Re: [asterisk-users] a problem on Ubuntu with Asterisk

[Rodrigo Gonzalez]

Scott Berry wrote:
> Have a nice day,
> Scott Berry
> E-mail:  [EMAIL PROTECTED]
>
> I  am studying out of the book Asterisk:  The Future of Telephony  on
> Chapter 4,   and right now for practicing using the built in Debian
> version of Asterisk for Ubuntu.  I am however having some problem where
> I cannot do "asterisk -r" and hook up to the asterisk CLI.  I have
> checked to see that "/var/run/asterisk/asterisk.ctl" is available which
> it is.  I have also set up the zaptel.conf, zapata.conf and also the
> extensions.conf as specified in the book.  The error I get is:
>
> "Unable to connect to asterisk remote
> (does /var/run/asterisk/asterisk.ctl exist?"  Yes it certainly does.
> Any help would be appreciated.  if need be i would be happy to send my
> extensions.conf, zaptel.conf, and zapata.conf to the lisOne other
> question I think I am correct on this but not sure does zaptel.conf and
> zapata.conf go in to "/etc?"
>
> Thanks for all the help.
>
>
>
>
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Permissions problems?

sudo asterisk -r


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Re: [asterisk-users] Differents routes for differents extensions

[Rodrigo Gonzalez]

Create different contexts and assign them to the extensions

[trunk1]
exten => .X,1,Dial()

[trunk2]
exten => .X,Dial()

and in sip.conf or iax.conf

[exten1]
...
context = trunk1

[exten2]

context=trunk2

Marco escribió:

Hi everybody,
I need to use different outbound routes from calls started by different
extensions; I mean, that the extension "A" when dialing "011543..." has
to get access always on the 1st trunk, the extension "B" when dialing
another number has always to access the outside world on the 2nd trunk,
and so on.
Some kind of solution I thought involved the use of a "fake dialcode",
whic is prepended to the dial number and then stripped from an "Outbound
route" section (and then the trunk is dialed):

ext. A "call 011543..." >  prepend 41 ---> Outbound route for "41|."
---> Appropriate trunk dial "."

The only matter is that I have NO clue on where to append this code for
outgoing calls from these specific extensions.
If anyone has a simpler idea (yeah, mine is pretty twisted %) ) or knows
how to put a code before the dial string of an extension, let me know!
Thanks in advance,

Marco



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Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

[Rodrigo Gonzalez]

Matt Watson escribió:

There is a .NET 1.1 library out there... I've played with it a little bit, but 
not enough that I could comment on how feature rich or stable it is...

http://www.voip-info.org/wiki/view/Asterisk+.NET

It'll more than likely not be compatible with AMI 1.1 however, which I believe 
is included in ast 1.6

--
Matt

  

Do you, or someone else,  know where to get some example about using it?

Thank you



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Re: [asterisk-users] callers in queue passed to agents who accept only one call at a time

[Rodrigo Gonzalez]

calllimit in sip.conf and you are done

Vieri escribió:

I have a queue I configured as "strict" and a cron
script I use to QueueAdd and QueueRemove agents
according to my company's requirements. Usually I have
2 or 3 agents at a time and the ring strategy is
ringall.

These agents use non-open-source Windows softphones
that do not let you configure it so that if they're on
the phone, a second call will be rejected (agent
busy). Instead, it's as if they had call waiting and
incoming calls keep popping up while they're
conversating with the first caller and they would like
to avoid this.

I guess the easiest solution would be to find an
open-source or free softphone that can be configured
to accept only one call at a time (currently using
SJphone).

Another solution would be if I could tell the Queue()
application that if an agent is InUse then don't pass
the call.

Still another yet more delicate solution would be to
have a custom script "receive" manager events related
to the queue which in turn replies with an agi
command. For example, whenever an agent answers a call
I think that an event such as QueueMemberStatus can be
triggered (although I don't know how). If the custom
script could receive this event in realtime then it
would run an agi command such as
QueueRemove(busyagent...). When the agent is free
again I suppose the same event is triggered and the
custom script can QueueAdd(freeagent...).

Could anyone please give me some pointers on this?

Thanks!

