[Asterisk-Users] Re: (no subject)

2005-09-14 Thread Pablo Allietti
On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote:


ok. didnt work :( i thinks is a pbx problem. because E1 is incomming in
the pbx. and all incomming calls go to 100.  thats the problem i will
try to solve this.



 It could potentially be both. I would look at your extensions.conf first
 though. What does the extension entry for that context look like.
 
 For instance I have an entry in my extensions.conf for dialing outside
 lines (outside being from asterisk to my PBX and then onto the outside
 world from there). The entry looks like this:
 
 [to-analog]
 exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN})
 exten = _9XXX.,2,Congestion
 exten = _9XXX.,103,Hangup
 
 
 To dial a PBX extension the entry would look almost the same:
 
 [to-pbx-extension]
 exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1})
 exten = _9XXX.,2,Congestion
 exten = _9XXX.,103,Hangup
 
 Hope this helps,
 
 -Matt
 
 On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
  hi all, i have a box with a te110p and a pbx siemens... connect both
  with a e1.
  with a xten soft i can call extensions numbers in my office example 100
  102 etc. but when i truy to go outside with the 9 before the call rings
  in the first extensions (100). this is a asterisk problem? or a pbx
  problem?
 
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --
 
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
---end quoted text---

-- 

.-

Pablo Allietti
LACNIC

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] Re: (no subject)

2005-09-14 Thread Sander
This is not a siemens pbx problem you set the
pridialplan = to national and that adds a number to the outgoing call or
something just use


Pridialplan = local
prilocaldialplan = local

and it should work

I tried to open the file kds again and now it showed me your configuration
:) don't know why it did not show me before

Sander

-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Pablo Allietti
Verzonden: woensdag 14 september 2005 17:31
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] Re: (no subject)

On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote:


ok. didnt work :( i thinks is a pbx problem. because E1 is incomming in the
pbx. and all incomming calls go to 100.  thats the problem i will try to
solve this.



 It could potentially be both. I would look at your extensions.conf 
 first though. What does the extension entry for that context look like.
 
 For instance I have an entry in my extensions.conf for dialing outside 
 lines (outside being from asterisk to my PBX and then onto the outside 
 world from there). The entry looks like this:
 
 [to-analog]
 exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN}) exten = _9XXX.,2,Congestion 
 exten = _9XXX.,103,Hangup
 
 
 To dial a PBX extension the entry would look almost the same:
 
 [to-pbx-extension]
 exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1}) exten = _9XXX.,2,Congestion 
 exten = _9XXX.,103,Hangup
 
 Hope this helps,
 
 -Matt
 
 On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
  hi all, i have a box with a te110p and a pbx siemens... connect both 
  with a e1.
  with a xten soft i can call extensions numbers in my office example 
  100
  102 etc. but when i truy to go outside with the 9 before the call 
  rings in the first extensions (100). this is a asterisk problem? or 
  a pbx problem?
 
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --
 
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
---end quoted text---

-- 

.-

Pablo Allietti
LACNIC

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Re: (no subject)

2005-09-14 Thread Pablo Allietti
On Wed, Sep 14, 2005 at 07:52:26PM +0200, Sander wrote:
 This is not a siemens pbx problem you set the
 pridialplan = to national and that adds a number to the outgoing call or
 something just use
 
 
 Pridialplan = local
 prilocaldialplan = local
 
 and it should work


no uuuaaa the same problem.. ring in the extension 100. 

 
 I tried to open the file kds again and now it showed me your configuration
 :) don't know why it did not show me before
 
 Sander
 
 -Oorspronkelijk bericht-
 Van: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Namens Pablo Allietti
 Verzonden: woensdag 14 september 2005 17:31
 Aan: Asterisk Users Mailing List - Non-Commercial Discussion
 Onderwerp: [Asterisk-Users] Re: (no subject)
 
 On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote:
 
 
 ok. didnt work :( i thinks is a pbx problem. because E1 is incomming in the
 pbx. and all incomming calls go to 100.  thats the problem i will try to
 solve this.
 
 
 
  It could potentially be both. I would look at your extensions.conf 
  first though. What does the extension entry for that context look like.
  
