Displaying 20 results from an estimated 8000 matches similar to: "IVR extension dialing error"
2012 Feb 06
2
Custom extension: dial a queue
Dear, I need to create a custom device extension in order to dial a
local queue.
Suppose my queue number is 8888, how can fill the Dial field from the
custom extension ???
Because if I put just 8888 or Local/8888, I don't succeed.
Thanks a lot
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
2010 Jun 16
4
Asterisk + E1 card
Dear all, I have to install an E1 card in my Asterisk 1.4.23 server
and here is my short question:
Is it necessary to install or update any Asterisk/Zaptel/Any extra
module or the default installation is good enough to just plug and run
the E1 card ????
Thanks a lot
Alejandro
2010 Mar 23
5
G.711a or G.711u ???
Dear all, I have an Asterisk SIP server in a LAN environment and I want your
opinion in order to decide the use of an audio codec:
What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip
calls ???
Thank you !!!
Alejandro
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2010 Aug 04
5
Asterisk and RAID
Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with
four HD's available, using CentOS as the OS.
What's the best RAID type recommendation ??? RAID 1 or RAID 5 ???
Regards
Alejandro
2010 Jun 22
6
Asterisk distribution for a Call Center
Dear all, I need to build a PBX based on Asterisk for a call center. I
have worked with raw Asterisk but it's hard to work for big
implementations think.
Also I have worked with Trixbox CE for a small bussines and it was
prette good, but I have not have many features like ACD. I know there
is another version called Trixbox PRO -specially Call Center edition-
that's not free but has got
2009 Feb 23
3
GSM codec is a good choice ???
Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
with GSM sound files.
The problem is I have IP phones Utopix HyperPhone 202 which support
only G.729a/u and G.723.1 high/low, but not GSM.
If I choose G.729A the "pass-throu" calls among users are OK, but
Asterisk can't transcode GSM to G.729A in voicemail calls.
My company doesn'y want to pay for a G.729
2011 Apr 11
2
Asterisk-Asterisk E1 connection
Dear, I have two Asterisk PBXs with an E1 interface/RJ-45 port in both
boxes. I need to connect both PBXs with E1/R2 and UTP cable.
What are the requirements to deploy the UTP cable ??? Straight-through
or crossover ??? What are the pinouts in both peers ???
Thanks a lot,
Alejandro
2011 May 06
1
Blacklist with *30
Dear, when I dial *30 in order to get instructions to blacklist an
extension, Idon't get the menu but I get a new dial tone.
What happen please ??? What can I do to solve this ???
Thanks a lot,
Alejandro
2010 Mar 16
1
Outbound route prefixes
Dear all, I use Trixbox as my PBX. Until a couple of days I've installed a
GSM Gateway to communicate with our three cellular phones:
15 64227777
15 64228888
15 64229999
The GSM Gateway has just one SIM.
I use the Free PBX web interface in order to set up the route and trunk
parameters:
Trunk:
*******
Name:
SIM1
Peer details:
host=10.10.1.2 (IP from GSM Gateway)
port=5060
type=peer
2010 Jun 03
1
Codec G.129 A vs A/B
Dear all, I've read that Asterisk supports only the G.729 A audio
codec. I have several Grandstream IP phones with G.729 A/B codec
implementation.
Does G.729 A/B mean both version A and version B, or A/B is a new
version different from A and B and it's not supported by Asterisk ???
Thanks a lot
Alejandro
2009 Mar 26
1
Sisky to connect Skype to Asterisk
Dear all, I've read some news about Sisky
(http://www.yeastar.com/Products/SiSkyEE.asp), a service to
interconnect Skype clients with SIP clients.
Does anybody test Sisky and can tell me about his experience ???
(Sisky runs on Windows because Skype and its API are more stable on this OS).
Regards,
Alejandro
2009 Nov 06
1
Need opinion about GSM codec for Internet
Dear all, I have implemented an Asterisk SIP server for a WAN VPN over
Internet. We have users distributed along all my country (Argentina) that
register to my Asterisk in order to talk among them.
I'll plan to have voice and voicemail with GSM codec, because we can't
afford the payment for the G.729 licenses (it's an administrative problem of
our company, not an echonomical problem).
2010 Mar 17
1
Adding an external dial code
Dear all, I have Asterisk managed by a FreePBX web console, and I want to
add an external dial code, in order to dial 9 to get external line/tone for
outgoing calls to the GSM network through my GSM gateway.
Where from Asterisk/FreePBX can I setup this feature ???
Thanks a lot.
Alejandro
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2010 Jul 25
1
Vicibox vs VicidialNow
Dear all, I need a call center asterisk's based solution and I see
there are two important solution for 120+ agents:
VicidialNow and ViciBox
Can you tell me the difference between these open source call center
solution please ???
Special thanks
Alejandro
2007 Apr 10
4
Asterisk without PSTN interface cards
People, I will install asterisk on my Debian Etch box without a PSTN
interface card. I want to use only softphones for the moment.
My question are:
1) Is it enough to install with "apt-get" the asterisk 1.2 or do I have
to get asterisk 1.4 manually ???
2) Do I have to configure a dummy PSTN interface in my case ??
And if you have a debian-asterisk howto, I really thank you.
Regards,
2010 Jun 28
1
Handling DTMF for number 4
Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for
mobile phone calls coming from a GSM Gateway.
All the components are set up in DTMFMODE = RFC2238, and so when the
caller from mobile touches the IP phone LAN extension, the call is
succesfully established. Everything is OK except for the DTMF for
number 4, because if the caller from mobile dial 1004 or 1014
extensions -which
2011 Feb 17
1
Setting two E1 cards
Dear, I always had one E1 card with one span, so I've never had any
problem in set it up through /etc/dahdi/sustem.conf and
/etc/asterisk/chan_dahdi.conf because I put span=1.
But now I have a PBX with two E1 cards with 4 span (8 span in total).
How do I have to define both card in system.conf and chan_dahdi.conf,
and how do I have to refer each span to the corresponding card ???
Thanks a
2010 Oct 19
1
E1 channels real time monitoring
Dear, I have an Asterisk PBX with two E1 cards: Digium TE122 and Sangoma
A101D. Sangoma card has SNMP support but Digium card has not, and also SNMP
does't give me ral time information.
Within CLI Asterisk I execute "dahdi show channels" but I don't get
information about channels usage.
What is the best way to have real time monitoring of E1 channels usage and
status ???
2010 Sep 23
1
Net2Phone SIP trunk problem
Dear, I have this scenario:
- PBX Asterisk 1.6.2.10 with private IP 192.168.0.10
- Behind a Cisco ASA firewall that connects to Internet
- SIP trunk to Net2Phone with these parameters (nat=no):
host=200.58.113.60
username=DOLLY
secret=123456
port=5060
type=peer
dtmfmode=rfc2833
disallow=all
allow=alaw&ulaw
nat=no
canreinvite=no
qualify=yes
-Softphones Xlite
The PBX can't register to