In article [EMAIL PROTECTED],
Maxim Litnitsky [EMAIL PROTECTED] wrote:
Hello all.
I am trying to use app_mysql.
It works for selects and functions, but does not want to work with
procedures.
Pls have a look:
Please could you post the relevant sections of your dialplan?
Calling function:
In article [EMAIL PROTECTED],
Steven [EMAIL PROTECTED] wrote:
This is great news.
Agreed!
Previously, stable was just considered a snapshot and if you ran
stable and encountered a bug, you had to switch to head to get the
fix.
I don't think this is correct. Pure bug fixes were always
Hi, all
Can someone tell me where to tell asterisk no to use 127.0.0.1 IP
(localhost)?
When I am registering with VoIP providers, they get my info as [EMAIL PROTECTED]
(This is SIP registration).
Also, in SIP logs, when calling I am getting things like this:
Executing
in your sip.conf
bind it to the ip you want to bind it , change the value from 0.0.0.0 to the ip
u want to
bind it to.
Rehan
Hi, all
Can someone tell me where to tell asterisk no to use 127.0.0.1 IP
(localhost)?
When I am registering with VoIP providers, they get my info as
[EMAIL
Maybe you have not configured correcly your sip.conf
externip=your_external_ip
try this
RumaTech escribió:
Hi, all
Can someone tell me where to tell asterisk no to use 127.0.0.1 IP
(localhost)?
When I am registering with VoIP providers, they get my info as
[EMAIL PROTECTED]
(This is SIP
Hi,
is the following possible? I would like to transfer a call to my
personal MeetMe conference room and get transferred there
automatically as well. Currently I can transfer the call to the
conference, have to hangup and then call the conference number myself. I
would love to have this in one
In article [EMAIL PROTECTED],
Koopmann, Jan-Peter [EMAIL PROTECTED] wrote:
Hi,
is the following possible? I would like to transfer a call to my
personal MeetMe conference room and get transferred there
automatically as well. Currently I can transfer the call to the
conference, have to
ola,
ok, i must give a example:
i call with analog-telephon trough a sip adapter my dsp from
asterisk-server, (Console/dsp),
my dsp accept the call, the line is open,but after i want to execute a
application/or change to another line directly without to hangup. And it
seems , when a line is
Dan Sully wrote:
We've been given a block of 23 numbers for the PRI. If I explictly set
the
incoming extension in extensions.conf like:
exten = 1153,1,Answer
or:
exten = _,1,Answer
I can get the incoming call. If I try and do:
exten = s,1,Answer
Why would an incoming call have a
Hi!
1 PRI to Telco
1 PRI to old PBX
Several SIP phones with the intention of having approx. 200.
Currently the traveling users have to VPN in first which I am sure is adding
extra overhead to the calls.
I have yet to open my server to the Internet to be accessible to travelers
without
I have two internet connections, the first is clean but the second, a
Cable TV connection, has a lot of jitter on it. When the first fails,
which it frequently does, I would like to fail over to the Cable
connection and put a static route to another provider. However, to use
it, I will need to
Hello,
I recently installed Asterisk 1.2.2 on one of our servers and it was
doing fine until this morning when it started to Rotate the
logfiles(event_log and messages[/var/log/asterisk/]) every few
seconds:
Rotated Logs Per SIGXFSZ (Exceeded file size limit)
They would never get more than
- Original Message -
From: Brian Roy [EMAIL PROTECTED]
Newsgroups: gmane.comp.telephony.pbx.asterisk.user
Sent: Friday, January 20, 2006 6:12 PM
Subject: Re: Dell PowerConnect 2724 Switch and QoS for VOIP?
On 1/20/06, Sean Tempesta [EMAIL PROTECTED] wrote:
Has anyone had any
I have this problem intermittently. I haven't found a solution yet. I
don't suppose you've tried using a different NTP server? I haven't tried
that either, but I might this afternoon. For me, it's hard to troubleshoot
because it doesn't happen all the time.
-ross
-Original Message-
Thanks! I'd love to see your extensions.conf file.
I appreciate it.
Tom
On Jan 20, 2006, at 8:31 PM, Alexander Lopez wrote:
Contact me off list, I have a sample extensions.conf file that I can
share. It has Paging (one to one and One to Many)
Ivr includes, time of da routing and it is
I know this is a somewhat odd application, but we have a very good
reason for needing it.
Basically, I want asterisk to automatically redial a caller unless they
exit the system properly.
Here are some pertinate sections of the dialplan.
