Hi Yehavi,
As Alex said, it depends of what exactly you want to implement.
You just have to evaluate your target service and to properly understand
what each piece of software is appropriate for and what it has to offer.
First of all, you have 2 complementary classes of software : you have
Hi,
I'm using opensips as the registrar server for my users.
I am redirecting calls going out to pstn to my asterisk server.
call flow is basically:
ua -- opensips server -- * server -- sip gateway provider
if (uri=~sip:00[0-...@sip\.myserver\.com) {
xlog(L_INFO, Call to PSTN\n);
Hi Moy,
thanks a lot for your fix, but I'm afraid it doesn't work. I looked your patch
over and I realize the code never passes by neither of the two lines you added
with returnstatus = AGI_RESULT_HANGUP. Even, it seems the execution doesn't
pass by res_agi.c at all, or at least, it doesn't
Hi,
I have a requirement where an IVR application on asterisk has to play a audio
file in g729 and when a digit is pressed, the call should switch to another
codec (say ulaw). So, What can I do in the extensions.conf to trigger a
re-negotiation of codec?
I used
exten =
Hi,
I have a requirement where an IVR application on asterisk has
to play a audio file in g729 and when a digit is pressed, the call
should switch to another codec (say ulaw). So, What can I do in the
extensions.conf to trigger a re-negotiation of codec?
I used
exten =
Hello,
I am using TE122 between an ericsson MD110 and asterisk server.
I set on ericsson side as master PRI NET and Asterisk is PRI CPE
Even i connect jumper on external clock side(on TE122).. or even my
configuration is as
span = 1,1,0,ccs,hdb3
i still see my TE122 as Internally Clocked on
Hi Steve,
Steve Underwood wrote:
In chan_dahdi.c there is now code that extends the buffering inside
dadhi when a FAX is detected, and puts the buffering back to normal at
the end. This isn't really a cure - its more of a bandaid. However, I
expect it has the desired effect if they have
Hi Lee,
Lee Howard wrote:
Hi Steve,
Steve Underwood wrote:
In chan_dahdi.c there is now code that extends the buffering inside
dadhi when a FAX is detected, and puts the buffering back to normal
at the end. This isn't really a cure - its more of a bandaid.
However, I expect it has the
You could use this on-call script to go until you got an acceptance
exten = s,501,Set(ONECELL=${DB(Cell/One)})
exten = s,n,Set(TWOCELL=${DB(Cell/Two)})
exten = s,n,Set(THREECELL=${DB(Cell/Three)})
exten = s,n,Macro(calleng,${TWOCELL},1)
exten = s,n,Macro(calleng,${ONECELL},2)
exten =
Hi! I have a question about agents in asterisk.
In first place, agent login to asterisk (from the telephone)
The question is:
Can an agent take a break (using a function *(some number)) from the phone?
Thanks to all
Regards
--
Ing Francisco Roqué
3Tech SRL
Plaza Paso Nº92, EP B
Buenos
JD wrote:
I've got a challenge (or clarification request if I am mistaken) for the
group.
I have a non-profit customer on asterisk 1.4 that has multiple
volunteers that work from home. The volunteers are willing to take calls
to help out the organization.
So, a formal queue is out. They
Steve Underwood wrote:
Hi Lee,
Lee Howard wrote:
Hi Steve,
Steve Underwood wrote:
In chan_dahdi.c there is now code that extends the buffering inside
dadhi when a FAX is detected, and puts the buffering back to normal
at the end. This isn't really a cure - its more of a bandaid.
Thanks James. I read it. But the cmd PauseQueueMember must be executed
from an extension.
As soon as the agent logged in, asterisk does not recognize the dtmf.
There can be an alternative solution?
Regards
Francisco
james.coll...@xtratelecom.es wrote:
You can use the PauseQueueMember
BJ Weschke wrote:
JD wrote:
I've got a challenge (or clarification request if I am mistaken) for the
group.
I have a non-profit customer on asterisk 1.4 that has multiple
volunteers that work from home. The volunteers are willing to take calls
to help out the organization.
So, a
On Apr 13, 2009, at 11:23 AM, JD wrote:
BJ Weschke wrote:
JD wrote:
I've got a challenge (or clarification request if I am mistaken)
for the
group.
