[Asterisk-Users] CallManager 3.1 (2c) and Asterisk Integration

2005-01-18 Thread Andres Junge
Hello
I want to provide a CCM (versiomn 3.1, no SIP) with VoiceMail 
capabilities using asterisk as a voicemail server. Have anyone tried to 
do this. How do I connect this? As an H323 trunk?

Thanks in advance
Andrés
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[Asterisk-Users] E100P not starting?

2004-12-02 Thread Andres Junge
Hello
I have installed my new E100P card in the same machin with two TDM04B cards.
I have followthe instructions on 
http://www.digium.com/index.php?menu=configuration#T_E100P_PRI

This is my zapata.conf
span=1,0,0,cas,ami
fxsls=1-15,17-31
dchan=16
fxsks=32-39
loadzone = us
defaultzone=us
I do *modprobe zaptel wct1xxp wcfxs
ans seems to work ok.
*
But when i do
ztcfg -vvv
i got
Zaptel Configuration
==
SPAN 1: CAS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Channel 01: FXS Loopstart (Default) (Slaves: 01)
Channel 02: FXS Loopstart (Default) (Slaves: 02)
Channel 03: FXS Loopstart (Default) (Slaves: 03)
Channel 04: FXS Loopstart (Default) (Slaves: 04)
Channel 05: FXS Loopstart (Default) (Slaves: 05)
Channel 06: FXS Loopstart (Default) (Slaves: 06)
Channel 07: FXS Loopstart (Default) (Slaves: 07)
Channel 08: FXS Loopstart (Default) (Slaves: 08)
Channel 09: FXS Loopstart (Default) (Slaves: 09)
Channel 10: FXS Loopstart (Default) (Slaves: 10)
Channel 11: FXS Loopstart (Default) (Slaves: 11)
Channel 12: FXS Loopstart (Default) (Slaves: 12)
Channel 13: FXS Loopstart (Default) (Slaves: 13)
Channel 14: FXS Loopstart (Default) (Slaves: 14)
Channel 15: FXS Loopstart (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)
Channel 17: FXS Loopstart (Default) (Slaves: 17)
Channel 18: FXS Loopstart (Default) (Slaves: 18)
Channel 19: FXS Loopstart (Default) (Slaves: 19)
Channel 20: FXS Loopstart (Default) (Slaves: 20)
Channel 21: FXS Loopstart (Default) (Slaves: 21)
Channel 22: FXS Loopstart (Default) (Slaves: 22)
Channel 23: FXS Loopstart (Default) (Slaves: 23)
Channel 24: FXS Loopstart (Default) (Slaves: 24)
Channel 25: FXS Loopstart (Default) (Slaves: 25)
Channel 26: FXS Loopstart (Default) (Slaves: 26)
Channel 27: FXS Loopstart (Default) (Slaves: 27)
Channel 28: FXS Loopstart (Default) (Slaves: 28)
Channel 29: FXS Loopstart (Default) (Slaves: 29)
Channel 30: FXS Loopstart (Default) (Slaves: 30)
Channel 31: FXS Loopstart (Default) (Slaves: 31)
Channel 32: FXS Kewlstart (Default) (Slaves: 32)
Channel 33: FXS Kewlstart (Default) (Slaves: 33)
Channel 34: FXS Kewlstart (Default) (Slaves: 34)
Channel 35: FXS Kewlstart (Default) (Slaves: 35)
Channel 36: FXS Kewlstart (Default) (Slaves: 36)
Channel 37: FXS Kewlstart (Default) (Slaves: 37)
Channel 38: FXS Kewlstart (Default) (Slaves: 38)
Channel 39: FXS Kewlstart (Default) (Slaves: 39)
39 channels configured.
ZT_SPANCONFIG failed on span 1: Invalid argument (22)
what i'm doing wrong?
thanxs in advance
andres.
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Re: [Asterisk-Users] Asterisk not startin anymore.

2004-12-01 Thread Andres Junge
Robert Barnes wrote:
On Wed, 1 Dec 2004 20:13:16 +1000, Robert Barnes
[EMAIL PROTECTED] wrote:
 

This has happenned to me now too - so I doubt that your hardware is faulty...
   

Oops - wcfxs was renamed to wctdm some time ago... Working again now.
RAB
 

I still use wcfxs (comes with my debian/sarge distro). Pull out the 
module now works ok.

Thanxs
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[Asterisk-Users] Asterisk not startin anymore.

