please help me to solve the problem
if U can solve it for a chocolate :) it is also ok
https://issues.asterisk.org/jira/browse/ASTERISK-26073
--
Best regards
Antony
tel. +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy
On Sat, 12 Mar 2016, Ivan Demkovitch wrote:
I created very simple automated attendand (with a help of book), below is code.
But logic is simple:
Depending on time - I want:
If during business hours - give them menu and handle extensions
If after hours - give them message and take to voicemail.
Hello group,
I’m developer myself but creating dial plans is little bit different I guess :)
I created very simple automated attendand (with a help of book), below is code.
But logic is simple:
Depending on time - I want:
If during business hours - give them menu and handle extensions
If
Hi,
I am trying to setup a WebRTC connection to asterisk 1.13.0.
Using Bria a regular SIP connection works, but using sipml5 on chrome, I
got nothing.
My network setup by the way: I am working behind a comcast cable modem, the
test setup is at digital ocean, and from my laptop I also have a
On 4/30/2014 7:24 PM, Jesse Thompson wrote:
impacted. However new files introduced into /var/spool/asterisk/outgoing/
folder get ignored. No messages spring up on asterisk -rvv console, nothing
shows up in the logs, the .call files just get snubbed. We're at a loss to
Are the new files being
Are the new files being named uniquely ?
there are bugs (e.g., jira# 11291) that have to do with files having the
same name.
my solution was to add .$$ on the end of the filename to ensure it was
unique.
Yep, the files get a -MM-DD_HH:ii:ss- timestamp prefix in their names
before
We've built an alert system at our company so that if our monitoring
software notices anything very bad happening, and we don't react to a text
message after a few minutes, then it will begin to call our telephones
directly. This seems to help a lot with staff who are asleep, or who might
not be
thanks a lot i will test and i will update you as soon as i have any
problem
2013/3/22 Asghar Mohammad asghar...@gmail.com
your dialplan nothing to do with bandwidth it dial out to digium card what
ever come in.
1.
if your providers calls come in via digium card and you want send out
using
ok thank you so much i use dial(zap/r2) instead of g2 and it works without
problem
now my question i have 2 providers i use g1 for the first and g2 for the
second
if i understand i must use r1 instead of g1 for the first provider and r2
instead of g2 for the second provider in order to use
Ya u r right. Value of 1 in r1 or g1 is group you mentioned in zapata.conf
On Mar 22, 2013 8:54 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
ok thank you so much i use dial(zap/r2) instead of g2 and it works without
problem
now my question i have 2 providers i use g1 for the
Hello bharat,
ok thank you so much for your help and support now i understand :)
2013/3/22 Bharat Lalcheta bharatlalch...@gmail.com
Ya u r right. Value of 1 in r1 or g1 is group you mentioned in zapata.conf
On Mar 22, 2013 8:54 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
ok
hi,
i think we miss understood you Question?
you need round robin on tdm trunk or on 2 internet connections?
what are you asking about burden-sharing between Wimax and FH?
On Fri, Mar 22, 2013 at 4:24 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
ok thank you so much i use
yes i want to use the burden-sharing between Wimax and FH using a diguim
cards
2013/3/22 Asghar Mohammad asghar...@gmail.com
hi,
i think we miss understood you Question?
you need round robin on tdm trunk or on 2 internet connections?
what are you asking about burden-sharing between Wimax
your dialplan nothing to do with bandwidth it dial out to digium card what
ever come in.
1.
if your providers calls come in via digium card and you want send out using
sip or any other tech. then use context defined in group 1 for provider 1
and context defined in group 2 for provider 2.
2.
if
hello list,
i have installed 2 diguim cards in my server using asterisk 1.4 (i use the
old version with zapata.conf and zaptel.conf)
i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i
want to active the round-robin for span 2 and 6) in order to activate the
WIMAX and FH
What do you mean by roundrobin here
On Mar 21, 2013 8:27 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
hello list,
i have installed 2 diguim cards in my server using asterisk 1.4 (i use the
old version with zapata.conf and zaptel.conf)
i want to use the span 1 for group 1 and
i mean the burden-sharing between Wimax and FH
2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com
What do you mean by roundrobin here
On Mar 21, 2013 8:27 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
hello list,
i have installed 2 diguim cards in my server using asterisk 1.4
If u want to dial in round robin use Dial(zap/r2/2) . It dials using
channel in round robin
On Mar 21, 2013 9:37 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
i mean the burden-sharing between Wimax and FH
2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com
What do you mean by
On Thu, 21 Mar 2013, Salaheddine Elharit wrote:
i have installed 2 diguim cards in my server using asterisk 1.4 (i use
the old version with zapata.conf and zaptel.conf)
question 2: what is difference between etc\zapataa.conf and
etc\asterisk\zapata.conf
There is no /etc/zapata.conf.
