When we have calls that originate click-to-daial apps that use the manager
interface they always originate from asterisk is there any way to change
the from name?
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asterisk-users mailing list
To
Hi,
My configuration is SipPhone--*1---*2.
My asterisk version is 1.4beta3.
I installed pwlib,openh323,chan_h323.
When i call from
SipPhone--(SIP)--asterisk1---(H323)--asterisk2,
there is no audio.
Using 'rtp debug', I can see that rtp packets are
being received.
Rtp packets are being exchanged.
Yes, I have done it. I am able to connect using odbc. Now able to write to
ms sql and also retrieve in db. Now my next steps is I need to write an app
which takes a phone call, asks for the user to input a number and then
queries a MS SQL db and reads the results a row at a time back to the
Other than rebooting the server or restarting Asterisk from cron does anyone
know how to kill a meetme room at midnight. Or perhaps other creative ways
people deal with callers who don't hang up.
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Nothing better, I tried some solutions, but nothing is changed.
After some minutes, or after an asterisk reload , it loses all my snom
subscriptions...
I have an asterisk 1.2.1 on my WRT54GL , all is ok, and I use SNOM 320 as
sip phones.
When they boot up the subscriptions are ok, and asterisk
On Wed, Nov 22, 2006 at 11:29:01PM -0500, Michelle Dupuis wrote:
48VDC is a long time telco standard - and has become the Power over Ethernet
standard.
Keep in mind that 'electricity' isn't the measure - it's power. Power is
not synonymous with voltage.
More to the point, there is a
On 19:18, Thu 23 Nov 06, Eric Bishop wrote:
Other than rebooting the server or restarting Asterisk from cron does anyone
know how to kill a meetme room at midnight. Or perhaps other creative ways
people deal with callers who don't hang up.
You can use soft hangup chan
--
Michiel van Baak
Julian J. M. wrote:
FYI, the interval at which the device is checked is 60seconds when OK,
and 10s when not OK.
It can be changed in channels/chan_sip.c. Look for this lines:
#define DEFAULT_FREQ_OK 60 * 1000 /* How often to check
for the host to be up */
#define
This would require a change in chan-capi. To get the extended tone
detection
indications, additional request/parameter via CAPI must be issued.
First, thanks for your reply.
Do you have the CxDtmf.pdf document, from Eicon ?
If I understand good, you have to enable DTMF facilities 248, 249 and
On Thu, 23 Nov 2006, Gregory Duchatelet wrote:
This would require a change in chan-capi. To get the extended tone
detection
indications, additional request/parameter via CAPI must be issued.
First, thanks for your reply.
Do you have the CxDtmf.pdf document, from Eicon ?
Yes.
If I
Hi,
Thank you for your response. As you said, I have tested. But, its not going and
simply hangup. What I have to do? Please tell me. Thank you.
Regards,
Chandra.
zero massive [EMAIL PROTECTED] wrote: Here you go:
[Custom-CLID]
exten = s,1,Answer
exten = s,2,Authenticate(12345)
exten =
Hi all,
All of a sudden all my IAX DIDs have gone down. I couldn't find any reason
other than that the ISP is blocking port 4569. DIDs register fine from my
home server, but not from office server, which is not behind any NAT. SIP
registers fine. I am trying to change IAX port but it apparantly
On Thu, Nov 23, 2006 at 12:40:03AM -0800, Brad Templeton wrote:
[snip]
The USA uses 120v for house current. That's enough to hurt you and can
kill you if you touch it wrong, though I've touched it a few times.
A lot of the world uses 220. This causes enough of a spark that they
require all
iax2 debug is giving following messages repeatedly.
Tx-Frame Retry[003] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
Timestamp: 1ms SCall: 00010 DCall: 0 [xxx.xxx.157.230:4569]
USERNAME: XXX9072835
REFRESH : 60
Tx-Frame Retry[002] -- OSeqno: 002
try this, pls give some feedback
###
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
fxsks=1-4
bchan=5-19,21-35
dchan=20
loadzone = us
defaultzone=us
###
On 11/22/06, Lincoln Zuljewic Silva [EMAIL PROTECTED] wrote:
This is the scenarios:
1 -
###
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
Hello,
Where should I find any updated AGI informations?
