Hello Experts,
Could anybody pl help resolve my query?
Thanks Regards,
Subbaiah Nachiappan
From: NACHIAPPAN, SUBBAIAH (SUBBAIAH)
Sent: Tuesday, October 28, 2014 6:04 PM
To: 'asterisk-users@lists.digium.com'
Subject: RE: Query on connecting 3G MSC with Asterisk PBX Server via SIP
Interface.
On 29/10/14 12:59 pm, A J Stiles wrote:
Imagine what would have happened to the human race if Ugg the Caveman decided
not to share the secret of making fire with everyone freely, but instead went
around demanding shiny beads with menaces from anyone who just wanted to keep
themselves warm .
Hello,
I am new to Asterisk forum :).
I have a requirement of terminating 3G Mobile originated calls (coming through
3G-MSC) to EPBX Phones via Asterisk PBX.
Setup:
Mobile Mobile Switching Center ( 3G)-SIP interface---Asterisk
PBX---SIP Phone.
I wanted to know if I
Hello Folks,
Forgot to mention the software Versions which I am using:
Asterisk: 1.8
Free PBX: 2.11
Asterisk NOW: 5.211.65
Thanks Regards,
Subbaiah Nachiappan
From: NACHIAPPAN, SUBBAIAH (SUBBAIAH)
Sent: Tuesday, October 28, 2014 5:52 PM
To: 'asterisk-users@lists.digium.com'
Subject: Query
Is there a way to list the names of the channel variables that are currently
defined on a given channel via AMI? I know of GetVar and SetVar, but GetVar
needs the name of the variable to get.
--
_
-- Bandwidth and Colocation
Check DumpChan
http://www.voip-info.org/wiki/view/Asterisk+cmd+DumpChan
http://wikiasterisk.com/index.php/Aplicaciones_B%C3%A1sicas#Aplicaci.C3.B3n_DumpChan
Regards,
Rafael Rincón
IP-COM, Inc
Senior Network Engineer
rrin...@ipcomnetwork.com
3100 SW 145th Ave. Suite 410
Miramar, FL 33027
+1
Hey list,
suppose I have several dates in a database-table, where these dates are
marked as 'set' or 'not set'.
If I do something like :
SELECT ID FROM my_table WHERE client=clientID AND set=yes
and this query results in several rows and thus several ID's like 2 5 7
11 13 14 17...
How can I
On Wed, 10 Jun 2009, Alex Samad wrote:
Hi
recently bought a soekris net5501 and a tdm410 to place in there.
I am having some issues attaching 12V power to the card via the molex
card - basically the box for the motherboard is too small.
I know this mighs sound odd, but do you really need
On Thu, Jun 11, 2009 at 09:02:37AM +0100, Gordon Henderson wrote:
On Wed, 10 Jun 2009, Alex Samad wrote:
Hi
recently bought a soekris net5501 and a tdm410 to place in there.
I am having some issues attaching 12V power to the card via the molex
card - basically the box for the
At 02:01 PM 6/10/2009, you wrote:
http://www.cyberguys.com/product-search/?keyword=molex
doesn't look like it, really need a 90 degree plug and I am in OZ not
usa so postage is going to kill me
I'd buy a standard one, pull the pins, cut off the wire end of the
plug, put it back in bend the
On Thu, Jun 11, 2009 at 11:14:47AM -0700, Ira wrote:
At 02:01 PM 6/10/2009, you wrote:
http://www.cyberguys.com/product-search/?keyword=molex
doesn't look like it, really need a 90 degree plug and I am in OZ not
usa so postage is going to kill me
I'd buy a standard one, pull the pins,
Hi
recently bought a soekris net5501 and a tdm410 to place in there.
I am having some issues attaching 12V power to the card via the molex
card - basically the box for the motherboard is too small.
I have read up about a PWR2400b and it seems to use 2wire pin, I am
guessing to connect to P8
On Wed, Jun 10, 2009 at 7:17 AM, Alex Samada...@samad.com.au wrote:
Hi
recently bought a soekris net5501 and a tdm410 to place in there.
I am having some issues attaching 12V power to the card via the molex
card - basically the box for the motherboard is too small.
You can probably find
On Wed, Jun 10, 2009 at 08:44:22AM -0400, David Backeberg wrote:
On Wed, Jun 10, 2009 at 7:17 AM, Alex Samada...@samad.com.au wrote:
Hi
recently bought a soekris net5501 and a tdm410 to place in there.