The 2
File is ok there is no etc/zapata file.
On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com wrote:
On Thu, 21 Mar 2013, Salaheddine Elharit wrote:
i have installed 2 diguim cards in my server using asterisk 1.4 (i use
the old version with zapata.conf and zaptel.conf)
question
how can i use Dial(zap/r2/2)
below an exemple from my extensions.conf
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
Use r2 instead of g2 in dial
Dial(Zap/r2/${EXTEN}
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
hi,
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup()
Note r
: [asterisk-users] Need help understanding CDR
Hi,
Attached is a sample CDR.
I need some help to understand the billsec column.
PS: the time value in billsec duration is same.
With reference to the attached log, what does the 10 sec / 6 sec / 2 sec
correspond to:
(a) Time between call
...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Sunday, March 17, 2013 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Need help understanding CDR
Hi,
Attached is a sample CDR.
I need some help to understand the billsec column.
PS: the time value in billsec
-Commercial Discussion
Subject: [asterisk-users] Need help understanding CDR
Hi,
Attached is a sample CDR.
I need some help to understand the billsec column.
PS: the time value in billsec duration is same.
With reference to the attached log, what does the 10 sec / 6 sec / 2 sec
correspond
, 2013 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Need help understanding CDR
Hi,
Attached is a sample CDR.
I need some help to understand the billsec column.
PS: the time value in billsec duration is same.
With reference to the attached
- Non-Commercial Discussion
Subject: [asterisk-users] Need help understanding CDR
Hi,
Attached is a sample CDR.
I need some help to understand the billsec column.
PS: the time value in billsec duration is same.
With reference to the attached log, what does the 10 sec / 6 sec / 2
sec
-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Sunday, March 17, 2013 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Need help understanding CDR
Hi,
Attached is a sample CDR.
I need
-
Non-Commercial Discussion
Subject: [asterisk-users] Need help
understanding CDR
Hi
Hi,
Attached is a sample CDR.
I need some help to understand the billsec column.
PS: the time value in billsec duration is same.
With reference to the attached log, what does the 10 sec / 6 sec / 2
sec correspond
to:
(a) Time between call connection to asterisk and disconnection from
-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Sunday, March 17, 2013 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Need help understanding CDR
Hi,
Attached is a sample CDR.
I need some help to understand the billsec column.
PS: the time
hi,
billsec is time in seconds after call has answered, duration is total time
in seconds of call.
as your calls answered imidiatly your billsec and duration is almost same.
On Sun, Mar 17, 2013 at 5:14 PM, RSCL Mumbai rscl.mum...@gmail.com wrote:
Hi,
Attached is a sample CDR.
I need some
] On Behalf Of RSCL Mumbai
Sent: Sunday, March 17, 2013 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Need help understanding CDR
Hi,
Attached is a sample CDR.
I need some help to understand the billsec column.
PS: the time value in billsec
] On Behalf Of RSCL Mumbai
Sent: Sunday, March 17, 2013 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Need help understanding CDR
Hi,
Attached is a sample CDR.
I need some help to understand the billsec column.
PS: the time value in billsec
Hi all! In need of some serious help. We currently run Trixbox and Cent Os on a
2u server for our small business's phones system.
We are using some Polycom Soundpoint IP phones. The whole thing came crashing
down over the Holidays and as of right now that's about
all we have working right now
On Thu, Jan 17, 2013 at 3:05 PM, Joe Ruffolo j...@mrkgroup.com wrote:
Hi all! In need of some serious help. We currently run Trixbox and Cent Os
on a 2u server for our small business’s phones system.