I have a non-profit customer on asterisk 1.4 that has multiple
volunteers that work from home. The volunteers are willing to take
You could make the agent busy with this kind of logic
exten = 2000,1,Answer
exten = 2000,2,SetMusicOnHold(default)
exten = 2000,n,WaitMusicOnHold(300)
exten = 2000,n,Background(vm-goodbye)
exten = 2000,n,Hangup
This would let the agent play MOH back to his/her self for 5 minutes and tie
up the
On Sunday 12 April 2009 02:34:10 am Julian Lyndon-Smith wrote:
Eric Chamberlain wrote:
On Apr 11, 2009, at 12:53 AM, Julian Lyndon-Smith wrote:
Eric Chamberlain wrote:
[snip]
Thank you, that bug does have useful information.
We are working on moving from res_config_odbc to
On Apr 13, 2009, at 8:52 AM, Tilghman Lesher wrote:
On Sunday 12 April 2009 02:34:10 am Julian Lyndon-Smith wrote:
Eric Chamberlain wrote:
On Apr 11, 2009, at 12:53 AM, Julian Lyndon-Smith wrote:
Eric Chamberlain wrote:
[snip]
Thank you, that bug does have useful information.
We are
Hi there,
this is the first time that I'm building an Asterisk-server.
I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.
Thought it would go easier...
I have 2 Grandstream
jonas kellens wrote:
Hi there,
this is the first time that I'm building an Asterisk-server.
I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first
internal communication with SIP.
Thought it would go
bindaddr = 0.0.0.0
I would set this to the ethernet interface IP address, I believe this may be
your issue.
Registration is only for receiving calls, if you are not seeing information on
the dial, then the phone is not talking to the server. I would make sure of
the settings in the
What do you see when you run asterisk –r and dial 210 or 211 from one of the
phones
James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.5,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including
In your sip.conf or sip_nat.conf (for elastix) set the variables:
externhost=
externip=
domain=
externrefresh=
localnet=
Regards
Francisco
Anthony Plack wrote:
bindaddr = 0.0.0.0
I would set this to the ethernet interface IP address, I believe this may be
your issue.
Mike,
thank you for your reply.
However I do not have the option of a DHCP-server. On the network where
Asterisk needs to be implemented all is configured statically, so also
the IP-phones need to be statically assigned an IP-address. Surely this
can not be thé problem...
Greetingz,
Jonas.
On
Do you have include=intern in the default context? If no, * will come back
with can't find peer 210 (or 211).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Monday, April 13, 2009 11:19 AM
To:
On Mon, 13 Apr 2009, Anthony Plack wrote:
bindaddr = 0.0.0.0
I would set this to the ethernet interface IP address, I believe this
may be your issue.
Binding to 0.0.0.0 means listen to all IP addresses on the box. It is
not the issue.
Thanks in advance,
Tony Plack,
this is the result form Asterisk CLI :
[r...@asterisk asterisk]# /usr/sbin/asterisk -vr
Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for
Alright again, what do you see on the CLI when you make a call to 210/211?
James Shigley
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Monday, April 13, 2009 12:07 PM
To: Asterisk Users Mailing List -
James,
when I run Asterisk -vr and I enter 210 on one phone to call the
other, nothing is displayed on the CommandLine...
I know this is not right, just don't know what is wrong. I really need
someone to guide me a bit...
[r...@asterisk asterisk]# /usr/sbin/asterisk -vr
Asterisk 1.4.24,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: April-13-09 1:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk-beginner : cannot make
On Monday 13 April 2009 11:14:14 am Eric Chamberlain wrote:
On Apr 13, 2009, at 8:52 AM, Tilghman Lesher wrote:
On Sunday 12 April 2009 02:34:10 am Julian Lyndon-Smith wrote:
Eric Chamberlain wrote:
On Apr 11, 2009, at 12:53 AM, Julian Lyndon-Smith wrote:
Eric Chamberlain wrote:
Danny,
this is from the Asterisk CLI :
asterisk*CLI dialplan reload
Dialplan reloaded.
== Parsing '/etc/asterisk/extensions.conf': Found
-- Registered extension context 'default'
-- Including context 'intern' in context 'default'
-- Registered extension context 'intern'
--
I'm running some mysql queries on the standard sql logging of calls, and am
interested if anyone has any they'd like to share to get good statistics. I'm
interested in # of calls per day, based on DST. Number of Calls per day based
on SRC, avg duration of calls, etc..
Thanks.
Jeremy Mann
I pick up the phone, and dial 211 on the BT201. This is the Asterisk
CLI :
Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895)
Verbosity is at least 5
asterisk*CLI
Nothing is displayed... it stays that way...
Jonas.