Vieri




  

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Re: [asterisk-users] Send received fax to different email account

[Rodrigo Gonzalez]

mark morreny escribió:

Dear all,

I am able to send and receive fax with Asterisk + iaxmodel + hylafax.  
What I want to be able to do is to

1. Stored the received fax in mysql
2. Send an email notification to he user corresponding to the incoming 
phone number

3. Send a SMS notification to the user's mobile phone

In the hylafax setup, it seems like it can only send email to one 
destination email address.  Is there something that can be configured?
If hylafax can't do it, can anyone suggest another open source package 
that can accomplish the requirement?
procmail in the receiving email account can do that and maybe moreI 
mean, use hylafax sending to [EMAIL PROTECTED] and at xxx.com use procmail to 
do whatever


Thanks alot in advance for your help.   Your help is greatly appreciated.

Thanks,
Mark


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Re: [asterisk-users] Asterisk with lumenvox

[Rodrigo Gonzalez]

Josué Conti escribió:

Hello all, how are you?
I would like to know from someone uses or has used the engines of
LumenVox for integration with the asterisk for voice recognition.

Best Regards

Josué

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I've configured for a customer. What do you need to know?



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Re: [asterisk-users] Entering code to restart the machine

[Rodrigo Gonzalez]

Steve Totaro escribió:

On Wed, Feb 27, 2008 at 10:26 AM, Alex Balashov
<[EMAIL PROTECTED]> wrote:
  

bilal ghayyad wrote:
 > Hi All;
 >
 > How can I configure Asterisk in that way:
 >
 > If I entered code (from my mobile when I call to the
 > Asterisk or from any Internal Phone), then the machine
 > do restart. I need this when I am far from the office
 > and I need to restart the machine and I do not have
 > Internet connection.

 The safest way is to call an AGI script (or System command or whatever)
 from the dial plan that uses some IPC mechanism to pass a message to an
 outside cron job that runs periodically and checks for this flag, and if
 so, restarts asterisk from the CLI ('asterisk -r -x 'stop now'; sleep
 10; asterisk).

 The sleep is desirable because the 'stop now' command is asynchronous
 and Asterisk does not necessarily shut down instantaneously when you
 issue it.

 --
 Alex Balashov
 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (706) 338-8599




The way I read the OP, he wishes to reboot the box, not just restart
asterisk.  In which case, simply having
exten=777,1,Authenticate(whatever)
exten=777,n,System(reboot)

Would be the easiest and somewhat protected way of rebooting.

Thanks,
Steve Totaro
  
If asterisk is not running as root allow asterisk user to reboot adding 
to /etc/sudoers for example and...

exten=777,n,system(sudo reboot)




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Re: [asterisk-users] Restricting registration for peer 'iaxmodem0' to 60 seconds

[Rodrigo Gonzalez]

Michelle Dupuis escribió:
I have setup hylafax today, along with iaxmodem.  I'm just starting 
the debugging process and see the following message every 60 seconds:
 
[Feb 19 17:44:16] NOTICE[1996]: chan_iax2.c:6021 update_registry: 
Restricting registration for peer 'iaxmodem0' to 60 seconds (requested 
300)
 
Can someone tell me what this means?  Why is it there?  And how do I 
get rid of it!
 
Thanks,

MD
Mean that maxregexpire in iax.conf is defined as 60, change it if you 
want to allow iax phones to request a longet expiration




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Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)

[Rodrigo Gonzalez]

Olivier wrote:
>
> 2007/6/15, Steve Underwood <[EMAIL PROTECTED] 
> >:
>
> ...
>
> The t38modem
> program from openh323 does this, and it has to do some nasty
> things to
> work. :-\
>
> Steve
>
> Is this openh323 project alive ?
> Latest news date from 2003 (http://www.openh323.org/) !
> 
>
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yes, in sourceforge

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Re: [asterisk-users] MeetMe Error

[Rodrigo Gonzalez]

Manolet Gmail wrote:

2007/4/18, Ronaldo <[EMAIL PROTECTED]>:

Hi Manolet,

You have to install zaptel in order to make MeetMe application to work.
MeetMe needs a kind of timer device that is provided by zaptel package.
Eventhough you don't have a zaptel card you need to install its package.

Search for MeetMe application in http://www.voip-info.org/ and you will
find documentation about how to do that.

Good Luck.