We are using some Polycom Soundpoint IP phones. The whole thing came
crashing down over
On 01/17/2013 09:05 PM, Joe Ruffolo wrote:
Hi all! In need of some serious help. We currently run Trixbox and Cent
Os on a 2u server for our small business’s phones system.
Afaik Trixbox is no longer maintained and their forum are hardly active
anymore so it may be a bit of a challenge to get
Hi Joe
On 18/01/2013, at 9:05 AM, Joe Ruffolo j...@mrkgroup.com wrote:
Hi all! In need of some serious help. We currently run Trixbox and Cent Os on
a 2u server for our small business’s phones system.
We are using some Polycom Soundpoint IP phones. The whole thing came crashing
down over
Hi,
I'd like to replace my current VOIP provider with an Asterisk based
solution. I have some ideas I want to run by the list to see if they
are possible, and get answers to a couple questions.
Take a look at gafachi (https://www.gafachi.com/), good voice quality and
stable.
Larry
Hi,
I'd like to replace my current VOIP provider with an Asterisk based
solution. I have some ideas I want to run by the list to see if they
are possible, and get answers to a couple questions.
I want to setup two Asterisk servers that are linked to each other:
- The first server would be
On 29/11/12 6:33 pm, Dyweni - Asterisk-Users wrote:
I want to setup two Asterisk servers that are linked to each other:
- The first server would be my external (public) server and would live
in a real data center. The second server would be my internal
(private) server and would live in my
AFAIK its a propreitary card from Aculab and wont work on Asterisk unless
you buy software or support or both from them.
My advice is to dump it n get a digium card in same or lesser cost which
you need to pay aculab.
Mitul Limbani
On Nov 12, 2012 1:23 PM, RAJNI VANZA rajniva...@gmail.com wrote:
Hi All,
I need to install and configuration of Aculab prosody X PCI card with
Asterisk-1.8.9.1 on Centos-5.7 system.
I will try for that but not success. so, please suggest me way to achieve
it.
Thanks in Advance.
--
Best Regards,
Rajni Vanza
--
It's nearly there now, just need a few more votes in order for it to
trigger the next phase. Please take a moment to vote if you're
interested:
http://area51.stackexchange.com/proposals/12932/telephony/
On Mon, 9 May 2011, Simon P. Ditner wrote:
For those of that are fans of
For those of that are fans of stackoverflow.com, and stackexchange.com,
there's an effort to define a telephony stackexchange site. It's still in
the definition phase. What it needs to move forwards is more votes on
on/off topic questions, and perhaps some better questions to vote for or
Hi list
i setup successfull asterisk version 1.4 + opensips,
Opensips is the Registrar Server, Asterisk is the IVR server
the call flow
IP phone ---INVITE 1001 opensips - ASterisk INVITE
5001---opensips --- Busy|cancel|404..---asterisk---wait 10s to bye ---IP
phone (5000)
my case
Timothy,
The Warp analog modules version A.1 are not compatible with the PADS 2.x
software base.
Replacing the modules to A.3 or newer and upgrading the software to PADS
2.1 or newer should resolve these issues.
Best Regards,
Rod Boileau
Manager, Customer Care
PIKA Technologies
--
Timothy C Litwiller wrote:
This upgrade says it has a special procedure and changes the layout of
the files it uses - so I am not sure I can downgrade again. I've asked
on the Pikawarp.org forum but so far no answer. if it goes a few more
days I will have to try something - the people in
I've tried the forums at pikawarp.org and it seems no one is there anymore.
Is there someone here that can help.
last week I decided it was time to upgrade the rom on the machine to try
to get the newest freepbx.
so I followed the instructions to upgrade. I resetup the extensions and
all the
-users] need help on setup rtp directly between 2 sip
clients
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http
Hi all
my asterisk server, 2 sip client softphones are the same LAN
asterisk ip address : 192.168.1.5
sip client 1 : 192.168.1.4
sip client 2 : 192.168.1.2
asterisk starts ok with sip
setup the sip.conf
[test]
type=friend
username=test
secret=1000
host=dynamic
context=cucku
directmedia=yes
I guess to do what you want you need to dial directly between the phones.