[AUTOBCSTART]
EXTEN=001,1,Meetme(${ENC}|pq)
Christian,
Why is this this setting on by default? I don't understand why anyone
would want this behavior. -Mike
Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
-Original Message-
From: Christian Stredicke [mailto:[EMAIL
sip debug
rtp debug
enable all the log levels in console in logger.conf
regards
On 1/20/06, RumaTech [EMAIL PROTECTED] wrote:
Hi, all
I am trying to call to particular destination via SIPNET (one of the VoIP
providers).
I can succesfully dial and I can hear waiting tone, however nothing
check incominglimit and outgoinglimit in sip.conf http://www.voip-info.org
On 1/20/06, Kristian Larsson [EMAIL PROTECTED] wrote:
Hey!
I'm having a small problem. I'm using Realtime to
store SIP account information. Dialing works just
fine, but when dialing a person already on the
phone I
Was there a resolution to this issue? The GXP-2000 seems to be a very
popular phone, so I can't imagine others on the list not experiencing
this? Or is this part of a batch with unresolvable problems that I need
to send back to the seller?
Well, I'm using dozens of these phones without this
The idea was that passwords will not be provisoned automatically, you
must enter them manually on the phone. Which makes sense in scenarios
where you completely automatically provision phones and hand out the
password to the users.
But maybe you are right, we should turn this off by default. I
After making some research, I am done.Thanks!yrving rivas [EMAIL PROTECTED] escribió:Hi all:I just bought and installed a TDM04B (4 fxo ports). It is running and ok. I have 2 lines from my provider. I can make two incoming calls come through my box to its extensions, but only one
* Doug Lytle shaped the electrons to say...
exten = 1153,1,Answer
I can get the incoming call. If I try and do:
exten = s,1,Answer
Why would an incoming call have a destination of 1153? My incoming
don't have a destination until the end user selects something from and
IVR or and operator
Hi,
I just upgraded by 1.0.x home server to 1.2.2. Overall the upgrade went
fine, but a strange problem has cropped up with the CALLERID name value of
incoming calls from the X101P card. When an incoming call is presented (via
Vonage ATA), the calledid value getting double quotes up from:
--
Hi Tony,
Look at the option 'G(context^exten^pri)' in the Dial application.
Thanks for the hint but I am not sure if this will help me. Either I am
too blind to see the solution or I stated the question in an unclear
way. :-) What I want is this:
1. Customer calls me or I call customer.
2. In
Having learned the hard way...here's something I noticed in the code you
posted:
callerid=My name 404 xxx
Try changing to:
callerid=My name 404 xxx
Some subtle differences between the bash and asterisk interpreters can cause
experienced users to scratch their heads...(unless there is
On Saturday, January 21, 2006 1:44 PM Tony Mountifield wrote:
- edit the code to provide another option to turn off that message, or
http://bugs.digium.com/view.php?id=6316
Regards,
JP
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Hello All,
What is the advantage of jitterbuffer on
zap channel? do you have any suggestion on the setting for home usage? Is there
anydisadvantage in using it?
Cheers,
Anto
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[EMAIL PROTECTED] is believed to have said:
Hey ho,
A few days ago we released the linux version of the phone, today we are
very happy to have the mac version ready for a little field test.
Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php
At the same time, we also
Hi,
I am trying to connect the one cellular whit my asterisk box, with a
cable usb of cellular me, to be able to call from the asterisk, someone
has proved this?
Carlos Rojas
Lima - Peru
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Hi Michelle,
Thanks for the response. I made the change but it didn't make a difference.
I'm sure you're right that these is something amiss in my configuration
files, but I'm not sure where. You'd think that Zapata.conf and
extensions.conf would be the only ones.
Anyway, I have a quick fix in
I have tried my Sony Ericsson T616 cell with Cingula service connected via phonelabs Doc-N-Talk and it works great with asterisk.
Doc-N-Talk is connected to x100p FXO port which received incoming cell calls and routes them to asterisk and the same port routes outbound calls from asterisk through
We are considering it yes, but i don't know how hard or easy it would be.
I guess we will first try to make the other versions like we want them
to be and then start looking at other os'es.
Zoa
Aldo Bergamini wrote:
[EMAIL PROTECTED] is believed to have said:
Hey ho,
A few days ago
In article [EMAIL PROTECTED],
Koopmann, Jan-Peter [EMAIL PROTECTED] wrote:
Look at the option 'G(context^exten^pri)' in the Dial application.
Thanks for the hint but I am not sure if this will help me. Either I am
too blind to see the solution or I stated the question in an unclear
way.