I am using wiki now but there are many outdated info (old pages) and
might some detail changed since it written.
For example I need to playback a sound file and there is a STREAM FILE
command. The wiki page notice a bug but I don't know
On Thu, Nov 23, 2006 at 03:05:38AM -0800, Crazy Boy wrote:
Hi,
Thank you for your response. As you said, I have tested.
But, its not going and simply hangup. What I have to do? Please tell me.
Please provide the dialplan you use as well as a trace of the CLI from
when you get a call. Set
On Thu, Nov 23, 2006 at 11:49:50AM +, Marco Mouta wrote:
try this, pls give some feedback
This one is evidently false:
###
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
It claims that the T1 span is the first one. However:
fxsks=1-4
The analog span is the first one. Which is generally
I have all three running on the same box. I say OT because it appears
asterisk is doing it's job just fine. It must be an IAXmodem or
faxgetty (hylafax) problem
When faxes work, they look great. I have ten IAXmodems setup with
different ports and they register fine. I have ten faxgettys
[EMAIL PROTECTED] wrote:
We are looking for a site running Queumetrics in Sydney, Australia.
We have been contacted by a company in Sydney, as a few staff members of a
company that are currently running Queuemetrics would like to see a fully
running installation for training and decision
I'm using 2.6 kernel on RHEL 4. Yes, all are of the same memory size.
Date: Thu, 23 Nov 2006 08:20:27 +0200
From: Tzafrir Cohen [EMAIL PROTECTED]
Subject: Re: [asterisk-users] More than one asterisk process
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type:
On a modern server without IDE drives, you dont even need RAID to
accomplish this. Problems arise at around 50-60 calls in my experience
(HPDL 360, 3Ghz, Gig of RAM and RAID 1 mirroring. I run a cron job that
checks files sizes and when they do not change within a specified period
of time,
I figure the issue is probably on their side... but just want to
figure out what.
When you say 'users hanging up' you mean your VOIP users... or people
who called in?
On 11/22/06, Tristan [EMAIL PROTECTED] wrote:
This happens when a call is offered to asterisk on a B-Channel that's
already
Hi,
try our latest beta version 6.5.2 which can be found here:
http://www.snom.com/wiki/index.php/Snom360/Firmware/Beta_Versions
http://www.snom.com/wiki/index.php/Snom320/Firmware/Beta_Versions
http://www.snom.com/wiki/index.php/Snom300/Firmware/Beta_Versions
Release Notes:
HI,
I'm not able to find some good doc or manual regarding Integration of
Asterisk with SER. Bacially, I want to forward my calls from SER to
asterisk. If some one already done this please guide me.
thanks in advance
arun
___
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Ok, now it works:
ideiafix:~# modprobe zaptel
ideiafix:~# modprobe wcte11xp
ZT_CHANCONFIG failed on channel 32: No such device or address (6)
FATAL: Error running install command for wcte11xp
ideiafix:~# modprobe wctdm
ideiafix:~# modprobe wcte11xp
Order to load: zaptel, wctdm, wcte11xp
Steve Totaro wrote:
Steve,
You neglet to mention:
Distro
Version of HylaFAX
Version of iaxmodem
Version of Asterisk
How you're connecting to the PSTN (From previous conversations, I'm
guessing PRI)
I can't say that I'm not experiencing the same issue as you, 99% of our
faxes
Earle Clubb wrote:
- What service provider/technology do you use for origination/termination?
- What hardware/software do you use and how does it all tie together?
- What tasks do you use * to accomplish?
- Any other pertinent info.
Until last summer I had Asterisk doing the normal call
Hello;
Maybe this is a little off-topic, but I need help. I need to repair a
cisco 7970, but in my country(spain) cisco is only selling, they don't
repair if you're not client. Because I bought on ebay, I'm not client,
so I have no chance.