2004-11-28 Thread Andres Junge
Hello.
I have this problem. In my asterisk box, I was running debian woody with 
asterisk package from backports.org. Last friday I upgraded from debian 
to sarge and change from kernel 2.4.18-1-686 to kernel 2.6.8-1-686, 
rebuild zaptel kernel module and also upgrade to asterisk 1.0.2.  But 
now asterisk won't start.  Here is more info

#asterisk -
(last lines)
[chan_zap.so] = (Zapata Telephony w/PRI)
 == Parsing '/etc/asterisk/zapata.conf': Found
Nov 28 14:00:30 WARNING[1077059712]: chan_zap.c:765 zt_open: Unable to 
specify channel 1: No such device
Nov 28 14:00:30 ERROR[1077059712]: chan_zap.c:6195 mkintf: Unable to 
open channel 1: No such device
here = 0, tmp-channel = 1, channel = 1
Nov 28 14:00:30 ERROR[1077059712]: chan_zap.c:9130 setup_zap: Unable to 
register channel '1'
Nov 28 14:00:30 WARNING[1077059712]: loader.c:334 ast_load_resource: 
chan_zap.so: load_module failed, returning -1
 == Unregistered channel type 'Tor'
 == Unregistered channel type 'Zap'
Nov 28 14:00:30 WARNING[1077059712]: loader.c:429 load_modules: Loading 
module chan_zap.so failed!

When I modprobe wcfxs I got this:
Nov 28 14:01:47 voiplab kernel: Freshmaker version: 71
Nov 28 14:01:47 voiplab kernel: Freshmaker passed register test
Nov 28 14:01:50 voiplab kernel: Timeout waiting for calibration of module 0
Nov 28 14:01:52 voiplab kernel: Timeout waiting for calibration of module 0
Nov 28 14:01:52 voiplab kernel: Proslic Failed on Second Attempt to Auto 
Calibrate
Nov 28 14:01:53 voiplab kernel: Proslic Failed on Second Attempt to 
Calibrate Manually. (Try -DNO_CALIBRATION in Makefile)
Nov 28 14:01:53 voiplab kernel: Module 0: FAILED FXS (FCC)
Nov 28 14:01:54 voiplab kernel: Module 1: Installed -- AUTO FXS/DPO
Nov 28 14:01:54 voiplab kernel: Module 2: Installed -- AUTO FXO (FCC mode)
Nov 28 14:01:55 voiplab kernel: Module 3: Installed -- AUTO FXO (FCC mode)
Nov 28 14:01:55 voiplab kernel: Found a Wildcard TDM: Wildcard TDM400P 
REV E/F (4 modules)

It seems that the first FXS module of my TDM22B is broken. Is that 
correct? In that case how can I disable it? Just open the case and pull 
it out? Or can I apply a configuration parameter to disable it?

Does this modules have a warranty? For how long?
Thanx in advance.
Andrés
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Re: [Asterisk-Users] Asterisk not startin anymore.

2004-11-28 Thread Andres Junge
el Flynn wrote:
Andres Junge wrote:
snip
It seems that the first FXS module of my TDM22B is broken. Is that 
correct? In that case how can I disable it? Just open the case and 
pull it out? Or can I apply a configuration parameter to disable it?

You should be able to do so by removing all reference to that 
particular module in /etc/zaptel.conf and /etc/asterisk/zapata.conf, 
without having to pull the module out.
The module havin problem is the first one (Channel 1) so do i need to 
renumber or just start the configs in /etc/zaptel.conf from channel 2?

although I remembered having a bum FXO module on my TDM22B, but it 
didn't cause the problem you encountered. might have been a different 
hardware problem though.

It worked on my previous asterisk! This problem arise when I upgraded to 
asterisk 1.0.2


Does this modules have a warranty? For how long?
IIRC, all Digium hardware should have a one-year warranty on it, 
although you may have to check with the people you bought it from.

I have bought it directly from digium. Do I check with them?
flynn
Thanx for your time.
Andrés
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[Asterisk-Users] CCM -(H323) - *

2004-08-12 Thread Andres Junge
Hi
I have found in 
http://lists.digium.com/pipermail/asterisk-users/2004-July/056111.html 
(Hack to make * - (H323) - CCM - IOS GW work) that i need a special 
version of chan_h323, because of the External RTP problem. Do you know 
exactly which version is it? Or do i need an unofficial patch?

Thanx
Andrés
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Re: [Asterisk-Users] Asterisk and Cisco Call Manager.

2004-08-10 Thread Andres Junge
No problem. First you need to know if the problem is that the 
CallManager is not sending anything or if Asterisk is not handling the 
conection. You can use tcpdump or ethereal for that.

