Re: [Asterisk-Users] more dids added to goiax.com

2005-10-20 Thread Steve Daniels
If he did communicate with you, how could you then reply.. Just playing 
devils advocate here to what seems an obvious caveat in the thinking process 
going on there :-)


Steve :-)


- Original Message - 
From: John Novack

To: Steve Totaro
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, October 19, 2005 3:18 PM
Subject: Re: [Asterisk-Users] more dids added to goiax.com




Steve Totaro wrote:


I made two attempts this morning to send some comments off list, and both 
got returned due to some sort of spam filter, so would hope that any future 
controls will not suffer from that inability to communicate.



Maybe if you posted what the returned email said then he could remove or 
alter that filter.  both got returned due to some sort of spam filter, 
doesn't help anyone.



I saw no reason to clog up this list with  details that aren't Asterisk 
related.

If he wants further information he can communicate directly with me.

JN



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Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk

2005-10-18 Thread Steve Daniels


- Original Message - 
From: Tzafrir Cohen [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Monday, October 17, 2005 7:18 PM
Subject: Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk



On Mon, Oct 17, 2005 at 06:56:39PM +0100, Steve Daniels wrote:

Try a a good old
netstat -a | grep 5038


netstat -lntup | grep 5038

-l is generally preferable than -a: only gives listening ports. -t: tcp
sockets only. -p: also pid and name of the process that has the socket.


-a show's all sockets regardless of state
though I admit including -p would check that it was actually * listening on 
that port,
and make sure that no other process is preventing * from binding to that 
port.





That will tell you if * is listening and what it's listening on.
Then if it show's * is listening, it must be a permit =, or a firewall
issue.


Though most firewalls allow everything on the interface lo. My guess is
that either asterisk is not running or it has not been *restarted* since
the configuration change. Reloading is not enough to change the
IP address(es) Asterisk listens on or make it start listen at all
(right?)

Make sure you restart the process, it sounds as if you've probably restarted 
it a fair few times though..



--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk

2005-10-17 Thread Steve Daniels

Try a a good old
netstat -a | grep 5038
That will tell you if * is listening and what it's listening on.
Then if it show's * is listening, it must be a permit =, or a firewall 
issue.


HTH

Steve
- Original Message - 
From: Chuck Bunn [EMAIL PROTECTED]

To: Asterisk - Users asterisk-users@lists.digium.com
Sent: Monday, October 17, 2005 5:22 AM
Subject: [Asterisk-Users] Cannot telnet to port 5038 on asterisk



Hi,

I cannot do the following:

telnet 127.0.0.1 5038

I get connection refused and this is preventing AMP from installing. I had 
this working when I was using FC3 but I had to upgrade to FC4 for another 
application. So I am running PHP5, MYSQL 4 with FC4 and asterisk is 
running (I had this problem before with FC3 and it turned out asterisk was 
not running) I am using 1.2.0 beta1 Asterisk code.


Thanks
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Re: [Asterisk-Users] Wanting to Make a PocketPC have a secureConnection to asterisk server

2005-10-13 Thread Steve Daniels



VPN?
IAX and an SSH Tunnel?

  - Original Message - 
  From: 
  Kellner, 
  Peter 
  To: asterisk-users@lists.digium.com 
  
  Sent: Thursday, October 13, 2005 5:18 
  AM
  Subject: [Asterisk-Users] Wanting to Make 
  a PocketPC have a secureConnection to asterisk server
  
  
  Does anyone know of a good 
  solution to create a secure (encrypted) connection from a pocketpc (IPAQ 6515 
  in my case) to an asterisk server?
  
  Thanks
  
  Peter 
  Kellner
  http://PeterKellner.net
  
  
  

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Re: [Asterisk-Users] New Application: Broadcast

2005-10-13 Thread Steve Daniels

What excatly does it do?
What messages does it send out?
And what software needs to be configured to listen for these messages?

