9]+);
if (${REGEX(${pattern} ${${var}})})
but that does not work, the backslash is removed, as seen in the log
I have outgoing call files working. I am trying to get the manager to
originate a call.
My outgoing call file that works looks like:
Channel: SIP/devcentos5x64_to_panel/mediaport
SetVar: MEETME_PLAYFILE=/home/silentm/record/pc.1725.wav
SetVar: agi_pa_recno=1725
Context: smvoice-dialout
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
Hi!
I am going nuts using REGEXP. I just want to verify if a variable
contains a valid +E164 phone number.
These, the the pattern is ^\+[0-9]+
First I tried:
Set(pattern=^\+[0-9]+);
if (${REGEX(${pattern} ${${var}})})
but that
D Tucny schrieb:
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
Hi!
I am going nuts using REGEXP. I just want to verify if a variable
contains a valid +E164 phone number.
These, the the pattern is ^\+[0-9]+
First I
I am connecting to a siemens hipath 4000 with dahdi 2.1.0.4
and asterisk 1.4.23 using a Te210P card.
the phone guy is saying that the lines are reporting always BUSY.
however on my end the status shows OK.
Anyone seen this? Is there something different about connecting PRI to
siemens hipath?
All,
Quick question that hopefully someone out there will know the answer to...
We were previously running Asterisk 1.4.(something) (I forget which one) on
Debian. Due to an office move, I am temporarily routing our calls through
an Ubuntu box that I have. It runs Asterisk 1.4.17-dfsg-2ubuntu1
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Klaus Darilion a écrit :
Take a look at http://bugs.digium.com/view.php?id=7494
Thanks for the pointer; I'm already monitoring this issue, but there
seems to be no progress on that, unfortunately.
Unfortunately it is not yet included in
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
D Tucny schrieb:
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
Hi!
I am going nuts using REGEXP. I just want to verify if a variable
contains a valid +E164 phone number.
Hello Jerry, I'm using asterisk-1.2.18 with Sangoma A104D interconnect
with Siemens HiPath 4000 in Brazil and works fine, no problem.
Please, look below my asterisk configurations for your help:
zapata.conf
[trunkgroups]
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
How is your features.conf set up? Do you have a Parking function in your
dialplan? The answer that comes to mind is that you are somehow using
parkandannounce instead of park and something is just mis-coded. In my
shop, I have hints registered, so core show hints will tell me which
lots are in
I hope someone can help me out with this issue. It has been dogging me
for months and I can't seem to get it to go away.
I have a Rhino Ceros box running Asterisk 1.4.21.2 connected via eth0
(nVidia MCP61 Ethernet) to a RedFone FoneBRIDGE2 dual-port with EC. The
FB is the latest hardware rev and
Danny,
I have parkext set to 7000, parkpos set to 7060-7069, context is set to
parkedcalls. In extensions.conf I just include = parkedcalls
When I dial 7000 from my desk phone (which used to render a parking location
and then play hold music), I get this on the CLI:
-- Executing
I think you need to use ParkAndAnnounce instead of Park to get the call back.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy G. Gault
Sent: Wednesday, February 04, 2009 11:16 AM
To: Asterisk Users Mailing List - Non-Commercial
You could try adding this to the default section of your dialplan
(extensions.conf)
; park a call in the lot
exten = 7000,1,Answer
exten = 7000,n,Park()
exten = 7000,n,Playback(vm-goodbye)
exten = 7000,n,Hangup()
Without this, * makes an implicit Park in your dialplan, with it you have
Hi,
Just so you know, some parking bugs were fixed in 1.4.23.1, so it might be
a good idea to update.
Regards,
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy G.
Gault
Sent: Wednesday, February 04, 2009 10:53
Take a look at http://bugs.digium.com/view.php?id=7494
Unfortunately it is not yet included in Asterisk, as the patch is
somehow a workaround (e.g. faking AOC-E based on last AOC-D).
Nevertheless a customer of us uses it for some years now (Astersik 1.2)
without any problems.
regards
klaus
If you just want to trigger click2dial you can use SIPTAPI. (make sure
to specify type=friend in sip.conf for this account)
klaus
Jeff LaCoursiere schrieb:
Funny how a topic will come up that you have never dealt with before, and
suddenly it comes up from multiple directions at the same
Mike,
Okay. That seems to be the answer. I was able to compile it from source
(couldn't find any .deb packages) and parking works as it should. However,
upgrading broke the ability to use any of our Zap channels (even using
--with-zaptel/usr/src/modules/zaptel when doing ./configure in
I am still trying to figure out a reasonable way to exit the chanspy
application in a dialplan.
For the most part I understand how things are working and there is one
change I would like to propose.
The way the 1.4.23.1 code seems to work is that if there are no channels
that match the
I found out more information..
the OTHER end is configured for OPS - off premise switch.
What settings does that correlate to in asterisk?
It sounds like is basically T1, b8zs, em wink...
However I changed my side to the above (switchtype is still national)
singalling is em_w
I have not heard
Jerry Geis wrote:
I have outgoing call files working. I am trying to get the manager to
originate a call.
My outgoing call file that works looks like:
Channel: SIP/devcentos5x64_to_panel/mediaport
SetVar: MEETME_PLAYFILE=/home/silentm/record/pc.1725.wav
SetVar: agi_pa_recno=1725
Context:
I am having a problem getting MOH to work with mpg123 on 1.6. I created
a bug ticket
and I am not getting any where so I am looking here for help.