Could anyone please help me with that:
I have an analog telephone connected to Asterisk using a Sipura 2002 ATA. When
calling the extension, the caller ID presented is always the number of that
extension rather than the number of the calling one.
While I learned that this is the new standard
Hello All ,
On Sat, 21 Jan 2006, Alberto Sagredo wrote:
Maybe you have not configured correcly your sip.conf
externip=your_external_ip
try this
RumaTech escribió:
Something right down this alley .
What happens if I have more than one interface I want asterisk
to
Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk
server somewhere where there was no NAT for the * box that the SIP phones
wouldn't create any issues.
How do you people with Hosted PBX handle the deployment of SIP phones behind
NAT firewalls? Is it just elbow grease
I can confirm that this is the issue. I now have to toggle it off manually
on 120 phones. I can tell you, in the real world, you don't hand out
passwords to users for their phones, they will not understand why you need a
password for a phone. You may want to consider changing the default
settings.
Hello,
Saturday, January 21, 2006, 7:34:50 PM, you wrote:
3. I transfer the call to my personal MeetMe room. In this step I
would like not only the customer but also me to be connected to the
MeetMe room automatically. Basically I can continue to chat with the
customer without him noticing
On Saturday 21 January 2006 20:30, Conrad Beckert wrote:
Could anyone please help me with that:
I have an analog telephone connected to Asterisk using a Sipura 2002 ATA.
When calling the extension, the caller ID presented is always the number of
that extension rather than the number of the
Thank you very much for the help, it seems to me that I have seen it in wiki of asterisk
RegardsOn 1/21/06, Nilesh Londhe [EMAIL PROTECTED] wrote:
I have tried my Sony Ericsson T616 cell with Cingula service
connected via phonelabs Doc-N-Talk and it works great with asterisk.
Doc-N-Talk is
Hello,
I have 2 asterisk servers,
1 - is the main server doing all the dial out.
2 - is located on a site, sendig all the calls via IAX2 to the main server.
They worked fine, but suddenly I see the below:
Error on server 2:
chan_iax2.c:5550 socket_read: Call rejected by 192.168.0.2: No
We have PSTN lines connected to FXO lines of a TDM400B. I just got a
complaint that overseas callers who are using cellphones sometimes find
that DTMF digits aren't working - they press digits and the menu goes on
as if they hadn't pressed anything. Since it sometimes works, and other
IVRs
Hi,
I flashed my ATCom AT320 phone (PA1888S based) with IAX firmware instead of
SIP but now call transfer doesn't work neither using phone buttons nor using
Asterisk features.
I heard that it can be a real problem.
Any help?
Mimmus
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We're trying to use some AudioCodes MP104 FXO units as gateways to
Asterisk but cannot get them to reliably detect DTMF. Some landline
calls get most digits but some are duplicated. Some cell phone calls get
0% DTMF recognition.
Anyone with experience with these units have any suggestions?
The version 1.2 Dial() command does not use the n+101 jumping
behaviour by default. I know about the j option and setting
priorityjumping=yes as described here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial
But if I use the default behaviour does that mean I have to check the
Wouldn't it be easier to keep the agents in the table all the time, and
simply update the logged_in status column for that row?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Tuesday, August 30, 2005 12:23 PM
To: Asterisk Users
Hello all,
I am working on a creating some intelligent failover dial-plan
logic and I'm running into something that I'd like some feedback on.
Basically, it appears that if you place a call to an IAX2 peer that
refuses the connection, or is unavailable, a NOANSWER dialstatus is
Setup your hostname correctly. Asteisk is getting its hostname from the
hostname function call and using that from then on.
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of RumaTech
Sent: Saturday, January 21, 2006 5:27 AM
To: Asterisk Users
I think the solution needs a little more thinking about
I am reliefed. I almost thought I had missed something that obvious...
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To UNSUBSCRIBE or update
I am trying to move from IAX2 to SIP for voipbuster, moving at the same
time to sip1.voipbuster.com.
When I try calling out, I see that there is SIP exchange, and in many
cases also RTP data being exchanged.
Hover in a very large number of attempts the connection is not
established. Half of the
Most often the simple addition of nat=yes in the relevant sip.conf
stanza is all that's required to make a remote SIP phone work from
behind a firewall.
for example
[2201]
user=blah
secret=blah
auth=blah
allow=blah
host=dynamic
nat=yes
I've been running 4 remote SIP phones across the
On 1/19/06, Douglas Garstang [EMAIL PROTECTED] wrote:
The O'Reilly TFOT book is full of errors. Two that pop into my head instantly
are it's referring to regcontext being able to execute dialplan commands upon
SIP registration and it's use of auth= in sip.conf in the DUNDi section. I
Contact me off list if interested.