Answer these questions and maybe more people will download the source :-)

Steve
(Not being an arse just reckon a better description is needed)
- Original Message - 
From: Begumisa Gerald M [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Thursday, October 13, 2005 3:08 AM
Subject: [Asterisk-Users] New Application: Broadcast



Hello,

I've released an Asterisk application under the terms of the GNU GPL.  You
may find it here:

http://psg.com/~begg/projects/

A short exerpt from the README:

--
Broadcast is an Asterisk (http://www.asterisk.org) application which you
may use to send a generic message over TCP/IP to any number of computers
running software configured to listen for these types of messages. Being
written in C, Broadcast will be dynamically loaded onto the Asterisk
program on startup, making it a highly reliable and scalable option when
compared with other solutions based on the Asterisk Gateway Interface
(AGI) system...
--

Hope someone finds it useful!

Cheers,
Gerald.

PS:
Sorry for the cross posts!
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Re: [Asterisk-Users] AGI and set_callerid for number and name

2005-10-13 Thread Steve Daniels

What version are you using?
Try SetCIDName(Fred)
Check voip-info's wiki

HTH

Steve

- Original Message - 
From: Serge Lhermitte [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Wednesday, October 12, 2005 5:57 PM
Subject: [Asterisk-Users] AGI and set_callerid for number and name



Hi,


I've been trying to use the set_callerid function in the AGI. It sets
the CallerIDname properly but I can't figure out how to set the
CallerIDnumber. 


Is it at at possible ?

Cheers.
SL



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Re: [Asterisk-Users] Maximum retries exceeded on call.

2005-10-13 Thread Steve Daniels

Using SIP? IAX?

One way sound is usually a SIP and nat/firewall problem, make sure ports are 
forwarded.


Steve
- Original Message - 
From: Peter Ankerstål [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Wednesday, October 12, 2005 10:39 PM
Subject: [Asterisk-Users] Maximum retries exceeded on call.


I have set up a asterisk-server behind NAT and peers to another asterisk
and uses that one for outgoing calls. I have som clients on my asterisk
and they could register to it well over internet. Not a problem. But when
they try to call me the asterisk-server tells me this:

Oct 12 23:21:38 WARNING[23360]: chan_sip.c:695 retrans_pkt: Maximum retries 
exceeded on call [EMAIL PROTECTED] for seqno 
32458501 (Non-critical Response)


Configs can be found at http://www.pulia.nu/~peter/asterisk/

When they call me they can hear me but I get no sound. Weird.
Any Ideas?



--
MVH
Peter Ankerstål.
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Re: [Asterisk-Users] enable mysql in asterisk

2005-10-11 Thread Steve Daniels

http://voip-info.org/wiki/ is your friend.
More specifically: 
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime


HTH

Steve

P.S.
Google should be your best friend, always ask him questions before the 
mailing list ;-)


- Original Message - 
From: julien bos

To: asterisk-users@lists.digium.com
Sent: Monday, October 10, 2005 10:55 PM
Subject: [Asterisk-Users] enable mysql in asterisk


hi all expert,

I am testing asterisk like small sip server, i installed asterisk in debian.
It runs very well. I can use softphone to register, but each time i have
modify the sip.conf, i find that it's not good way.

So if i understand, avec asterisk version 1.2 i can use mysql to stock
the information of the sip account. So can you show me how can i
do that? Can you give me the link to document? Thank you so much.

Julien



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Re: [Asterisk-Users] Skye gateway?

2005-09-25 Thread Steve Daniels
On Sun, 2005-09-25 at 11:58 -0300, Antonio José dos Santos Brandão
wrote:
 I`m trying a skype usb gateway pluged to a asterisk FXO port.
 It's working well, but have to leave a winxp machine online.
 http://www.skyvoice.com.br/
 
 But, http://www.voip-info.org/wiki-Skype+Gateways looks better,
 integrating skype to sip without using a fxo port to integrate
 asterisk.

From the wiki: PSGw Info http://www.rsdevs.com/psgw.shtml seems to
integrate skype and asterisk using a windows machine, however looking at
it's routing table setup I'm a bit confused. It looks to me as though
when you connect to the sip side of it, you always get connected to a
certain user http://www.rsdevs.com/psgw_sip.shtml scroll down to the
image at the bottom. I'm not sure exactly how useful this would be..

Can anyone help me see the light?

Steve

 
 --
 Antonio José dos Santos Brandão
 Virgos Tecnologia da Informação
 www.virgos.com.br - São Carlos,SP
 


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