I am having some issues attaching 12V power to the card via the molex
card -
Alex Samad wrote:
I have read up about a PWR2400b and it seems to use 2wire pin, I am
guessing to connect to P8 just behind the molex connector on the tdm410.
can any one here confirm this, or have any info on the pwr2400b - ie how
it connects to the cards. The web site is a bit devoid of
On Wed, Jun 10, 2009 at 05:49:22PM -0500, Kevin P. Fleming wrote:
Alex Samad wrote:
I have read up about a PWR2400b and it seems to use 2wire pin, I am
guessing to connect to P8 just behind the molex connector on the tdm410.
can any one here confirm this, or have any info on the
Hi asterisk users,
I am in need of information about how to configure the
sip.conf and extension.conf for subscribers to support the dialog event
package rfc 4235. I am using asterisk 1.6.0.1 version.
The below are the configuration of sip.conf and extension.conf files
which I
Hello,
One of our client company is providing hosted contact center solutions with
Cisco IPCC. To keep the Call Recording cost at low, they are planning to use
Asterisk / Free Switch. Can anyone integrate Cisco IPCC with Asterisk for
call recording ?
Regards,
Kashif Naeem
Business Development
Hello All,
Can anybody suggest bluetooth head phone which can be used to place calls
with eyebeam or any other soft phone.
Regards,
--
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com
Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766
Email: [EMAIL PROTECTED]
MSN:
Ya i had some months ago. it works fine. what you need to know else...
JehanZaib Younis
Date: Wed, 19 Mar 2008 14:04:35 +0500From: [EMAIL PROTECTED]:
asterisk-users@lists.digium.com; [EMAIL PROTECTED]: Query about Bluetooth Head
phone
Hello All,
Can anybody suggest bluetooth head phone
Hi,
I am running asterisk PBX ( digium TE120P card configured) on one
system. It is connected to E1 card running application on the other system.
After establishing sync between two card, I am able to place call from sip
phone to E1 card running application. I want to pass the callerid, when
You can set the caller-id in many different ways but the easiest in by
setting it in the sip.conf profile for the extension.
So you can just add a line like this to your sip.conf under the extension:
callerid=Your Name 5554441212
Hope this helps..
Regards,
Todd R.
--
Prestige Messaging
Live
Hi Sanchal,
115 in your case is just DIALLED NUMBER and it will be searched by you E1
trunk.
If you want change your CALLERID, you would insert one default or would
insert one to each user.
the command is the same sendt by Todd:
callerid=Your Name 5554441212
but you can work with function
Hi ,
I am trying to dial in from two sip phones on one end, through
digium card to E1 card running application on another end.
with following configuration
/etc/asterisk/zapata.conf
group=1
context=default
euroisdn=EuroISDN
signalling= pri_net
Hi,
I am able to dial through asterisk PBX having TE120P card to E1 card
running application. Communication was established successfully
Now, I want to do the reverse way out. I am using the following
configurations
1)zaptel.conf
span=1,1,0,ccs,hdb3,crc4
defaultzone=us
On 26 Jul 2007 17:25:30 +0530, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel:
PRI
Error: We think we're the CPE, but they think they're the CPE too.
Hi,
Do the following steps are required while configuring D-channel
1) In zconfig.h file of zaptel package
uncomment #define CONFIG_ZAPATA_NET
make sethdlc-new
make install
2) modprobe wcte12xp
ztcfg
3) sethdlc hdlc0 cisco
Step 3 is
: [EMAIL PROTECTED]; [EMAIL PROTECTED];
[EMAIL PROTECTED]
Subject: [asterisk-users] Query
Hi,
I am facing problem in configuring D-channel. I did the following
configuration for TE-120P card
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
On Thu, Jul 26, 2007 at 05:25:30PM +0530, [EMAIL PROTECTED] wrote:
Hi,
I am facing problem in configuring D-channel. I did the following
configuration for TE-120P card
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
Hi,
I am facing problem in configuring D-channel. I did the following
configuration for TE-120P card
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
/etc/asterisk/zaptel.conf
group=1
Hi,
I am facing problem in configuring D-channel. I did the following
configuration for TE-120P card
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
Hi,
It looks like your configuration file zapata.conf syntax is wrong. Have
a look in the sample files how to set it up correctly, and if you are
still having troubles, paste your zapata.conf here.
Cheers,
Dimitri
[EMAIL PROTECTED] wrote:
Hi,
I have put Digium TE120P card in PCI slot.
For this you have to make entry in sip.conf.
it will be better if you use host=dynamic in both the phones in sip.conf
and what is the IP you are putting in phones which are on your PC.
Also check that your both sip phones which are on PC, are sending requestr to
asterisk server or not.
Kesh.