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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Relax your PRI is fine. What Xo is sending you is 4 digits os DID.
If for example you have 1130-1153 as the last 4 digits of your Number
you can use this to rout your calls.
exten = 1130,1,Goto(ivr,s,1)
Exten = 1140,1,Goto(extensionss,100,1)
Exten = _X.,1,Zapateller
Is the above config.
Has anyone successfully Installing the none commercial intel g729 codecs
into [EMAIL PROTECTED] 2.2?
I tried to follow the instruction from
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ and
http://aussievoip.com.au/tiki-index.php?page=G729-Install but I can't. I
did it
On Sat, 21 Jan 2006 08:32:26 -0500
Doug Lytle [EMAIL PROTECTED] wrote:
Why would an incoming call have a destination of 1153?
On my asterisk, when a call comes from E1 the default destination is the last 4
digits.
--
Iuri Gomes Diniz adm.iuri (at) digi.com.br
Network Admin and Programmer
Hi!
is the following possible? I would like to transfer a call to my
personal MeetMe conference room and get transferred there
automatically as well. Currently I can transfer the call to the
conference, have to hangup and then call the conference number myself. I
would love to have this in
* Alexander Lopez shaped the electrons to say...
Relax your PRI is fine. What Xo is sending you is 4 digits os DID.
If for example you have 1130-1153 as the last 4 digits of your Number
you can use this to rout your calls.
exten = 1130,1,Goto(ivr,s,1)
Exten = 1140,1,Goto(extensionss,100,1)
Can any body give me an example how to configure h323 in Asterisk.
Which files do I need to configure? just extensions.conf and h323.conf ?
Thanks,
Patricio
_
Descubre la descarga digital con MSN Music. Más de un millón de
Hello,
I'am configuring my asterisk (located behind a dsl router with nat) to act as
gateway between sip and an home switching system connected over a X100P
compatible winmodem.
I use register = to register asterisk as client at a SIP proxy.
If i use the Dial command in extensions.conf to
On Sat, January 21, 2006 22:10, MapsAir said:
Has anyone successfully Installing the none commercial intel g729 codecs
into [EMAIL PROTECTED] 2.2?
I tried to follow the instruction from
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ and
Hi,
MapsAir wrote:
Has anyone successfully Installing the none commercial intel g729 codecs
into [EMAIL PROTECTED] 2.2?
Installed them today. Installing from source didn't work for me (Debian,
Asterisk 1.2 from svn) but just adding the binaries (see the wiki on
voip.org) did the job. Have you
On Sat, January 21, 2006 23:21, Franz Bräuer said:
Hi,
MapsAir wrote:
Has anyone successfully Installing the none commercial intel g729 codecs
into [EMAIL PROTECTED] 2.2?
Installed them today. Installing from source didn't work for me (Debian,
Asterisk 1.2 from svn) but just adding the
Greetings!
On January 19, 2006, I featured VoIP and open source telephony pioneer,
Mark Spencer, on my podcast, Technology Coffee. To listen to this interview, visit http://www.ronaldlewis.com/coffee.
Also, Tom Keating, CTO and VP at TMC Labs, has blogged about it as well.
On Sat, 21 Jan 2006, Christian Stredicke wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael Welter
Sent: Friday, January 20, 2006 7:01 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re:
Hi guys,anyone knows how to disable the WARNINGS in cli, i set verbose 0 butthe warning still show..Thanks,Lito
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I've the same problem with sip1.sipdiscount.com. The calls are not
connecting but are billed.
Con fecha 21/1/2006, Francesco Peeters (Asterisk)
[EMAIL PROTECTED] escribió:
I am trying to move from IAX2 to SIP for voipbuster, moving at the same
time to sip1.voipbuster.com.
When I try calling
Dan, Basically you go send tip send tip, receive tip receive tip and
so forth on both connectors.
I needed to use one cat 5 out of the ec to the channel bank, and one
crossover t1 cable to the zaptel card. Receive in means receive
basically (because it's the incoming signal).
You should be
Hello Dan,
Have a look at this link:
http://www.adcomcorp.com/asterisk/tellabs
I got those pictures up there, may be of help. In essence, 1 pair is
either a tx pair or an rx pair.
If I recall, Orange and Green should be one side, and blue brown should
be the other side.
I tried to upload
I used:
cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds
iaxyprov astcc
and in the same order I try to compile it.
Asterisk ends with the lines below. It complains of a newer libpri, but
I just did it a step before!
