Displaying 20 results from an estimated 37 matches for "wcselbi".
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wcselby
2011 Apr 12
0
No subject
[0004f2xxxxxx](poly650)
defaultuser=0004f2xxxxxx
callerid="Front Desk" <1600>
mailbox=1600
*setvar=callidnum=1234561600*
and from extensions.conf:
[outgoing]
; Outbound unrestricted domestic calls
exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN}
on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.)
*exten =>
2012 Feb 16
1
Park() ignores 'r' option which should disable music on hold in favour of ringing tone
When I receive a call, I want to automatically park it from the dialplan so that I can retrieve it later. However, I don't want callers to be aware that they are being parked, so I want to play a ringing tone to the caller. Park() is supposed to be able to do this:
Park([timeout][,return_context[,return_exten[,return_priority[,options[,parking_lot_name]]]]])
options
r: Send ringing
2011 Aug 05
1
Ring delay problem
Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and
Celeron), and last days when I call from one extension to another of the
same PBX after I dial the number the rings sound after 20 seconds.
In the CLI log, when I debug the AGI, I see always goes good until
dialparties.agi, and after that there are 20 seconds without any log, and so
the ring sound.
I've read
2011 Jun 25
1
Cisco IP Phones and Skinny in asterisk 1.8.4.2 "tooooooooooooooooo"
Hi All;
Again, the Cisco IP Phones 7942G and using Skinny:
I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file.
The phones are registering, but when we use them to place a call, we only hear tooooooooooo in the handset and we do not hear voice (even when we dial the digits, we only hear toooooooo .. but it dials and destination
2012 May 29
2
Fax Server for Asterisk
Hello,
For those customers with only analog lines, who ask for fax2email and
email2fax, whats the most reliable solution available and tested with
Asterisk?
Thanks
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2011 Apr 05
5
IAS trunk error AES encryption disabled. Install OpenSSL.
Hey Guys!
I am getting this wired error when i am calling IAX trunk. Everything works! but i want to get rid on these RED WARNING messages.. what is wrong here ? I have func_aes.so module loaded. also i remove and test but still same error.
-Satish
== Using SIP RTP CoS mark 5
-- Executing [7623 at from-sip:1] Macro("SIP/7527-0000000d", "orasebcamdial,7623") in
2011 Oct 19
1
Asterisk call transfers not working
Hello:
We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0
running. Everything seems to be ok but call transfers. This is the issue:
*A, B, C and D are in FXS ports*.
1) A calls B. B anwers.
2) B tries to transfer the call to C dialing *2 (code for attended
transfer).
3) A hears MOH. B dials number C.
4) Asterisk says the dialed number is incorrect or non existing.
We tried
2013 May 01
1
multiple provider for incoming
Matt,
At some point you need to consider how much is too much...
I run a call center with more then 125 commissioned phone sales reps and more than 60 customer service reps. We run dual servers, fiber from one provider and 6 bonded T1's from another provider. We purchase our so trunks from a wholesale company who is a major provider to resellers. Being so, their network is extremely
2011 Jan 10
0
No subject
Moh show files
This will show you if your class is set up correctly.
------=_NextPart_000_016C_01CBF83B.306A1A90
Content-Type: text/html;
charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable
<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
2010 Jun 29
2
Anyone can share their config file for Cisco phone please?
I have an *ipphone 7965G* which has to be connected to Asterisk. It has been
flashed with SIP firmware but the config file doesn't seem to work maybe I
am missing something in it.
I appreciate it if you can share your working sample config file with me.
Thanks
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2011 Apr 08
2
MOH not working
I am using Elastix. Asterisk is used for PBX application in Elastix. I want
to make customize MOH. But not able to use MOH. I make simple extension in
asterisk conf file but no success :(
Below are the details of configuration files.
Even default MOH is also not working....
*Asterisk Version 1.6.2.17.2
*
*1) Extension.conf*
[incoming]
exten => 6000,1,Answer
exten =>
2010 Aug 03
1
Asterisk 1.6 and PrivacyManager with SIP
Hi all,
My latest Asterisk system is based on Debian squeeze with Asterisk
1.6.2.6-1 and SIP only. One of my favorite features that I had working
with Asterisk 1.4 is the PrivacyManager. However, this was not
straightforward, because anonymous SIP calls arrive with
${CALLERID(num)} = "anonymous", instead of being blank. So, to get it
to work I added the first three rules to
2011 Apr 08
9
send voicemail to multiple emails
Is there a way for asterisk's voicemail to send an email (including
voicemail attachment) to multiple email addresses?
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2010 Sep 24
3
should trixbox system hang when ISP drops connection?
NEWBIE alert: i'm a linux person, not an asterisk person so i'm
certainly capable of handling any linux-flavoured solution you can
suggest. here's a note i got from a local company i know (some proper
names removed):
===== start =====
Now and again our ISP goes down and when it does give us a hicup, the
Asterisk system shuts down (not very forgiving). When it shuts down
our phone
2012 Jul 05
7
FreePBX: using context other than the default context and the generation for the configuration
Hi All;
If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal).
Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I
2012 Jun 17
1
Missing voicemail prompt beginning
Hello,
I am using the voicemail module of asterisk. When I did some test calls
from my mobile phone, sometimes the beginning of the prompt was missing,
e.g. instead of something like "number 12345 not available" I was only
hearing "345 not available". Verbose level 5 on the asterisk console didn't
give me any hint on this, it only shows that playback of the prompt started
2010 Jun 06
1
Error of FreePBX after installing from Yum Repository of Asterisk
Hi Guys,
Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When
trying to dial a number, I get this:
tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/
op_server.pl line 3367.
Use of uninitialized value in concatenation (.) or string at
/var/www/html/panel/op_server.pl line 3372.
Use of uninitialized value in pattern match (m//) at
2011 Sep 28
2
PSTN connectivity
Hi All,
I am trying to connect my asterisk box with freepbx to PSTN. I
have purchased x100p FXO card and installed in my asterisk server. My
freepbx detected the x100p FXO card and i can see the card specific details
in command line. I have configured the following things.
1. OUTBOUND caller id and Dialing rules in Freepbx.
2. INBOUND route
When i call to the PSTN number before
2011 Oct 04
3
Asterisk (Trixbox) - VirtualBox - Linux Host
someone have been installed Asterisk (Trixbox) on VirtualBox which is
installed on a Linux host (Ubuntu server 10.04 specifically).
I want to know if it is convenient or not, and the reaseons if i should on
shouldn't do it.
Thanks in advance.!
--
Esteban L. Cacavelos de Amoriza
Cel: 0981 220 429
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2012 Feb 01
2
Getting one way audio even NAT is configured
Hi all,
I'm getting one way audio when calling over the SIP trunk i.e. end device B
(remote end of SIP trunk) can hear device A (softphone registered with
Asterisk) but device A can't hear device B. Even though I configured same
NAT configurations on other servers and they are working good. The NAT
configuration is listed below;
localnet=130.0.0.0/130.0.0.0
externhost=12.131.12.13