Re: [Asterisk-Users] Cisco 2600 and ASTERISK

2003-09-26 Thread Sean Figgins
On Fri, 26 Sep 2003, Bartosz Jozwiak wrote:

 I have pointed it to Asterisk for sure not to local cisco ethernet.
 I think there is something wrong with the router.

In the exerpt from the config you posted (below), the destination in the
dial-peer is the same as the address on the enternet interface.  Perhaps
this is not your actual config?

In any event, if this is correct, it should work.

   dial-peer voice 1000 voip
max-conn 4
destination-pattern 
req-qos guaranteed-delay
codec g711ulaw
ip precedence 5
no vad
session target ipv4:66.178.37.169
   !
   !
   interface Ethernet0/0
ip address 66.178.37.169 255.255.254.0
no ip directed-broadcast
half-duplex

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Cisco 2600 and ASTERISK

2003-09-25 Thread Sean Figgins
Looks like your sip destination is the same IP as the IP on the ethernet
interface.  I am pretty sure that this IP needs to be the sip server, not
the router.

-Sean

On Wed, 24 Sep 2003, Bartosz Jozwiak wrote:

 This is my configuration of my cisco router and still it does not want to
 work :(


 Current configuration:
 !
 version 12.0
 service timestamps debug uptime
 service timestamps log uptime
 service password-encryption
 !
 hostname asterisk
 !
 aaa new-model
 aaa authentication login default local
 enable secret 5 $1$bJzJ$bjJ.hc0TbiopbjjMUnyhg/
 !
 username admin password 7 07002C494908
 !
 !
 !
 !
 ip subnet-zero
 ip name-server 66.178.37.211
 !
 !
 !
 !
 voice-port 1/0/0
 !
 voice-port 1/0/1
 !
 voice-port 1/1/0
 !
 voice-port 1/1/1
  connection plar 
 !
 !
 dial-peer voice 1000 voip
  max-conn 4
  destination-pattern 
  req-qos guaranteed-delay
  codec g711ulaw
  ip precedence 5
  no vad
  session target ipv4:66.178.37.169
 !
 !
 interface Ethernet0/0
  ip address 66.178.37.169 255.255.254.0
  no ip directed-broadcast
  half-duplex
 !
 interface Serial0/0
  no ip address
  no ip directed-broadcast
  shutdown
 !
 interface Ethernet0/1
  no ip address
  no ip directed-broadcast
  shutdown
  half-duplex
 !
 ip classless
 ip route 0.0.0.0 0.0.0.0 66.178.36.4
 no ip http server
 !
 !
 line con 0
  transport input none
 line aux 0
 line vty 0 4
 !
 no scheduler allocate
 end


 - Original Message -
 From: Brian West [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Wednesday, September 24, 2003 2:25 PM
 Subject: RE: [Asterisk-Users] Cisco 2600 and ASTERISK


 
 http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080093f62.shtml
 
  That covers the thridparty h323 stuff with *
 
  bkw
 
  On Wed, 24 Sep 2003, Sean Figgins wrote:
 
  
   That is about what I have been seing for help.  Has anyone any clue what
   to di with a 2600 that has a T1 adapter on a high-density high-density
   voice port adapter?
  
   BTW...  Because I am lazy, what does plar do?
  
   -Sean
  
   On Wed, 24 Sep 2003, Brian West wrote:
  
This is simple to do..
   
voice-port 1/0/0
 connection plar 
!
voice-port 1/0/1
 connection plar 
!
dial-peer voice 1000 voip
 max-conn 4
 destination-pattern 
 req-qos guaranteed-delay
 codec g711ulaw
 ip precedence 5
 no vad
 session target ipv4:x.x.x.x
!
   
in h323.conf set the context=blah
   
[blah]
   
exten = ,1,Goto(s,1)
   
   
Done... its really that simple.  I have this working with a 2600 and a
1750.
   
bkw
   
On Wed, 24 Sep 2003, Joseph Finley wrote:
   
 I too would like to see it.  I've tried many times with the help of
 a few
 and never got it to work.  It always results in a fast busy.