Ronaldo

Manolet Gmail wrote:
> Hi! i have an error using the meetme aplication, and just dont work..
> my meetme.conf is:
>
> [rooms]
> conf = 700
>
> i calling from a sip phone, the extension number is 600. there is the
> error:
>
> Executing [EMAIL PROTECTED]:1] MeetMe("SIP/600-09111e58",
> "700|MI") in new stack
> WARNING[20055]: channel.c:3024 ast_request: No channel type registered
> for 'zap'
> WARNING[20055]: app_meetme.c:755 build_conf: Unable to open pseudo
> channel - trying device
> WARNING[20055]: app_meetme.c:758 build_conf: Unable to open pseudo 
device

>  Playing 'conf-invalid' (language 'es')
> Spawn extension (numberplan-custom-1, 700, 1) exited non-zero on
> 'SIP/600-09111e58'
>
> i dont have any zap interface. how to solve this?
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but i have zaptel 1.4.1 installed... there is any special
configuration or something?
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enable ztdummy and compile again
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Re: [asterisk-users] MeetMe Error

[Rodrigo Gonzalez]

Manolet Gmail wrote:

Hi! i have an error using the meetme aplication, and just dont work..
my meetme.conf is:

[rooms]
conf = 700

i calling from a sip phone, the extension number is 600. there is the 
error:


Executing [EMAIL PROTECTED]:1] MeetMe("SIP/600-09111e58",
"700|MI") in new stack
WARNING[20055]: channel.c:3024 ast_request: No channel type registered 
for 'zap'

WARNING[20055]: app_meetme.c:755 build_conf: Unable to open pseudo
channel - trying device
WARNING[20055]: app_meetme.c:758 build_conf: Unable to open pseudo device
 Playing 'conf-invalid' (language 'es')
Spawn extension (numberplan-custom-1, 700, 1) exited non-zero on
'SIP/600-09111e58'

i dont have any zap interface. how to solve this?
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Compile and install ztdummy from zaptel package, I think that will fix 
your issue

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Re: [asterisk-users] a2billing

[Rodrigo Gonzalez]

Pezhman Lali wrote:

hi
My a2billing adds "|HrL" automatically to dial string,
I can not find the source of this task,
I need to remove "r" from all dial strings,
Thanks for your help.
Best
Mani

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/etc/asterisk/a2billing.conf

Check there, the dial string is in the conf file
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[asterisk-users] transfers and CDR

[Rodrigo Gonzalez]

Hi everybody,

A question, how do I follow a call that is transferred? is the any event 
or something in the CDR that would let me find all the call sequence?


Thanks

Rodrigo
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Re: [asterisk-users] this i a test

[Rodrigo Gonzalez]

Bayrouni wrote:

Sorry for disturbing, but I sent some messages today and I am not seeing
them on this list.
Can sombody tell me, in case this message appear on the list.

Thank you
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Yes, here

Go to http://lists.digium.com/mailman/listinfo/asterisk-users

Login and check that you have Receive your own posts to the list? in yes 
if you want to receive your own emails

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Re: [asterisk-users] h323 how to set it up?

[Rodrigo Gonzalez]

Florea Igor wrote:

Hi all,
I have some questions about h323. Is it mandatory to install a oh323 or I can 
do h323 calls without patching or adding any new drivers ti asterisk?
I did compile the asterisk with channel driver chan_h323 but it still gives me 
this error:
[Feb 28 18:12:58] WARNING[1902]: app_dial.c:1081 dial_exec_full: Unable to 
create channel of type 'H323' (cause 66 - Channel not implemented)

what shoul I do to have it implemented?
Can somebody recommend some references on how to set up h323 ?
Thx,
Igor

This message was scanned by Barracuda Networks
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 Read README file in channels/h323
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[asterisk-users] jingle + asterisk 1.4

[Rodrigo Gonzalez]

Hi,

can someone give me a link to a howto about that?

I want to use jabbin with asterisk but dont find how to register jabbin 
client in asterisk so it can make calls.


Thanks

Rodrigo
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Re: [asterisk-users] "Unable to launch Sendmail" warning

[Rodrigo Gonzalez]

Olivier wrote:

Hi,

 From a bristuffed 1.0.8 Asterisk on a Gentoo system, I've got this :
WARNING[8094]: Unable to launch '/usr/sbin/sendmail -t'

Where could it come from ?

Regards




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is sendmail binary there and it's executable?

ls -l /usr/sbin/sendmail

Check too that user that run asterisk has permission to it
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Re: [asterisk-users] error when compiling zaptel-1.4

[Rodrigo Gonzalez]

Alyed Tzompa wrote:

The error lies here:

 >make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
 >make: *** arch/i386/boot: No such file or directory. Stop.

do you have the kernel-headers installed? (e.g. 
glibc-kernheaders-2.4-9.1.87.i386.rpm for Fedora)



Alyed


Return-Path: <[EMAIL PROTECTED]>
Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by 
maila11.webcontrolcenter.com with SMTP;