I tried to repair by myself, the problem is on the LCD
Doug Lytle wrote:
Steve Totaro wrote:
Steve,
You neglet to mention:
Distro
Version of HylaFAX
Version of iaxmodem
Version of Asterisk
How you're connecting to the PSTN (From previous conversations, I'm
guessing PRI)
I can't say that I'm not experiencing the same issue as
Hi everybody,
I've installed future packages (asterisk 1.2 and freepbx) from
Xorcom's Repository in a debian etch, but when i want to uninstall
freepbx-panel, i got this error:
dialer:~# apt-get remove --purge freepbx-panel
Leyendo lista de paquetes... Hecho
Creando árbol de dependencias...
I am using asterisk to receive call from a DID provider . In configured
everything in freepbx properly and its working . I forwarded incoming calls
from did to a certain extension . Now i tried calling from another sip
provider to this box , when i call from other provider to my DID number
then
On Thu, Nov 23, 2006 at 12:47:27PM -0300, Lincoln Zuljewic Silva wrote:
Ok, now it works:
ideiafix:~# modprobe zaptel
ideiafix:~# modprobe wcte11xp
ZT_CHANCONFIG failed on channel 32: No such device or address (6)
FATAL: Error running install command for wcte11xp
ideiafix:~# modprobe wctdm
On Thu, Nov 23, 2006 at 10:32:44AM -0300, Ard wrote:
I'm using 2.6 kernel on RHEL 4. Yes, all are of the same memory size.
That's strange. What is the output of:
ps auxww | grep asterisk
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
Hi,
I must say that i'm not very used with customization of FOP. I've a box
runing Flash Op.Panel, and i notice that the screen is full of buttons from
my sip users, as well as Zapata channels.
The problem is that i have more Zapata channels as well as SIP users, is
there any way to get a
Lachek Butalek wrote:
My mission is to get one * box to dial another * box' extensions. I
have set this up previously without any issues by making a simple IAX
trunk/extension pair on the two boxes and create a dial plan with a
prefix like 9|XXX to select an extension on the other box.
My
MG == Michael Graves [EMAIL PROTECTED] writes:
MG Who will benefit as long as calls must typically pass into
MG existing PSTN infrstructure, and so be transcoded into G.711? It
MG seems to me that only systems that are IP end-to-end stand to show
MG the improvements...or am I mistunderstanding?
Benny Amorsen wrote:
MG == Michael Graves [EMAIL PROTECTED] writes:
MG Who will benefit as long as calls must typically pass into
MG existing PSTN infrstructure, and so be transcoded into G.711? It
MG seems to me that only systems that are IP end-to-end stand to show
MG the improvements...or
Hello;
Maybe this is a little off-topic, but I need help. I need to repair a
cisco 7970, but in my country(spain) cisco is only selling, they don't
repair if you're not client. Because I bought on ebay, I'm not client,
so I have no chance.
I tried to repair by myself, the problem is on the LCD
We're looking at using 4 or 8 port T1 cards with echo cancellation and are
evaluating brands to go with. We know that Sangoma has excellent solutions
especially when it comes to echo. But we still have to hear about actual
performance of a Digium card using the same Octasic DSP echo
Hi All,
I'm having a problem after reinstalling the operating system.
Festival works fine for SIP, but when IAX users are calling the same
extension they don't hear the festival and I see the next message on
console:
NOTICE[3996]: chan_iax2.c:2995 iax2_read: I should never be called!
Hello;
Maybe this is a little off-topic, but I need help. I need to repair a
cisco 7970, but in my country(spain) cisco is only selling, they don't
repair if you're not client. Because I bought on ebay, I'm not client,
so I have no chance.
I tried to repair by myself, the problem is on the LCD
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
CP
On Nov 22, 2006, at 8:40 PM, ram wrote:
Hi
where can i buy that Book
Ram
On 11/22/06, Patrick [EMAIL PROTECTED] wrote: On Wed,
2006-11-22 at 15:45 +0530, ram wrote:
[snip]
Nov 22 15:43:23 WARNING[14623]:
Just use Set(CALLERID(name)) in your dialplan - that's what we do.