On Mon, 2009-04-13 at 11:59 -0500, James A. Shigley
These are the settings on my BT201 (GXP1200 is the same interface) :
Account Name:(e.g., MyCompany)
SIP Server:(e.g., sip.mycompany.com, or IP address)
Outbound Proxy:(e.g., proxy.myprovider.com, or IP address)
SIP User ID:(the user part of an SIP address)
-- I put here the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
jonas kellens wrote:
I pick up the phone, and dial 211 on the BT201. This is the Asterisk CLI :
/Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895)/
/Verbosity is at least 5/
/asterisk*CLI /
Nothing is displayed... it
Danny Nicholas wrote:
Do you have include=intern in the default context? If no, * will come
back with can't find peer 210 (or 211).
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *jonas
kellens
*Sent:* Monday, April
Barry,
there is a 'send' button but pushing it before or after dialing '211'
does not really change anything...
I get no dial tone, no ring tone on the other phone and no output on the
Asterisk CLI...
I thought this would go easier... Don't know what is going on here.
I followed the book
jonas kellens wrote:
Hi there,
I notice (on the Asterisk CLI) that my SIP-phones do not register.
They have a fixed IP and there account information is
If your phones don't register, then your not going to be able to make a
call.
The Grandstream phones have a web interface (At least if
Hey there again !
I've changed some things now :
1) IP-phones get there IP from a DHCP
2) sip-accounts simplified :
[r...@asterisk asterisk]# cat sip.conf
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
[210]
type=friend
context=intern
host=dynamic
Hi
On Mon, Apr 13, 2009 at 06:18:58PM +0200, jonas kellens wrote:
I pick up the phone of the BT201 and dial 211... nothing happens.
I pick up the phone of the GXP1200 and dial 210... nothing happens.
I would love to have your feedback on this. Where could this problem be
situated ?
Your
On 13 Apr 2009, at 20:52, jonas kellens wrote:
Hey there again !
If you are new to all this wouldn't going with some pre-made dialplan
be useful? Go for something like FreePBX
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
On Mon, 13 Apr 2009, jonas kellens wrote:
1) IP-phones get there IP from a DHCP
The source of the address is not the issue.
I still see no register-message on the CLI. This really should happen
now, as they are defined host=dynamic !
I suspect you have not [correctly] configured the phones
jonas kellens escribió:
Hey there again !
Hey, just my two cents:
I've changed some things now :
1) IP-phones get there IP from a DHCP
2) sip-accounts simplified :
/[r...@asterisk asterisk]# cat sip.conf/
/[general]/
/context=default/
/port=5060/
/bindaddr=0.0.0.0/
/srvlookup=yes/
Hi Tzafrir,
yet with the first test, things get wrong :
asterisk*CLI logger show channels
Channel Type StatusConfiguration
--- ---
/var/log/asterisk/messages File Enabled- Warning
1) IP-phones get there IP from a DHCP
The source of the address is not the issue.
I still see no register-message on the CLI. This really should happen
now, as they are defined host=dynamic !
I suspect you have not [correctly] configured the phones to register to the Asterisk server.
You
On Mon, Apr 13, 2009 at 10:39:49PM +0200, jonas kellens wrote:
Hi Tzafrir,
yet with the first test, things get wrong :
asterisk*CLI logger show channels
Channel Type StatusConfiguration
--- --
[r...@asterisk asterisk]# netstat -a -n -p | grep 5060
udp0 0 0.0.0.0:50600.0.0.0:*
3047/asterisk
[r...@asterisk asterisk]# /usr/sbin/tcpdump port 5060
tcpdump: verbose output suppressed, use -v or -vv for full protocol
decode
listening on eth1, link-type EN10MB
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
short and some IVR's are unhappy or don't detect it. I did poke
around, but it looks like when RFC2833 is used, it actually generates
rtp packets of some sort, so
On Mon, 13 Apr 2009, jonas kellens wrote:
[r...@asterisk asterisk]# /usr/sbin/tcpdump port 5060
tcpdump: verbose output suppressed, use -v or -vv for full protocol
decode
listening on eth1, link-type EN10MB (Ethernet), capture size 96 bytes
23:04:59.522498 IP 192.168.4.114.sip
John covici wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
short and some IVR's are unhappy or don't detect it. I did poke
around, but it looks like when RFC2833 is used, it actually generates
rtp
Hi everybody!
I'm triying to install a Sangoma A200-R FXO card on a Debian Linux 5
(Lenny), 2.6.26 kernel.