Thu, 8 Feb 2007 12:43:57 -0700
Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])
by lists.digium.com (Postfix) with ESMTP id 130B52FC8C2;
Thu, 8 Feb 2007 10:58:02 -0700 (MST)


when i compile zaptel
make linux26
make install
i got these errors:

make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C /lib/modules/2.4.27-3-386/build
SUBDIRS=/usr/src/zaptel-1.4/datamods clean
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory. Stop.
make: Entering an unknown directorymake: Leaving an unknown
directorymake[2]: *** [archclean] Error 2
make[2]: Leaving directory `/usr/src/kernel-headers-2.4.27-3-386'
make[1]: *** [clean] Error 2
make[1]: Leaving directory `/usr/src/zaptel-1.4/datamods'
make: *** [clean] Error 2


You have a debian there I think cause of the kernel name and version.

2 things

uname -r

Make sure that the running kernel is the same version 2.4.27-3-386

Second run

make

To use make linux26 you need to have kernel 2.6

You can install kernel 2.6, kernel-headers for the 2.6 version and the 
you can use make linux26




any idea
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Re: [asterisk-users] Can't get asterisk to compile chan_zap (was "NewIssue")

[Rodrigo Gonzalez]

David Ruggles wrote:

I captured the output of ./configure and found the following lines:


checking zaptel/tonezone.h usability... yes
checking zaptel/tonezone.h presence... yes
checking for zaptel/tonezone.h... yes

checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes


So it seems to be finding the /usr/include/zaptel directory and files fine.
Is there anything else I can do that might offer information that could help
track this problem down?

Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200  [EMAIL PROTECTED]


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check too that menuselect-tree has an entry for chan_zap (it's in source 
root)

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Re: [asterisk-users] Can't get asterisk to compile chan_zap (was "NewIssue")

[Rodrigo Gonzalez]

David Ruggles wrote:

I captured the output of ./configure and found the following lines:


checking zaptel/tonezone.h usability... yes
checking zaptel/tonezone.h presence... yes
checking for zaptel/tonezone.h... yes

checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes


So it seems to be finding the /usr/include/zaptel directory and files fine.
Is there anything else I can do that might offer information that could help
track this problem down?

Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200  [EMAIL PROTECTED]


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I didnt follow your original thread...

chan_zap does not appear in menuselect?

Does it exist in channels directory?
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Re: [asterisk-users] Asterisk very slow when internet down

[Rodrigo Gonzalez]

Peter Mitchell wrote:
Has anyone seen this issue with asterisk running like a dog when the 
internet is down ?  Internal calls, incoming ISDN calls etc all seem to 
be affected.  There is a local DNS server that is always available so 
I’m not sure why asterisk is so unresponsive.


 

I’ve seen this on two different systems, and on 1 of them I commented 
out my SIP providers in sip.conf and it ran ok again.


 


Thanks

Peter.


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Check that it's not doing SRV request in sip.conf
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Re: [asterisk-users] chan_sip loading delay in Asterisk 1.2.10

[Rodrigo Gonzalez]

ast guy wrote:

Hi,
I'm running Asterisk 1.2.10  on gentoo linux and facing strange kind of 
issue.

1. chan_sip.so takes about 10 secs to load up when asterisk starts.
2. When I dialout using SIP it takes 20 secs to output " -- Called SIP
[EMAIL PROTECTED]" and get ring back from B party...
Is there any config that I can check to reduce both delays?

-ag
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Maybe an issue with DNS? Do you have DNS Srv allowed? Do you have a good 
DNS Server?

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Re: [asterisk-users] Re: Switching from FreeBSD to Linux - which distro?

[Rodrigo Gonzalez]

yum can be used...

direct download from 
http://isoredirect.centos.org/centos/4/os/i386/CentOS/RPMS/


Tomislav Parčina wrote:

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
  
Debian is my fave, but for Asterisk I use CentOS. It's a free-of-cost clone of 
Red Hat Enterprise Linux, so it's very stable and reliable, and Asterisk runs 
great on it. Debian is good too. They have Asterisk packages, but they're 
generally a little bit old. Source installations work fine. Both have large, 
active developer and user communities.



Hi Carla!

Can you tell me from where do you download rpm's for Cent OS 4?



--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] asterisk manager originate command

[Rodrigo Gonzalez]

My code is using phpagi-asmanagerbut what is sent is...

Action: Originate
Channel: SIP/802
Context: from-internal
Exten: < number to dial >
Priority: 1
Callerid: 802



Michael Collins wrote:

There is no dial command, I'm sending originate action from asterisk
manager.