 Joe


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
 Jozwiak
 Sent: Wednesday, September 24, 2003 9:46 AM
 To: ASTERISK USERS
 Subject: [Asterisk-Users] Cisco 2600 and ASTERISK


 Hello,

 Could somebody tell me if I can connect CISCO 2600 router with
 support of
 H.323 to Asterisk ?
 If it is possible could somebody tell me how to do it.
 I would like to document it and put on some website so everyone can
 see it.

 Regards,

 -- bart


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
   
  
   ___
   Asterisk-Users mailing list
   [EMAIL PROTECTED]
   http://lists.digium.com/mailman/listinfo/asterisk-users
  
  ___
  Asterisk-Users mailing list
  [EMAIL PROTECTED]
  http://lists.digium.com/mailman/listinfo/asterisk-users
 

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] Cisco 2600 and ASTERISK

2003-09-24 Thread Sean Figgins

That is about what I have been seing for help.  Has anyone any clue what
to di with a 2600 that has a T1 adapter on a high-density high-density
voice port adapter?

BTW...  Because I am lazy, what does plar do?

-Sean

On Wed, 24 Sep 2003, Brian West wrote:

 This is simple to do..

 voice-port 1/0/0
  connection plar 
 !
 voice-port 1/0/1
  connection plar 
 !
 dial-peer voice 1000 voip
  max-conn 4
  destination-pattern 
  req-qos guaranteed-delay
  codec g711ulaw
  ip precedence 5
  no vad
  session target ipv4:x.x.x.x
 !

 in h323.conf set the context=blah

 [blah]

 exten = ,1,Goto(s,1)


 Done... its really that simple.  I have this working with a 2600 and a
 1750.

 bkw

 On Wed, 24 Sep 2003, Joseph Finley wrote:

  I too would like to see it.  I've tried many times with the help of a few
  and never got it to work.  It always results in a fast busy.
 
  Joe
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak
  Sent: Wednesday, September 24, 2003 9:46 AM
  To: ASTERISK USERS
  Subject: [Asterisk-Users] Cisco 2600 and ASTERISK
 
 
  Hello,
 
  Could somebody tell me if I can connect CISCO 2600 router with support of
  H.323 to Asterisk ?
  If it is possible could somebody tell me how to do it.
  I would like to document it and put on some website so everyone can see it.
 
  Regards,
 
  -- bart
 
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Recommended OS

2003-09-23 Thread Sean Figgins
On Tue, 23 Sep 2003, Michael A. Miller wrote:

 Is there a recommended OS that Asterisk should be used with? I have been
 trying to get Asterisk running on Red Hat 9.0 with little success.

I have * running on Redhat 9.0.  It seems to work fine with SIP and my
Cisco 7960 phones running 5.x firmware.  I have been having a very
difficult time getting h.323 to work so I can connect to a remote Cisco
Callmanager, though.  Specifically, I don't seem to get h.323 to compile
correctly.

I woudl prefer to use FreeBSD, and there was some talk a couple weeks ago
about getting * ported for use there.  I haven't seen an update for a
while, though.

-Sean



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet

2003-09-10 Thread Sean Figgins
On Wed, 10 Sep 2003, Jim Mercer wrote:

 i've not done an autoconf before, and i suspect it will require not a small
 amount of tweaking.

 i suspect the BSD patches will head in the direction of a number of those
 tweaks.

Some of the patches that get applied in the normal /usr/ports tree under
FreeBSD is only in regards to location of files and directories.  This
usually solves a great number of consistancy problems.

 while we could all wait around for someone to magically autoconf the code,
 in practical terms, i'm content to release the tweaks i'm making for
 freebsd/netbsd/openbsd/OSX (when i get time to clean it up, next week).

Are you going to have your system running H.323 as well?  My current
challange is getting H.323 up and running so I can talk to a local Cisco
voice gateway, and a remote Cisco call manager.  Getting everything just
right in Linux, which I am not remotely familiar with any longer, is
causing me to pull my hair out.

I'll be very interested in your tweaks.