Can't do it with xlite but you can with SJphones
Don't remember the exact syntax but guess it's something like
sip:usern...@the.phones.ip:5060
Alyed
2010/3/26 haloha haloha...@gmail.com
Hi all
my asterisk server, 2
Hi Alyed
xilte softphone work perfectly on other sip server(opensips server)
Don't remember the exact syntax but guess it's something like
sip:usern...@the.phones.ip:
5060
you mean i config the extension.conf look like exten =
1000,1,Dial(SIP/1...@ip address:5060), is it right?
the problem
If your sofphones are registering to the asterisk, then asterisk needs to be
in the middle, otherwise there's no way your 101 sofpthone user can actually
know where (by where I mean which IP) is the 102 softphone user.
UNLESS (yes, there's a big unless) you dial from 101 DIRECTLY to 102. How?
Hi Alyed
so the asterisk is in middle in all version, right? thank you for your
explanation
all devices i mean are asterisk + softphones
my goal is asterisk is on internet - WAN IP address and the softphones are
in NAT but the xlite supports the ICE function that is why i ask the media
should be
so the asterisk is in middle in all version, right? thank you for your
explanation
is the one whom everyone goes and says hey I'm 101 and live downstairs can
I play with you guys?
my goal is asterisk is on internet - WAN IP address and the softphones are
in NAT but the xlite supports the ICE
I have a server running Asterisk 1.4 with Asterisk GUI 2.0. I have (2)
extensions that do not have physical IP phones registered to them, and
I want these extensions to auto-forward to external cellphone
numbers.
Anyone have an easy methodology for doing this, via the GUI or via conf
file
Steve wrote:
Patterns and wildcards are your friend.
Maybe something like:
[example]
exten = _!,1, verbose(1,[${CONTEXT}:${EXTEN}])
exten = _!,2, answer()
exten = 1,3,goto(sales,s,1)
exten = 2,3,
I'm completely new to asterisk and while we have access to Nortel experts none
of them know asterisk and since I'm the network guy I've been lumped with this.
This is what I'm trying to accomplish
We have a CS1000 that's sip capable.
I want to be able to connect an Asterisk box to the cs1000
I am revising our DialPlan strategy for our Asterisk system (1.4.2) and
looking for some info on 'best practices' for this. Here's what I'm
trying to do:
I have an ACD menu that gives the caller the options as follows:
- Press 1 for sales
- Press 2 for support
- Press 3 for customer service
-
If you add option 6 to the menu for the first position and use the
read command for the 2 last position and use a second line that looks
something like:
exten 6,n,Dial(SIP,6${ENTERED_NUMBER},20,t)
it should work.
The {ENTERED_NUMBER} should be the variable filled with the read
command.
Have you tried...
exten = 6XX,1,Dial(SIP,${EXTEN},20,t)
--
Jarrod Lash, jar...@fed-com.com
Federated Communications, LLC.
www.fed-com.com
Office: +1-412-357-2127
Mobile: +1-412-999-0049
Fax: +1-412-545-8368
On Wed, Dec 9, 2009 at 12:53 PM, Myles Wakeham my...@techsol.org wrote:
I am
On Wed, 9 Dec 2009, Myles Wakeham wrote:
I have an ACD menu that gives the caller the options as follows:
- Press 1 for sales
- Press 2 for support
- Press 3 for customer service
- Press 8 for a 'Dial by Name' list
or enter the extension number at anytime to directly dial that extension.
-users] Need help/suggestions for DialPlan
On Wed, 9 Dec 2009, Myles Wakeham wrote:
I have an ACD menu that gives the caller the options as follows:
- Press 1 for sales
- Press 2 for support
- Press 3 for customer service
- Press 8 for a 'Dial by Name' list
or enter the extension number
-users] Need help/suggestions for DialPlan
On Wed, 9 Dec 2009, Myles Wakeham wrote:
I have an ACD menu that gives the caller the options as follows:
- Press 1 for sales
- Press 2 for support
- Press 3 for customer service
- Press 8 for a 'Dial by Name' list
or enter the extension number
-users] Need help/suggestions for DialPlan
On Wed, 9 Dec 2009, Myles Wakeham wrote:
I have an ACD menu that gives the caller the options as follows:
- Press 1 for sales
- Press 2 for support
- Press 3 for customer service
- Press 8 for a 'Dial by Name' list
or enter the extension number
Hello, I would appreciate if someone can give some help on what I want:
When someone call my box (from outside), to a certain ZAP port, it will put
him on hold, and immediately the box calls to outside SIP trunk to a
preconfigured certain number, then when the other party picks up the phone,
I have a DID but for some reason is not working in asterisk-1.6
The same sip connection in asterisk-1.4 is working OK, but it doesn't work with
asterisk-1.6
Here is my sip.conf section:
...