To install wanpipe driver I type:
WANPIPE_FOLDER# ./Setup install
Everything seems to be ok. There are no broken dependencies and the
hardware is well detected, even zaptel is
Y
On Mon, Apr 13, 2009 at 6:05 AM, Sai P. Varanasi
saiprabhak...@yahoo.com wrote:
Hi,
I have a requirement where an IVR application on asterisk has to play a
audio file in g729 and when a digit is pressed, the call should switch to
another codec (say ulaw). So, What can I do in the
sorry,
You can set SIP_CODEC before the call is answered ... most likely as
one of the first priorities.
It causes the 200 OK to INVITE contain the codec you specify as the first one.
I'm not aware of reinviting while in a call other than to switchover
to T.38 ... it can be coded in but I'm not
Hi All,
-My asterisk will not save voicemail greetings when you call in and
record them.
-It also will not save voicemail messages after emailing them,even
though delete=no.
-Folder permissions are fine, no errors in asterisk cli.
-If i go into /var/spool/asterisk/voicemail/default/200 and
Which Wanpipe version did you download?
2009/4/13 Giovanni Andrés Nopal Pascual giova...@voip.unam.mx:
Hi everybody!
I'm triying to install a Sangoma A200-R FXO card on a Debian Linux 5
(Lenny), 2.6.26 kernel.
To install wanpipe driver I type:
WANPIPE_FOLDER# ./Setup install
On Monday 13 April 2009 05:54:38 pm Dr. Kenneth Noisewater wrote:
Hi All,
-My asterisk will not save voicemail greetings when you call in and
record them.
-It also will not save voicemail messages after emailing them,even
though delete=no.
-Folder permissions are fine, no errors in asterisk
Hi All,
I'm in the process of writing an install script and I would like to change some
settings for the install process but I don't want the user to go into
menuselect and make the changes manually.
Is there a way to make the changes to menuselect from the CLI?
As an example, selecting the
I very probably did build them with ODBC or MySQL support. IMAP I
don't think so, but where would I look for configs that tell asterisk
to use such support? I'm almost positive I compiled it to support
database, but I definitely never configured it for use. Or is this
something it does
Is a drive image out of the question?
PaulH
David Klaverstyn wrote:
Hi All,
I’m in the process of writing an install script and I would like to
change some settings for the install process but I don’t want the user
to go into menuselect and make the changes manually.
Is there a way to
I have an application that needs to vary the menu choices available based
upon the availability of an external resource at a given time. What I have
in mind is a system that can uplink a user to one of many different
satellites. Due to the nature of orbital mechanics a satellite may be out
of
Duh, read the subject.
I suggest to try 3.3.16 beta, given that is probably a kernel version issue.
On Mon, Apr 13, 2009 at 7:15 PM, Moises Silva moises.si...@gmail.com wrote:
Which Wanpipe version did you download?
2009/4/13 Giovanni Andrés Nopal Pascual giova...@voip.unam.mx:
Hi
On Mon, 13 Apr 2009, Eric Fort wrote:
I have an application that needs to vary the menu choices available
based upon the availability of an external resource at a given time.
What I have in mind is a system that can uplink a user to one of many
different satellites. Due to the nature of
hi
you can use AGI or a database internal or external
then if you know all the satellites and are a few you can
if(${SAT1}=1)
playback(SAT1)
if(${SAT2}=1)
playback(SAT2)
.
.
.
or you can use an agi
David
2009/4/14 Eric Fort eric.f...@gmail.com
I have an application that needs to vary the
2009/4/14 jonas kellens jonas.kell...@telenet.be
[r...@asterisk asterisk]# netstat -a -n -p | grep 5060
udp0 0 0.0.0.0:50600.0.0.0:*
3047/asterisk
[r...@asterisk asterisk]# /usr/sbin/tcpdump port 5060
tcpdump: verbose output suppressed, use -v or -vv for full
Hi All,
Just wanted to post a follow up in case anyone else has the same issue
in the future.
I recompiled Asterisk and in the makemenu system there is a Voicemail
Build Options, in there there is []ODBC Storage and []IMAP Storage.
I had ODBC Storage checked on my last compile, I unchecked
2009/4/12 Martin asteriskl...@callthem.info:
1) your asterisk box talks to OpenSIPS
yes , he talk with opensips
2) in that case OpenSIPS should traverse NAT
no , my users are of opensips , asterisk is set mode comedia
3) you should not do nat=yes for that device since Asterisk talks to
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