Oops, I didn't ask my question correctly.  You're right, it isn't a
"dial" command.  What I wanted to know was the contents of your
originate action, e.g.:

Channel=> 'zap/g0/' . $dialed_num

(From one of my Perl scripts using
POE::Component::Client::Asterisk::Manager)

Thanks,
MC
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Re: [asterisk-users] asterisk manager originate command

[Rodrigo Gonzalez]

There is no dial command, I'm sending originate action from asterisk 
manager.


Michael Collins wrote:

I want to know how to get the uniqueid or a call started from asterisk
manager using Originate command.



Are you wanting the uniqueid for the call right after it is started,
i.e., while it is still in progress?  What is in your Dial command?

-MC
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[asterisk-users] asterisk manager originate command

[Rodrigo Gonzalez]

Hi everybody,

I want to know how to get the uniqueid or a call started from asterisk 
manager using Originate command.


Best regards

Rodrigo Gonzalez
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Re: [asterisk-users] app_prepaid won't load - undefined symbol mysql_num_fields

[Rodrigo Gonzalez]

Install mysql devel package (depend on your distribution) and will work, 
it's not finding the library libmysqlclient




Mosiuoa Tsietsi wrote:

Hi again,

I downloaded the bristuff-0.3.0-PRE-1s.tar.gz archive from
http://www.junghanns.net which has a script you can run to download the
sources for asterisk (1.2.10), libpri (1.2.3) and zaptel (1.2.6).  It
also has patches for the above as well.  Another script helps build the
sources for each for you one-time.

Asterisk build properly and when I copy across my app_prepaid.c file
into */apps folder, I can successfully run 


$ make clean && make && make install

I separately have the sources for asterisk-addons-1.2.4 in which I have
successfully run 


$ make clean && make install

after I have built asterisk.

It's when I try 
$ asterisk -gc that I get the error:


[ Booting...Nov  4 22:49:49 NOTICE[1861]: cdr.c:1191 do_reload: CDR
simple logging enabled.
.Nov  4 22:49:49 WARNING[1861]: cdr_addon_mysql.c:361 my_load_module:
MySQL database sock file not specified.  Using default
Nov  4 22:49:49 WARNING[1861]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/app_prepaid.so: undefined
symbol: mysql_num_fields
Nov  4 22:49:49 WARNING[1861]: loader.c:499 load_modules: Loading module
app_prepaid.so failed!

The line in question in the source (app_prepaid.c) is :

num_fields = mysql_num_fields(result); 


where result is of type *MYSQL_RES, which is defined in my mysql.h
header.

Please help.  



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Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages

[Rodrigo Gonzalez]

If "it is not parsing the index.php" mean that you see the code in your 
browser, install php


Alok Mohapatra wrote:


Hi All,

  I have installed Asterisk 1.2.10 on Fedora 5. I have 
installed Asterisk Management Portal (AMP) for web interface.


After installing properly when opening in the webpage it is not 
parsing the index.php for the AMP. My Database is MySQL.and web server 
is Apache 2.2.


 

Please let me know is this configuration problem or this is the 
problem with Apache (Apache 2.2) .


 


Thanks and Regards

Alok Mohapatra

 




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Re: [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem

[Rodrigo Gonzalez]

User that web server is running has to have read permissions to file 
/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt


Easier option is to run apache as asterisk user

Zeeshan Zakaria wrote:
Anybody knows why ARI gives this error message when I enter extension 
number and password.
 
*Warning*: 
file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt): 
failed to open stream: Permission denied in 
*/var/www/html/recordings/modules/voicemail.module* on line *525*
It doesn't show the voicemails, although it shows that there is 1 or 2 
voicemails in the INBOX.


--
Zeeshan A Zakaria


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Re: [asterisk-users] Something is trashing /var/run

[Rodrigo Gonzalez]

mkdir /var/run/asterisk

in /etc/asterisk/asterisk.conf change where you see /var/run with 
/var/run/asterisk


Jim Lynch wrote:
For some reason, asterisk is changing ownership of all the files in 
/var/run to itself.  /var/run/* now belongs to asterisk.asterisk after 
a reboot.


I installed zaptel, wanpipe and asterisk on a fresh install of 
CentOS.  I did the same thing on Friday and had the same problem, so I 
scrubbed the disk and tried again.


Any ideas what's going on here ?

Thanks,
Jim.
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Re: [asterisk-users] Lumenvox speech recognition

[Rodrigo Gonzalez]

I've worked with it using Asterisk, and worked really fine

Michael Welter wrote:

Does anyone have experience with this product?
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