[actio-out]
type=friend
secret=password
user=48746612254
username=48746612254
fromuser=48746612254
Hi list,
I am new to asterisk. I need help for installing and configure Asterisk
IVR,OBD,IBD Server.
We have a PRI line,I need to know what are the system requirements and
hardware requirement for Asterisk *IVR*,*OBD*(Outbound
dialer),*IBD*(Inbound dialer).
Thanks and Regards,
Kiran Reddy
Hi list,
I am new to asterisk. I need help for installing and configure Asterisk
IVR,OBD,IBD Server.
We have a PRI line,I need to know what are the system requirements and
hardware requirement for Asterisk *IVR*,*OBD*(Outbound
dialer),*IBD*(Inbound dialer).
Thanks and Regards,
Kiran Reddy
Hi list,
I am new to asterisk. I need help for installing and configure Asterisk
IVR,OBD,IBD Server.
We have a PRI line,I need to know what are the system requirements and
hardware requirement for Asterisk *IVR*,*OBD*(Outbound
dialer),*IBD*(Inbound dialer).
Thanks and Regards,
Kiran Reddy
Hi list,
I am new to asterisk. I need help for installing and configure Asterisk
IVR,OBD,IBD Server.
We have a PRI line,I need to know what are the system requirements and
hardware requirement for Asterisk *IVR*,*OBD*(Outbound
dialer),*IBD*(Inbound dialer).
Thanks and Regards,
Kiran Reddy
On Wed, 21 Oct 2009, kiran.re...@mpowerglobal.in wrote:
I am new to asterisk. I need help for installing and configure Asterisk
IVR,OBD,IBD Server.
4 posts in 3 hours?
1) Don't repost, you just annoy people that may have helped you.
2) Ask specific questions, not I know nothing, please tell
Hi,
I have to implement asterisk to get a IVR system so that simultaneously 4-8
users can call and redirect them to other phones preferably GSM mobiles.
I will use PSTN lines. Also I need to know if these calls can be redirected
some how to GSM mobiles.
I do not have any pre-requisite in this
Hello all,
I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon, but
without success.
Is there a proper online manual that describes all the steps to follow
and debugging/monitoring information?
When I type in the CLI module show, cdr_addon_mysql.so is not listed,
although in
On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote:
Hello all,
I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon, but
without success.
Is there a proper online manual that describes all the steps to follow
and debugging/monitoring information?
When I type in the CLI
Message
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 17:17:59
On Sunday 30 August 2009 08:30:54 Cyprus VoIP
realize that there were some issues to
resolve first, as in the menuselect, I see that both
app_addon_sql_mysql
and cdr_addon_mysql have dependencies problems.
What should I do to resolve that?
Thanks.
Original Message
Subject: Re: [asterisk-users] Need help - CDR MySQL
I think that the missing component is mysqlclient, but when i yum
update mysql, it does nothing.
Anyone know how to download the RPM? I'm using CentOS 5.3.
Thanks.
Original Message
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Pascal Bruno tipas...@gmail.com
3.
Thanks.
Original Message
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Pascal Bruno tipas...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 18:28:35
You have to fix the dependency iss
: cdr_manager
CDR registered backend: cdr-custom
CDR registered backend: Adaptive ODBC
CDR registered backend: csv
Should there be a reference here to MySQL?
Anything else I should set?
Thanks.
Original Message
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: hh174 oliv
Cyprus VoIP wrote:
I think that the missing component is mysqlclient, but when i yum
update mysql, it does nothing.
You need to make sure that mysql-devel is installed and then re-compile
add-ons
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a
I already did that and now cdr_addon_mysql.so is loaded, but I still
don't get anything into the database. How can I debug it?
Original Message
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Doug Lytle supp...@drdos.info
To: Asterisk Users Mailing List - Non
Cyprus VoIP wrote:
I already did that and now cdr_addon_mysql.so is loaded, but I still
don't get anything into the database. How can I debug it?