-Sean

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet

2003-09-09 Thread Sean Figgins
On Mon, 8 Sep 2003, Jim Mercer wrote:

  Can we bribe you? :)

 sure, pay my rent for 3 months and give me a 50 plasma TV to play in the
 background.

Is that all?  That sounds rather cheap, compared to the things direction
that I'd have to go if I wanted to stick to the cisci CM route, with
licenses for every endpoint that I want to connect.

Realistically...  I just can not comprehend how to get stuff to work
correctly with Linux.  I used to be a Linux nut years ago, but once I
found FreeBSD with it's ports collection, I wondered why anyone ever
bothered with Linux and it's completely messed up software install
requirements.

Right now, under RedHat 9.0, I have * running, but no hardware, and I
can't figure out how to get h.323 operational so I can talk to my cisco
gateway with the PRI interface...  I'm only guessing that FreeBSD would be
much easier for non-programmers like myself.

-Sean

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] chan_h323.c

2003-08-20 Thread Sean Figgins
On Mon, 18 Aug 2003, Mark Spencer wrote:

 It's up one directly.  It just moved.

 Run make in h323 then do make install on asterisk again.

 On Mon, 18 Aug 2003, John Fortman wrote:

  What happened to chan_h323.c in the asterisk cvs?  I got ast_h323.cpp,
  ast_h323.h and chan_h323.h but no chan_h323.c.  Hence chan_h323.so was
  not created so no h323 support in asterisk.
 
  Just wondering when to expect it again because I was stupid and didn't
  make a backup of the asterisk code before wiping the directory for a
  rebuild.

Has anyone gotten H.323 channel complied on Redhat 9.0?  Every time I try,
I get a ton of errors from ptlib.  I'm about ready to punt this sucker out
the door.  I really like what I have seen out of asterisk so far...

Example of errors:

In file included from /usr/include/ptlib/contain.h:218,
 from /usr/include/ptlib.h:137,
 from ast_h323.h:29,
 from ast_h323.cpp:27:
/usr/include/ptlib/object.h:585: parse error before `(' token
/usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1201: parse error before `(' token
/usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
/usr/include/ptlib/object.h:1201: conflicts with previous declaration `int
   PObject::BOOL'
/usr/include/ptlib/object.h:1214: parse error before `(' token
/usr/include/ptlib/object.h:1265: syntax error before `operator'
/usr/include/ptlib/object.h:1214: duplicate member `PObject::BOOL'
/usr/include/ptlib/object.h:1274: ISO C++ forbids defining types within
return
   type


Any help would be appriciated, even if it's a recommendation to another
flavor of linux.

Thanks
Sean

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] chan_h323.c

2003-08-20 Thread Sean Figgins
Great!  Thanks for the recommendation.  I'll beat on Redhat a little bit
longer, then try to load slackware and give that a whirl.

Thanks again.
Sean

On Wed, 20 Aug 2003, John Fortman wrote:

 I have been using Slackware 9 with the sources for ptlib_unix, pwlib_unix,
 openh323, asterisk, zaptel and libpri in /root/src

 1) load all 6 packages (+ ohphone if you need an h323 client) from cvs to
 /root/src
 2) /root/src/pwlib: configure, make, make install, ldconfig (not all that
 sure why, but Slackware requires ldconfig to be run)
 3) /root/src/openh323: configure, make, make install, ldconfig
 4) /root/src/zaptel: make, make install (reload and reconfigure your zaptel
 card)
 5) /root/src/libpri: make, make install (I don't have a PRI card so I don't
 do anything here)
 6) /root/src/asterisk/channels/h323:
 - edit Makfile
 - set PWLIBDIR = $(HOME)/src/pwlib
 - set OPENH323DIR = $(HOME)/src/openh323
 - make, make install (installs openh323.a) (make samples if you do not
 have h323.conf in /etc/asterisk when done)
 7) /root/src/asterisk: make, make install, make samples
 8) asterisk -vvvc
 - the last section should load chan_h323

 I haven't had any problems compiling this from CVS for almost a month on at
 least three different systems with some version of Slackware.  I have had
 problems with other things like transferring calls but that's a different
 issue.