CP
On Nov 23, 2006, at 12:00 AM, Eric Bishop wrote:
When we have calls that originate click-to-daial apps that use the
manager interface they always originate from asterisk is there any
way to change the from name?
Have a look at the OpenSER and Asterisk part of
http://openser.org/dokuwiki/doku.php
and
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER
Arun Kumar wrote:
HI,
I'm not able to find some good doc or manual regarding Integration of
Asterisk with SER.
Vincent Delporte wrote:
Hi
Most of our customers have generic names like Hospital, so I need
to
rewrite their caller ID name by looking up the number in a database
on
the Asterisk server, and rewriting the name such as Reading
Hospital
so that we know who's calling.
Any idea if this can be
Dear Asterisk People,
I am having problems putting a SIP image on a 7970. I was wondering if anyone
can help?
First problem is the phone is running version
Load IDJar70.2-5-47-17.sbn
Boot Load ID7970_64054100.bin Version5.0(0.6S)
So I did read that you couldn't simply put the latest SIP
I have a rather technical question here. I'm looking
at the code in app/app_voicemail.c, I'm wondering when
the vmauthenticate() function is called.
Aside from being called by load_module() as follows:
res |= ast_register_application(app4, vmauthenticate,
synopsis_vmauthenticate,
I have release my routines for PRI circuit monitoring. You, your client or
anyone can be notified by phone, beeper, email or txtmsg that your circuit
is down. If Asterisk crashes due to an oscillating circuit (as I have found
it sometimes does), sendmail is usually intact and email notification
This is the output.
[EMAIL PROTECTED] ~]# ps auxw | grep asterisk
root 4392 0.0 0.6 50604 13968 ? Ssl 11:02 0:00 asterisk
root 5050 0.0 0.4 38416 9268 ?S11:07 0:00 asterisk
root 5242 0.0 0.4 38528 9420 ?S11:09 0:00 asterisk
root 5495
On 22 Nov 2006, at 14:18, Adrian Marsh wrote:
[Adrian Marsh]
Thanks Tim,
Notransfer is commented out (so I guess means = transfer).
How does Asterisk know that the IN and OUT IPs are the same A*k box?
(They may not be I guess). If the IPs are different, wouldn't it need
to join the calls
Hello,
I'm using Slackware 11.0. I've installed unixODBC from the source files.
I've built and tested an odbc connection.
I'm trying to install Asterisk 1.4. I can't get it to recognize the
unixODBC installation. I've tried using the --with-odbc=/usr/local flag
to the configure process.
On Wed, 22 Nov 2006 19:20:54 +, Steve Kennedy wrote:
On Wed, Nov 22, 2006 at 06:58:13PM +0100, Huib van Wees wrote:
On 11/22/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Why Aastra phones use more electricity, i.e. 48VDC whereas other
phones use much less, e.g. Grandstream
Hi,
I'm deploying a SER + Asterisk architecture, where SER is used as SIP
registrar, and Asterisk is used for voicemail and PSTN gateway.
This system is already able to make Call Transfers (Blind and Attended)
internally between phones registered in SER, although, I can't make
Call
Hi everybody,
just to confirm that I understood it right (and that the info isn't
obsolete):
I have to store the voicemail audio data in an external mysql DB. In
http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage I
read that this is only possible via ODBC and *NOT* via native
At 22:07 22/11/2006 -0700, Marco Mouta wrote:
You can do it using AstDB, just load the database with callerid names and
numbers and then include a lookup on database in all incoming calls, so
you can override whatever you wanted:)
Thanks everyone. Indeed, it seems like using the embedded
On Thu, Nov 23, 2006 at 08:55:41PM +, Tim Panton wrote:
I've asked gradwell about my second point (still waiting...), but your
thoughts are the same as mine. In theory it should be ok, because I
have to authenticate the IAX connection with a username/password,
which
in turn they own
Hi List,
Can any one please let me know how to pass arguments to the agi script from
the dialplan?