  For instance I have an entry in my extensions.conf for dialing outside 
  lines (outside being from asterisk to my PBX and then onto the outside 
  world from there). The entry looks like this:
  
  [to-analog]
  exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN}) exten = _9XXX.,2,Congestion 
  exten = _9XXX.,103,Hangup
  
  
  To dial a PBX extension the entry would look almost the same:
  
  [to-pbx-extension]
  exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1}) exten = _9XXX.,2,Congestion 
  exten = _9XXX.,103,Hangup
  
  Hope this helps,
  
  -Matt
  
  On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
   hi all, i have a box with a te110p and a pbx siemens... connect both 
   with a e1.
   with a xten soft i can call extensions numbers in my office example 
   100
   102 etc. but when i truy to go outside with the 9 before the call 
   rings in the first extensions (100). this is a asterisk problem? or 
   a pbx problem?
  
  ___
  --Bandwidth and Colocation sponsored by Easynews.com --
  
  Asterisk-Users mailing list
  Asterisk-Users@lists.digium.com
  http://lists.digium.com/mailman/listinfo/asterisk-users
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 ---end quoted text---
 
 -- 
 
 .-
 
 Pablo Allietti
 LACNIC
 
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --
 
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --
 
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
---end quoted text---

-- 

.-

Pablo Allietti
LACNIC

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] Re: (no subject)

2005-09-14 Thread Sander
 
Ok it was a problem with my provider it could not see the right numbers
comming in :)
You can start maintenence in the manager e tool from siemens and start a
trace or start call monitoring on extension 100 then you can see the number
the asterisk has given to the pbx to dial.  


-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Pablo Allietti
Verzonden: woensdag 14 september 2005 21:17
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] Re: (no subject)

On Wed, Sep 14, 2005 at 07:52:26PM +0200, Sander wrote:
 This is not a siemens pbx problem you set the pridialplan = to 
 national and that adds a number to the outgoing call or something just 
 use
 
 
 Pridialplan = local
 prilocaldialplan = local
 
 and it should work


no uuuaaa the same problem.. ring in the extension 100. 

 
 I tried to open the file kds again and now it showed me your 
 configuration
 :) don't know why it did not show me before
 
 Sander
 
 -Oorspronkelijk bericht-
 Van: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Namens Pablo Allietti
 Verzonden: woensdag 14 september 2005 17:31
 Aan: Asterisk Users Mailing List - Non-Commercial Discussion
 Onderwerp: [Asterisk-Users] Re: (no subject)
 
 On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote:
 
 
 ok. didnt work :( i thinks is a pbx problem. because E1 is incomming 
 in the pbx. and all incomming calls go to 100.  thats the problem i 
 will try to solve this.
 
 
 
  It could potentially be both. I would look at your extensions.conf 
  first though. What does the extension entry for that context look like.
  
  For instance I have an entry in my extensions.conf for dialing 
  outside lines (outside being from asterisk to my PBX and then onto 
  the outside world from there). The entry looks like this:
  
  [to-analog]
  exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN}) exten = _9XXX.,2,Congestion 
  exten = _9XXX.,103,Hangup
  
  
  To dial a PBX extension the entry would look almost the same:
  
  [to-pbx-extension]
  exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1}) exten = 
  _9XXX.,2,Congestion exten = _9XXX.,103,Hangup
  
  Hope this helps,
  
  -Matt
  
  On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
   hi all, i have a box with a te110p and a pbx siemens... connect 
   both with a e1.
   with a xten soft i can call extensions numbers in my office 
   example 100
   102 etc. but when i truy to go outside with the 9 before the call 
   rings in the first extensions (100). this is a asterisk problem? 
   or a pbx problem?
  
  ___
  --Bandwidth and Colocation sponsored by Easynews.com --
  
  Asterisk-Users mailing list
  Asterisk-Users@lists.digium.com
  http://lists.digium.com/mailman/listinfo/asterisk-users
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 ---end quoted text---
 
 --
 
 .-
 
 Pablo Allietti
 LACNIC
 
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --
 
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --
 
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
---end quoted text---

-- 

.-

Pablo Allietti
LACNIC

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] RE: No subject by Steve M

2004-09-12 Thread Thomas Hutton

Just responding in case this may be of help to somebody with firewalling
issues.  Not sure if this is off on a tangent to the original
question...