I'm guessing that you've already set the database up correctly and that
you've entered the correct info in the cdr_mysql.conf?
Doug
--
Ben
On 31/08/09 9:28 AM, Cyprus VoIP wrote:
I already did that and now cdr_addon_mysql.so is loaded, but I still
don't get anything into the database. How can I debug it?
To start with do cdr mysql status.
If that shows as connected, turn debug on (inside
/etc/asterisk/logger.conf) and do a
hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get
thorugh: here is my sip debug outout: thx for ur help!!
asterisk-users@lists.digium.com
--- (13 headers 16 lines) ---
Sending to AA.BBB.CCC.DD : 28127 (NAT)
Using INVITE request as basis request -
On Thursday 30 April 2009 09:37:50 Oguzhan Kayhan wrote:
Ok I found an example script that said to be work..
but i have some errors.
Here is the script and then the error msgs.
exten = *666*,2,GotoIf($[${DB(CFBOOLEAN/${CALLERID(NUM)})} = 1]?3:102)
The DB returns nothing, so it evaluates $[
Hello,
I am trying to enable call forwarding feature on asterisk 1.6.0.9 with
asterisk-gui. Sure there is no menu for that on gui but, when i try to
write some example scripts to extensions.conf to make it work. I totally
failed.
I dont wanna install smthing like freepbx etc on the system so, i
Hello,
I am trying to enable call forwarding feature on asterisk 1.6.0.9 with
asterisk-gui. Sure there is no menu for that on gui but, when i try to
write some example scripts to extensions.conf to make it work. I totally
failed.
I dont wanna install smthing like freepbx etc on the system
On Thursday 30 April 2009 09:37:50 Oguzhan Kayhan wrote:
Ok I found an example script that said to be work..
but i have some errors.
Here is the script and then the error msgs.
exten = *666*,2,GotoIf($[${DB(CFBOOLEAN/${CALLERID(NUM)})} = 1]?3:102)
The DB returns nothing, so it evaluates $[ =
On Thu, Mar 26, 2009 at 10:22 PM, David fire ddf...@gmail.com wrote:
you can use the asterisk Manager or AMI.
there is a very good java project asterisk-java but there are librarys for
almost every languaje.
look for Asterisk Manager and AMI www.voip-info.org is a good place to
start
I would be grateful if someone could tell me where to find the docs to get
started on the following problem:
A program needs to be written to place a SIP call to a certain extension on
another Asterisk system, and see whether the call state ratchets up to
ringing, then drop, and take action on
you can use the asterisk Manager or AMI.
there is a very good java project asterisk-java but there are librarys for
almost every languaje.
look for Asterisk Manager and AMI www.voip-info.org is a good place to start
David
2009/3/27 eric weaver ecwea...@gmail.com
I would be grateful if someone
Please don't cross-post.
Max Alex schrieb:
I am using asterisk 1.4.19,
I have setup the dialplans to get the incoming call and that will be sent to
another context by local channel,
In another context i have setup the ring group, that portion is working
fine.
I have noticed that when i have
Hi All,
I am using asterisk 1.4.19,
I have setup the dialplans to get the incoming call and that will be sent to
another context by local channel,
In another context i have setup the ring group, that portion is working
fine.
I have noticed that when i have set one of the extension in call
D Tuncy, if you don't mind, can you show me your config files which you are
using to successfully register phones on two servers. I tried various
different things, and once it got registered on two servers, but couldn't
dialout on any. Now it is again back on only one server and I don't remember
Zeeshan,
I've put them on a server so that you should be able to access them...
http://astinfo.newportcoastsoftware.com/SIPDefault.cnf
http://astinfo.newportcoastsoftware.com/SIPmacaddress.cnf
d
2009/1/21 Zeeshan Zakaria zisha...@gmail.com
D Tuncy, if you don't mind, can you show me your
Thanks for these scripts. Good thing is that now phone has got registered on
two servers.
Now a new weird problem I am having is that it is not loading all the
entries from SIPDefault.cnf. It is loading only proxy related settings and
ignoring the rest.
Also I have 6 lines on this phone, but it
i need a program for monitorin my bandwitch of my asterisk server
--
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