Salu2
Andrés
Chad Whitten escribió:
Mind sharing how you got asterisk working with callmanager as an h323 gateway?  
I have configured callmanager with the ip address of my asterisk server and 
have setup asterisk with h323 and routed a call pattern to asterisk box but 
im not getting anything at the asterisk end.  

i have loaded the chan_h323.so module
im guessing my h323.conf needs work.
On Monday 09 August 2004 13:39, Andres Junge wrote:
 

I'm in the process of doing the same thing. My approach is to declare
asterisk as h323 gateway for the Cisco Call Manager, then define a route
pattern to call asterisk. The strange thing that i'm dealing with now
is, that the inbound RTP stream is going from the phone directly to
asterisk and asterisk is sending the outbound RTP stream to asterisk. I
don't know if this is a problem in asterisk or in the call manager.
Salu2
Andrés
Gurdeep Singh Bagga Guru escribió:
   

Hi All,
I am new to Asterisk and VOIP. I managed to get it working with sip(X-
Pro) and skinny(Cisco 7940,7960).
I have a call manager to which all the phones are connected. I would
like some assistance integrating CCM with Asterisk.
I was trying to understand the H323.conf file, but got nothing in it.
Any steps, any config, any help would be highly appreciated.
Thanks  Regards,
Gurdeep (Guru)
+91-11-35372111
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk and Cisco Call Manager.

2004-08-09 Thread Andres Junge
I'm in the process of doing the same thing. My approach is to declare 
asterisk as h323 gateway for the Cisco Call Manager, then define a route 
pattern to call asterisk. The strange thing that i'm dealing with now 
is, that the inbound RTP stream is going from the phone directly to 
asterisk and asterisk is sending the outbound RTP stream to asterisk. I 
don't know if this is a problem in asterisk or in the call manager.

Salu2
Andrés
Gurdeep Singh Bagga Guru escribió:
Hi All,
I am new to Asterisk and VOIP. I managed to get it working with sip(X-
Pro) and skinny(Cisco 7940,7960).
I have a call manager to which all the phones are connected. I would 
like some assistance integrating CCM with Asterisk. 

I was trying to understand the H323.conf file, but got nothing in it.
Any steps, any config, any help would be highly appreciated.
Thanks  Regards,
Gurdeep (Guru)
+91-11-35372111
[EMAIL PROTECTED]
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[Asterisk-Users] Strange H323 problem

2004-08-09 Thread Andres Junge
Hello.
I have a very strange H323 problem. This is the situation: I have a 
Cisco 7960 phone (with IP address 10.1.1.21) connected to a Cisco 
CallManager (with IP address 10.1.1.10) and an Asterisk with IP address 
10.1.1.22. I have managed to make the CallManager to call to asterisk 
using a route pattern. The strange thing is that when the call is 
established one RTP stream goes from 10.1.1.21 (phone) to 10.1.1.22 
(asterisk) and the other goes from 10.1.1.22 (asterisk) to 10.1.1.10 
(CallManager).
Can somebody point me if the problem is on the asterisk side or in the 
CallManager side?

If somebody needs it, I have  the complete  ethereal session in 
http://www.totexa.cl/ccm_session

Thanks.
Andrés
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Re: [Asterisk-Users] TDM04B Dead?

2004-07-23 Thread Andres Junge
What is a RMA?
Isamar Maia escribió:
Just for curiosity, Let us know how much time you'll gonna get a RMA of
it.
Isamar
On Thu, 22 Jul 2004, Andres Junge wrote:
 

I had the same problem, and it was that the power suppply coudn't handle
the new card. My solution (until i get a new power supply) was to unplug
a very big fan that i have in the case.
Salu2
Andr?s
Greg Hulands escribi?:
   

Hi,
I just received in the mail my TDM04B card and put it in the computer,
now the computer won't even show the video card bios or the post
screen. From the digium website I could not find any specific
requirements for the pci card, like 32 or 64 bit slot. The motherboard
for the computer I put it in is an Asus A7V333 with PCI 2.2 compliant
slots. I am thinking that maybe I just got a dud card. Is there
anything I need to change or I can test to see why it is not letting
the computer boot?
Any help is greatly appreciated.
Regards,
Greg
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Re: [Asterisk-Users] TDM04B Dead?

2004-07-22 Thread Andres Junge
I had the same problem, and it was that the power suppply coudn't handle 
the new card. My solution (until i get a new power supply) was to unplug 
a very big fan that i have in the case.

Salu2
Andrés
Greg Hulands escribió:
Hi,
I just received in the mail my TDM04B card and put it in the computer, 
now the computer won't even show the video card bios or the post 
screen. From the digium website I could not find any specific 
requirements for the pci card, like 32 or 64 bit slot. The motherboard 
for the computer I put it in is an Asus A7V333 with PCI 2.2 compliant 
slots. I am thinking that maybe I just got a dud card. Is there 
anything I need to change or I can test to see why it is not letting 
the computer boot?

Any help is greatly appreciated.
Regards,
Greg
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