Here are three different forms of common firewall scripts and ways of
getting SIP to work behind them.  The third one has some additional
stuff beyond just SIP although I can't remember why I wrote it that way.

I've been having no fun using sip phones that try to figure things out
with third party STUN servers.  It seems better to use a good linux
firewall like it was intended.  

***

Redhat 9 scripts - basic firewall rules for SIP forwarding:
file: /etc/rc.d/init.d/firewall

In the Services section, add this:
SIP=your.internal.ip.here # VIOP SIP 

Add the following code amongst the service scripts toward the bottom:

##
#SIP #
##
   
   
function SIP_WAN {
   $IPT -A INPUT -p udp -i $WANIFACE --dport 5060 -j ACCEPT
   $IPT -A INPUT -p udp -i $WANIFACE --dport 5004 -j ACCEPT
}
   
   
function SIP_PORT_FORWARDING {
   $IPT -A PREROUTING -t nat -i $WANIFACE -p udp --dport 5060 -j DNAT
--to $SIP:5060
   $IPT -A FORWARD -i $WANIFACE -p udp --dport 5060 -j ACCEPT
   $IPT -A PREROUTING -t nat -i $WANIFACE -p udp --dport 5004 -j DNAT
--to $SIP:5004
   $IPT -A FORWARD -i $WANIFACE -p udp --dport 5004 -j ACCEPT
}  

if [ $SIP = ON ]; then
   SIP_WAN
else
   if [ $SIP != OFF  ]; then
  SIP_PORT_FORWARDING
   fi
fi

Note: the two lines above beginning with two dashes (--) wrapped, they
should be at the end of the lines above them.

***

Basic rule for SuSEfirewall2:
file: /etc/sysconfig/SuSEfirewall2
In the section for FW_FORWARD_MASQ=
insert the following two lines:
0/0,internal.sip.ip.address,udp,5060,external.ip.address.here
0/0,internal.sip.ip.address,udp,5004,external.ip.address.here

Note: quotation marks are used in this section- although only at the
beginning and the end... it's a goofy syntax for writing a config file-
so if it doesn't work, and these are the only two ports you're
forwarding, it should look like this: 

FW_FORWARD_MASQ=0/0,internal.sip.ip.address,udp,5060,external.ip.address.heree
0/0,internal.sip.ip.address,udp,5004,external.ip.address.here

**

Ruleset for an old reliable IPChains firewall:  
file: /etc/rc.d/init.d/firewall
This actually opens up a few more holes for some outbound streams. 
Can't remember exactly why I did it this way but it works good.
# VIOP - asterisk
# vars
$EXT_IP=your.external.ip.here
$ASTERISK_IP=your.asterisk.server.ip
#
#chains
ipmasqadm portfw -a -P udp -L $EXT_IP 5060 -R $ASTERISK_IP 5060
ipchains -A portfw -s 0/0 1024: -d $EXT_IP 5060 -p 17 -j ACCEPT
ipmasqadm portfw -a -P udp -L $EXT_IP 4569 -R $ASTERISK_IP 4569
ipchains -A portfw -s 0/0 1024: -d $EXT_IP 4569 -p 17 -j ACCEPT
ipmasqadm portfw -a -P udp -L $EXT_IP 5036 -R $ASTERISK_IP 5036
ipchains -A portfw -s 0/0 1024: -d $EXT_IP 5036 -p 17 -j ACCEPT
# loop for a bunch of ports for streams
 port2=10001
 while [ $port2 -lt 10699 ]
 do
 ipmasqadm portfw -a -P udp -L $EXT_IP $port2 -R $ASTERISK_IP  $port2
 ipchains -A portfw -s 0/0 1024: -d $EXT_IP $port2 -p 17 -j ACCEPT
 ipchains -A portfw -s 0/0 $port2 -d $EXT_IP $port2 -p 17 -j ACCEPT
 port2=$((port2+1))
 done
#

*

I've also been able to run sip traffic over a vpn.  My ISP here seems
like it's doing some weird stuff with delaying packets on certain ports
- so I stuff a lot of stuff through a tunnel.  Problem is that the
encryption slows it down- which I fixed by running a pptpd daemon
without encryption.

This will be out of thread, excuse me, but my mozilla is broken and
won't take mailto registry fixes.  Will need to use a different client
in future posts.

TJH

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users