Please see http://bugs.digium.com/view.php?id=14387 for details.
--
Jonn Taylor
Taylor Telephone Systems, Inc
8334 Argenta Trail
Inver Grove
I am unable to get my 9133i to register with my asterisk server. I am
including config files below, this a simple test network so there's nothing
secret in the config files. I have upgraded the phone to the latest software
version (1.4.3) I'm not sure what the problem is. I can call the phone from
On Wed, Feb 4, 2009 at 4:00 PM, Gregory Boehnlein da...@nacs.net wrote:
Hello,
Is anyone running Asterisk 1.4 w/ RFC2833 to Level3's SONUS network?
We are unable to get reliable RFC 2833 DTMF working, and have instead had to
use G711ULAW w/ INBAND DTMF to get around the issue. Looks
On Tue, Feb 3, 2009 at 5:04 PM, Vieri rentor...@yahoo.com wrote:
I did but apparently, there's nothing in the guides that lets me do this.
It's something about supporting 484 responses that Grandstream GXW4008
seems to do and Linksys SPA8000 doesn't (or at least it's not documented).
In
On Wed, Feb 4, 2009 at 7:40 PM, Jerry Geis ge...@pagestation.com wrote:
Seems like the first call to Channel is being MADE successfully.
Then it goes to do Context and Exten: I get failed...
[smvoice-dialout]
exten = smvoice_single_mediaport,1,agi(smvoice)
exten =
For a simple (but flexible) case I would consider ODBC + func_odbc.
Here is the idea (in case you aren't aware of how it goes...)
- Make a DB available (your choice as long as it is accessible via ODBC)
- Create table in it with your contacts (say columns number and
name, maybe more)
-
Any suggestions?
Jerry
Are you sure asterisk is to behave as signalling=pri_cpe or should it
be pri_net ?
--
exvito
___
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asterisk-users mailing list
To UNSUBSCRIBE or
Jim Dickenson wrote:
I am still trying to figure out a reasonable way to exit the chanspy
application in a dialplan.
For the most part I understand how things are working and there is one
change I would like to propose.
The way the 1.4.23.1 code seems to work is that if there are no
Jerry Geis wrote:
Jerry Geis wrote:
I have outgoing call files working. I am trying to get the manager to
originate a call.
My outgoing call file that works looks like:
Channel: SIP/devcentos5x64_to_panel/mediaport
SetVar: MEETME_PLAYFILE=/home/silentm/record/pc.1725.wav
SetVar:
Hi Steve,
Thanks again for the response-- the answer you gave was more or less the answer
that I was expecting.
I was logging all packets to and from the phone, and I never saw an ACK from
the phone for the OK to Asterisk on the VM calls -- not an ACK directed to a
different location, just
Hi guys,
I'm building a server that need to host 2 digium TDM24 cards.
I know any 3U server with 2 PCI-E slots would do. Since I do prefer supermicro
server, but getting one configured is pretty darn hard.
Any suggestions here?
Cheers,
Kelvin Chan | Positronics Ent.
Product
Are you locked into the 3U form factor?
We're running Asterisk on a Dell PowerEdge 1950 (1U, 2 full height PCI-E slots
[one home to an AEX-804E], 3 drive bays, redundant power).
I both the 2950 and 2970 (both are 2U, variable number of drive bays based on
the config you choose, the 2950
Saw your post...let me know what suggestions arise (I do not watch the
list that closely -- your was flagged because my monitoring software
spotted your email address).
g.
--
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102
www.netvoice.ca www.ip-centrex.ca
On Wed, Feb 04, 2009 at 01:04:47PM +0300, bee-beeep wrote:
I have some OpenVOX A1200p cards, and driver for them so far works only with
dahdi-2.0.0
Sorry, looks like i don't understand, how to correctly rebuild driver:
rpmbuild --rebuild
Hi Everybody
I've a requirement for one of my operators for an autodialler for which i plan
to deploy asterisk (I already have 3 asterisk servers on PRI running very well
! ). The scene is like : Asterisk will call a customer and play a prompt that
prompts him to press 1 if he wishes to talk
Hello,
I have an issue with Digium TDM 400 card series. When I try to make
outgoing call (PSTN call) for example, the Zap channel could not be
created and busy channel message appeared. Below is the full log :
[Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Executing [...@macro-
Hi Sriram,
the customer should be billed a premium rate ex, Rs.9 per minute..
Will be billed by you or by telecomm company?
Regards
David
- Original Message -
From: Sriram
To: asterisk-users@lists.digium.com
Sent: Thursday, February 05, 2009 1:46 PM
Subject:
On Wednesday, February 4, 2009, Ex Vito wrote:
For a simple (but flexible) case I would consider ODBC +
func_odbc. Here is the idea (in case you aren't aware of how it
goes...)
[... snip ...]
It may be a bit more work than using the Ast DB or other means, but it
has the advantage
Sriram:
whats going on here??
unless you are developing a vas, in which case, the provider for whom you
are doing this will have to help you. each provider would be doing this
differently.
regards
Kinjal Dixit
On Thu, Feb 5, 2009 at 7:20 AM, da...@iaxtalk.com wrote:
Hi Sriram,
the
D Tucny schrieb:
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
D Tucny schrieb:
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
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