Hi,
I am trying to establish call through sip phone between two PC connected to
linux box on which asterisk server is running
1st PC is having IP Adress : 192.168.1.149
2nd PC is having IP Adress : 192.168.1.53
I am trying to dial from 1st PC to 2nd PC through asterisk server
The problem is
Hi,
I am trying to establish call through sip phone between two PC connected
to linux box on which asterisk server is running
1st PC is having IP Adress : 192.168.1.149
2nd PC is having IP Adress : 192.168.1.53
Now, I am tying to dial from 1st PC to 2nd PC
On 6/28/07, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
I am trying to establish call through sip phone between two
PC connected to linux box on which asterisk server is running
1st PC is having IP Adress : 192.168.1.149
2nd PC is having IP Adress : 192.168.1.53
El Thu, Jun 28 de 2007 a las 20:18 +0530, [EMAIL PROTECTED] comentaba:
Hi,
I am trying to establish call through sip phone between two PC
connected to linux box on which asterisk server is running
1st PC is having IP Adress : 192.168.1.149
2nd PC is having IP
I am not sure what exactly you wish to achieve. Just a basic SIP--to--SIP call
or ?
I am not much into the configs, but ya I can tell you that you can try using
FreePBX or Trixbox kind of setup to write ur Asterisk config file, rather then
u editing them, as it has macros, context etc...
Hi,
I have put Digium TE120P card in PCI slot. So, lspci command gives the
information in followimg format.
02:0a.0 Ethernet controller: Unknown device d161:0120 (rev
11)
Following modules are running when seen through lspci
wcte11xp
Hi,
Can any body tell me
(i) Does digium TE-120P card can be installed on Redhat linux 9i (2.4.20-8)
kernel
(ii) It is written in documentation that TE120P card be installed only
above 2.6.xx . So, which is the best suited one for it( 2.6.15 or 2.6.18 os
some other release)
(iii)
On Mon, Jun 25, 2007 at 12:10:04PM +0530, [EMAIL PROTECTED] wrote:
Hi,
Can any body tell me
(i) Does digium TE-120P card can be installed on Redhat linux 9i
(2.4.20-8) kernel
Why do you keep starting a new thread and not bother following up to
answers in existing threads?
--
Hi all,
Can anybody tell me that wether I should install DIGIUM-TE120P card on
redhat9.0 2.4.20-8 or 2.6.18 kernel. I am using kernel 2.6.18 but facing a very
serious problem of modutils and iptable.
Can anybody help me out.
Thanx and Regards
sanchal singh
Hi all,
Can anybody tell me that wether I should install DIGIUM-TE120P card on
redhat9.0 2.4.20-8 or 2.6.18. I am using kernel 2.6.18 but facing problem of
modutils and iptable.
Can anybody help me out of this.
Thanx and Regards
sanchal singh
On Fri, Jun 22, 2007 at 03:20:07PM +0530, [EMAIL PROTECTED] wrote:
Hi all,
Can anybody tell me that wether I should install DIGIUM-TE120P card
on redhat9.0 2.4.20-8 or 2.6.18 kernel. I am using kernel 2.6.18 but
facing a very serious problem of modutils and iptable.
What problems,
On 6/22/07, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi all,
Can anybody tell me that wether I should install DIGIUM-TE120P card
on redhat9.0 2.4.20-8 or 2.6.18. I am using kernel 2.6.18 but facing
problem of modutils and iptable.
Can anybody help me out of this.
Thanx and
ram wrote:
On 6/22/07, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]*
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi all,
Can anybody tell me that wether I should install
DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18. I am using
kernel 2.6.18 but
The best person to check with is Digium support. They have support matrix for
Kernel hardware on which ur card will perform.
Please check the compatibility matrix. Should work fine with
http://www.digium.com/en/supportcenter/documentation/viewdocs/TE120P
Digium support. 256-428-6000
Dear all,
We are connecting the PABX with Application server.What we are trying is that
when a nbr 1800 (this is not registered in PABX) is dialled the pabx should
route the call to Application server .The PABX should also have intelligence to
route the call by itself for its registered
Hello Lee,
Thanks a lot thats right but in i hearing tone when i click on buton but it
not take asterisk as a DTMF generate code so voice mail not identified.
thats problem . if u knw then reply me.
Regards,
gaur
On 5/18/07, Lee Jenkins [EMAIL PROTECTED] wrote:
gaurang sheladiya wrote:
Hello every body,
kindly i make one phone which is in java applet and there is no generate any
DTMF signal at client side only beep tones is hearing but not generate DTMF
at the back end side.
so plz if anyone know that DTMF generation proccess then plz reply me.