What do I miss?
chan_zap.c:62:2: #error You need newer
HAHAHA!
That's EXATCLY the same setup I'm running...even down to the cards in
the 600
Working like a champ.
-d
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Saturday, January 21, 2006 8:06 PM
To: Asterisk
Con fecha 21/1/2006, Francesco Peeters (Asterisk)
[EMAIL PROTECTED] escribió:
On Sat, January 21, 2006 23:21, Franz Bräuer said:
Hi,
MapsAir wrote:
Has anyone successfully Installing the none commercial intel g729 codecs
into [EMAIL PROTECTED] 2.2?
I'm using g723.1 and works very well.
How about when you have four or five SIP devices at a single location? Do
you manually assign each phone a separate port and add firewall/router
rules? I am looking for an inexpensive device or method that will allow
this happen automatically. Rather than going that route, my current
solution
Ronald Hartmann wrote:
Anyone know if it is possible to control how aggressively the
Aggressive mode behaves.
Meaning, is it possible to dial back the aggressive mode to have a happy
medium between
Regular and the Aggressive defaults.
I have a situation where Normal echo cancellation is not
Mike Myers wrote on Friday, 20 January 2006 1:56 PM:
Hi. I am thinking about building an asterisk system for a small business
and want to be able to page through the phones. It seems like to do this
asterisk needs auto-answer support in the phone. I know the SPA-841's
support this, as
Trevor G. Hammonds wrote:
How about when you have four or five SIP devices at a single location? Do
you manually assign each phone a separate port and add firewall/router
rules? I am looking for an inexpensive device or method that will allow
this happen automatically. Rather than going
On Sun, 2006-01-22 at 09:14 +0800, Ronald Wiplinger wrote:
I used:
cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds
iaxyprov astcc
and in the same order I try to compile it.
Asterisk ends with the lines below. It complains of a newer libpri, but
I just did it a step
I have searched and found a couple examples on how to put the app_cepstral.c
into Asterisk but it isn't working. I obviously am not understanding the
examples that I have found.
Copy the app_cepstral.c file to your asterisk source tree (apps folder) .
You'll also need to add a lines like these
On Sat, 2006-01-21 at 23:21 -0500, Lists wrote:
In file included from app_cepstral.c:15:
/usr/include/asterisk/file.h:27:2: #error You must include stdio.h before
file.h!
In file included from app_cepstral.c:15:
/usr/include/asterisk/file.h:56: error: syntax error before '*' token
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter aka
Bret McDanel
Sent: Saturday, January 21, 2006 11:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Compiling app_cepstral.c into Asterisk -
failing
On Sat, 2006-01-21 at 23:51 -0500, Lists wrote:
Thanks for that tip. I made that correction and now it is stopping
at:
gcc -D_GNU_SOURCE -shared -Xlinker -x -o app_cepstral.so
app_cepstral.c -lz
-lm -lswift -lceplex_us -lceplang_en -lz -ldl -L/opt/swift/lib
-I/opt/swift/include
In file
Dave Cotton wrote:
On Sun, 2006-01-22 at 09:14 +0800, Ronald Wiplinger wrote:
I used:
cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds
iaxyprov astcc
and in the same order I try to compile it.
Asterisk ends with the lines below. It complains of a newer libpri, but
I
Aloha,Anyone know how to disable call forward on a Polycom Phone. Calls being accidentilly being forwarded somewhere is the #1 trouble that we have to respond to.The real issue is the 'end call' button becomes 'forward' when the call endstherefore the user thinks they are pressing 'end call'
On Sun, 2006-01-22 at 13:55 +0800, Ronald Wiplinger wrote:
Dave Cotton wrote:
On Sun, 2006-01-22 at 09:14 +0800, Ronald Wiplinger wrote:
I used:
cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds
iaxyprov astcc
and in the same order I try to compile it.
Hey, all. I've got a non-PRI T1 that doesn't do DID correctly: I can't
get the DID from the proper variables, and, instead, I direct it based on
the four least valuable DTMF digits dialed by the T1 for in-bound calls.
Which really works pretty well; Asterisk plugs them quite nicely into
1. At the end of compiling asterisk I got a lot of warnings. How can I
solve that?
I used:
# svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel
# svn checkout http://svn.digium.com/svn/libpri/trunk libpri
# svn checkout http://svn.digium.com/svn/asterisk /trunk asterisk
# svn checkout
see inline
The version 1.2 Dial() command does not use the n+101 jumping
behaviour by default. I know about the j option and setting
priorityjumping=yes as described here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial
But if I use the default behaviour does that mean I
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