 John.

 - Original Message -
 From: Sean Figgins [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Wednesday, August 20, 2003 12:22 PM
 Subject: Re: [Asterisk-Users] chan_h323.c


  On Mon, 18 Aug 2003, Mark Spencer wrote:
 
   It's up one directly.  It just moved.
  
   Run make in h323 then do make install on asterisk again.
  
   On Mon, 18 Aug 2003, John Fortman wrote:
  
What happened to chan_h323.c in the asterisk cvs?  I got ast_h323.cpp,
ast_h323.h and chan_h323.h but no chan_h323.c.  Hence chan_h323.so was
not created so no h323 support in asterisk.
   
Just wondering when to expect it again because I was stupid and didn't
make a backup of the asterisk code before wiping the directory for a
rebuild.
 
  Has anyone gotten H.323 channel complied on Redhat 9.0?  Every time I try,
  I get a ton of errors from ptlib.  I'm about ready to punt this sucker out
  the door.  I really like what I have seen out of asterisk so far...
 
  Example of errors:
 
  In file included from /usr/include/ptlib/contain.h:218,
   from /usr/include/ptlib.h:137,
   from ast_h323.h:29,
   from ast_h323.cpp:27:
  /usr/include/ptlib/object.h:585: parse error before `(' token
  /usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
  /usr/include/ptlib/object.h:1201: parse error before `(' token
  /usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
  /usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
  /usr/include/ptlib/object.h:1201: conflicts with previous declaration `int
 PObject::BOOL'
  /usr/include/ptlib/object.h:1214: parse error before `(' token
  /usr/include/ptlib/object.h:1265: syntax error before `operator'
  /usr/include/ptlib/object.h:1214: duplicate member `PObject::BOOL'
  /usr/include/ptlib/object.h:1274: ISO C++ forbids defining types within
  return
 type
 
 
  Any help would be appriciated, even if it's a recommendation to another
  flavor of linux.
 
  Thanks
  Sean
 
  ___
  Asterisk-Users mailing list
  [EMAIL PROTECTED]
  http://lists.digium.com/mailman/listinfo/asterisk-users

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] h323 compile error

2003-08-14 Thread Sean Figgins
Excuse me for jumping in here.  I just subscribed, so I might be asking
something that has already been answered.  I tried searching the archive,
but didn't see an answer.

I was trying to compile under Redhat 9.  I think I got everything
installed that I need to compile, but I get the following errors:

In file included from /usr/include/ptlib/contain.h:218,
 from /usr/include/ptlib.h:137,
 from ast_h323.h:29,
 from ast_h323.cpp:27:
/usr/include/ptlib/object.h:585: parse error before `(' token
/usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1201: parse error before `(' token
/usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
/usr/include/ptlib/object.h:1201: conflicts with previous declaration `int
   PObject::BOOL'


I get 10 more pages of errors like this.  Has anyone encountered this
problem?  Any thought what I am doing wrong?

Thanks
Sean

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Voicemail2 - auto fill the dialing extension?

2003-08-12 Thread Sean Figgins
On Fri, 8 Aug 2003, Adams, Gavin wrote:

 Also, we decided to go with actual extension numbers on the phones
 instead of usernames per extension. On the Cisco phones, is there a way
 to change the name/number on the top line (white text on black) to the
 user's name, while having the extension number next to each presentation
 (line1, line2, etc)?

I can't answer the voicemail question, but this one I can.  The config
file for the phone can include the following line:

phone_label: User's Name ext. 1234

This will cause the text in quotes to appear in the black bar.

The line labels are a little bit of a problem.  The Cisco phones actually
try to authenticate with the text that is in there.  The cisco config
file:

line1_name: Line 1

This displays Line 1 on the phone, but also tried to login to the proxy
as Line [EMAIL PROTECTED].  Using other SIP proxies, you can do this with the auth
name/password, but with Asterisk...  It jsut doesn't seem to work.

There may be a work-around for this, I've only been looking at this for a
week.

-Sean

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users