Thank you.
gaurang sheladiya wrote:
Hello every body,
kindly i make one phone which is in java applet and there is no generate
any DTMF signal at client side only beep tones is hearing but not
generate DTMF at the back end side.
so plz if anyone know that DTMF generation proccess then plz reply me.
gaurang sheladiya wrote:
Hello every body,
kindly i make one phone which is in java applet and there is no generate
any DTMF signal at client side only beep tones is hearing but not
generate DTMF at the back end side.
so plz if anyone know that DTMF generation proccess then plz reply me.
Hi,
I have a working asterisk 1.4.0 with Mysql Realtime configuration, and today
I encountered this error.
Now, I have no acces to any information in mysql realtime, so nothing work
now !
[Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime:
Everything is fine.
Its a problem in your database. something might have corrupted...be
prepared to load a backup
Gregory Duchatelet wrote:
Hi,
I have a working asterisk 1.4.0 with Mysql Realtime configuration, and
today I encountered this error.
Now, I have no acces to any information in mysql
Hi,
I have a query regarding pulse dialing at 20 pps.
An Analog Phone is directly connected to the FXS port of Asterisk PBX.
When the analog phone pulse-dials at 20 pps, the pulse digits were not
decoded correctly by Asterisk. For e.g. when the user dials a 2,
Asterisk decodes the pulse digit
Hi,
I understand that 700 is the default extension to initiate a Call Park.
Does anyone know of a way to configure Asterisk such that it has
more than one park extension for e.g. parkexten = 700,800,900
regards,
Kwang Mien
___
--Bandwidth and
Hi users;
i am new in the mailing list and asterisk user . i
have to implement METHOD 3 of the link
(http://www.voip-info.org/wiki/view/Asterisk+at+largeview_comment_id=11963)
i have question that is:
Q:when lets i have getting a NOTIFY message and my
phone changes the tone to a MWI tone now if
Hi Users;
i have to implement MWI scenario like this:
IPphone,ATAopenserAsterisk
my users are registered at openser and voicemail box
is configured at asterisk.
MWI is send by ASTERISK to OPENSER and then OPENSER to
IPPHONE OR ATA.
My query is this;
Q:let say i got a
This is what I do:
[cf]exten = _*72XXX,1,DBput(CF/${CALLERIDNUM}=${CALLERIDNUM:-10:3}${EXTEN:3})exten = _*72XXX,2,Answerexten = _*72XXX,3,Playback(call-fwd-unconditional)exten = _*72XXX,4,Playback(is-set-to)
exten = _*72XXX,5,SayDigits(${EXTEN:3})exten =
All,
Could any one help me in configuring the feature codes for Call
forward feature in asterisk..?
How to configure the feature code *XX for activation /deactivation of
call forward for SIP users ?
Would appreciate , if somebody can help me more in detail .
Thanks Regards,
-Sathish
Hi,
My testbed is as follows:
sipphone -- Asterisk PBX 1 -- Asterisk PBX 2 -- PSTN -- Analog Phone
I understand that one of the fields in the CDR (Call Detail Record) is the
Answer field which is the time when call is answered.
Is it right that :
a) the Answer field of the CDR at Asterisk PBX
Hi,
Can anybody tell me Does Asterisk has a TAPI Interface
sanchal
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Hi,
Can anybody tell me that does asterisk have TAPI interface
sanchal
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To UNSUBSCRIBE or update options visit:
[EMAIL PROTECTED] wrote:
Hi,
Can anybody tell me that does asterisk have TAPI interface
sanchal
No, if you're a windows user, there is asttapi which uses the management
interface though.
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Hi,
I am in a problem. Can anybody help me out.
I am trying to establish connection using hyperterminal through IAXsoft
modem using asterisk PBX. I have done the following settings in the
configuraion files of asterisk.
1) iax.conf file:
[iaxmodem]
type=friend
Hi all
I am a new user of asterisk and want to implement three way
calling but not getting any information about how to configure
asterisk conf file, Dial Plan etc..
whether a Digium card is essential or not?
Is Three way calling is same as MeetMe or separate feature in
asterisk?
Can You please
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Wednesday, February 01, 2006 7:14 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] query about Three way calling
Hi all
I am a new user
hi friends !
can anybody tell me something about cdr configuration.
actually i want to confirm about the minimum requiremnts.
is it possible to configure it with mysql server and myodbc anly or unixodbc
is also required?
in case unixodbc is also requied than help me to send some download links
Hello Deepak,
yes, you can use mysql. the packages are in asterisk-addons.
there is a very good wiki page on the subject here:
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
hope this helps,
yair
On Apr 6, 2005 7:33 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
hi friends !
can
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