I read that it is possible to pass arguments to an AGI script here,
http://home.cogeco.ca/~camstuff/agi.html, by entering the variable followed
by a vertical bar but it doesn't seem to work
Good Evening, does anyone have information regarding integration of
asterisk voicemail with an hotel management software called Fidelio
made by the Micros Company.
The integration can be either opensource or paid.
please contact me offlist if you want.
Thanks,
Erick.
eaperezh (at) gmail (dot)
Aww, come on... not everybody has been here for ages or read through
years of digests
Try the voip-info WIKI:
http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx
Regards,
Scott
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
I've been using Gmail and thought you might like to try it out. Here's
an invitation to create an account.
---
Matt has invited you to open a free Gmail account.
To accept this invitation and register for your account, visit
I'd like to have a list of variables used in Asterisk 1.4, and which ones
from v1.2 were deprecated/changed.
Ex. Since switching from 1.2 to 1.4, nothing shows up when I want to display
the value of ${TIMESTAMP}.
___
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How do I assign the MWI to a SIP phone on my asterisk server that is coming
from an ITSP?
I see the SIP message come across as having a message waiting but how does
one get that
to go to an extension on my box.
Thanks
Tom
___
--Bandwidth and
Greetings,
I have tried with all conceivable means to get my asterisk (called a in this
discussion) to have two SIP user agents (called ua1 and ua2 in this discussion
running SJPHONE actually) to communicate directly with one another using RTP.
No matter what I do, the RTP traffic always
Paul wrote:
I have release my routines for PRI circuit monitoring. You, your
client or anyone can be notified by phone, beeper, email or txtmsg
that your circuit is down. If Asterisk crashes due to an oscillating
circuit (as I have found it sometimes does), sendmail is usually
intact and
I had the same issue, phone was working fine but 'sip show peers' didn't
show any phone registered. The reason was no sip registrar server was given
in the config or in web UI. For aastra phones, you need to specify proxy and
registrar servers separately.
So in aastra.cfg, you need to enter the
I went with FreeTDS to accomplish this at one point and it worked great
in Dev (no call volume). It seemed to work better than ODBC since it is
speaking with M$ SQL natively rather than through an additional layer
although there is much debate about this on the net.
We were doing a bunch of
Hi
i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect
my asterisk to a french analog line.
In my zaptel.conf, i have:
loadzone=fr
defaultzone=fr
fxols=3
when i load the module, i have in logs:
Nov 24 06:13:40 gw kernel: Freed a Wildcard
Nov 24 06:13:40 gw kernel: Zapata
Hi all
i try to run misdn with asterisk on an Fedora Core 6 x64 System
but after a installation of all the driver for MISDN with no errors.
I get the following errors in the Full log from Asterisk
logger.c: [app_exec.so]Nov 21 20:50:25 VERBOSE[21401] logger.c:
[app_exec.so] = (Executes
Dear Users,
I am fairly new to Digium and Asterisk. I wanted to know that if I use the
Digium product THREE Digium Wildcard TE412Ps (Quad E1 Card) how many calls
can I handle simultaneously.
I want to use the cards with the following Configurations:
Intel® Xeon 3.00GHz/800MHz, 2M Processor
We have placed 2 X 4 port cards in a Dell 2950 and it worked well - even
when it was recording 50% of the calls.
PaulH
On Fri, 2006-11-24 at 11:54 +0600, Imran M Yousuf wrote:
Dear Users,
I am fairly new to Digium and Asterisk. I wanted to know that if I use
the Digium product THREE
Dial() cmd seams unable to detect caller hangup?
so if the call file land in a exten, for example:
[callfile-landing]
exten=1,1,dial(SIP/XXX)
exten=1,n,hangup
when caller after conversation and hangup, the dial cmd is unable to detect
that and it will ring the caller and called party 2
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