We have the following problem with our asterisk
system:
We have a very long no. 0893168205 (14 digits) or perhaps
minus the 1st 3 digits as that is the area code.
Some people are having issues when they phone from certain
PABX´s or have to try many times to get through, which is really
This one is driving me crazy. So any suggestions will be very welcome.
My setup:
Suse Linux 9.0 (Pentium 4, 1GB)
Asterisk: current stable (1.0.3?) - tried the head CVS before Christmas but
did not fix it
2 X100P clones - one for a UK BT line, one connected to an ATA186
configured for a UK BT
Hello,
Am i right when i suppose that the chan_cornet will replace the oh.323.
[OPTIPOINT400_HFA]--[HIPAT4K][chan_cornet][ASTERISK]--[OPTIPOINT400_SIP].
Yes, that's the idea. Let's look if we have success :)
cu,
Steffen
___
Both of the X100Ps seem to randomly hang-up both incoming and outgoing
calls.
I think most people who use X100P cards (clone or originals) have had your
experience. So far as I can tell, the cause is always an interrupt problem.
Specifically that affected X100P cards share an interrupt with
Thanks for the reply Bill.
I am aware of the interrupts problem. To solve it I have already disabled
my serial ports freeing up interrupts 3 and 4 and these are allocated to
the two cards. This was done 2 months ago and has not solved the problem.
Is there any way that something can wake up
Hi
Does anybody knows about AOC for PRI channels ?
I was looking for it and I have found only support for Sirrix ISDN
channels. :(
thank you
dudl
___
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I have the same problems with the reload command.
I found, it is the chan_sip.so which causes the hang.
A simple reload of chan_sip.so leaves asterisk in this state, where it does
not react on any commands in the CLI.
I've tried to open a bug on bugtracker for it, but I was told it would be a
dupe
On Sun, 2005-01-09 at 12:10 +0100, Hecken, Guido wrote:
I have the same problems with the reload command.
I found, it is the chan_sip.so which causes the hang.
A simple reload of chan_sip.so leaves asterisk in this state, where it does
not react on any commands in the CLI.
I have noticed that
Dave,
yes I have an external registration to sipgate. I commented
register = xxx:[EMAIL PROTECTED]/xx in sip.conf
in context general, but the problem still remains.
Do I have to comment out the sipgate context in sip.conf as well?
[sipgate]
type=friend
username=x
Hi Clive,
I ran into a similar problem: I also have a eicon 4bri and tried to
install it on a dell server with redhat as 3.
The problem I have is that I always got a error message when doing
modprobe capi. The module is compiled in the kernel (it shows up with
lsmod).
I have not found any
Yay, I got it working.!
I added CAPI verbose reason reporting to the kernel, and modprobed
as follows:
capi, kernelcapi, divacapi, divas and then loaded divactrl and it works!!,
yaynow to figure out how to get Asterisk to load with capi...my .conf
files seem to be wrong
Thanks for
How does the phone know which server's address, did you set this with
DHCP or directly into the phone. If you already set this information
can you verify it in some way. If you are not seeing anything on the
server then the phone is most likely not talking to it for some reason.
Daniel Joos
OK Some more information.
I have changed interupts for the 2 X100Ps and the problem did not go away.
Looking at the detailed debug information I always get the following from
before the disconnect:
Jan 9 12:13:40 DEBUG[1992]: Exception on 17, channel 1
Jan 9
On Sun, 9 Jan 2005, dudlik wrote:
Does anybody knows about AOC for PRI channels ?
I was looking for it and I have found only support for Sirrix ISDN
channels. :(
From what I can see in the source support for AOC has not been implemented
in asterisk yet.
The three forms (AOC-S, AOC-D and
On Sun, 2005-01-09 at 12:39 +0100, Hecken, Guido wrote:
Do I have to comment out the sipgate context in sip.conf as well?
I'd try that as well.
If it still hangs you'll have to look else where.
--
Dave Cotton [EMAIL PROTECTED]
___
Hi there
I have
troubles while installing oh323.
I got
from cvs the asterisk_oh323, and then I tried to make install, in the end I have this:
...
/usr/include/ptlib/array.h:341:
`dynamic' undeclared (first use this function)
/usr/include/ptlib/array.h:341:
only constructors take
Hello,
is it possible to wait indefinitely (i.e. until user hangs up) somewhere
in the dialplan? I tried Wait(-1), but it doesn't work.
___
Asterisk-Users mailing list
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each vendor for rtp. Cisco uses one range, xlite another, asterisk
another, etc, etc. Mapping the sip port (udp 5060) is easy; mapping
the rtp ports and using the proper nat statements (possibly at both
the phone location and asterisk location) tends to be difficult. Then
X-Lite can be told
Well I guess I need to fix or create a channel now.
Asterisk Ready.
*CLI dial [EMAIL PROTECTED]
Jan 9 10:28:06 NOTICE[10750]: app_dial.c:743 dial_exec: Unable to create
channel of type 'Zap'
No luck when I dial [EMAIL PROTECTED]
David
David wrote:
Hello All, I'm trying to dial out with no
Hi,
Has anyone in australia got asterisk running on FreeBSD? how would i
pass the opermode=AUSTRALIA parameter to the wcfxs.ko module as kldload
doesnt let you pass parameters to the module like modprobe in Linux.
I tried to get the sysctl variable using sysctl -a it might use but
nothing
I downloaded latest * stable complile it successfully but
when compiling the asterisk-addons the res_config_mysql.so
is missing.
I followed the
instructions on wiki for Realtime.
What did you do wrong ?
Thanx,
___
Asterisk-Users
Hi,
I have Siemens combiset - it can gateway GSM phone to normal analog phone.
It has output where I can connect regulat analog phone.
How can I connect to combiset with Asterisk - via fxo or fxs ?
Thanks,
regards,
Robert.
___
Asterisk-Users
Serge Schumacher wrote:
I downloaded latest * stable complile it successfully but when compiling
the asterisk-addons the res_config_mysql.so is missing.
The stable version of Asterisk does not have Realtime support.
I followed the instructions on wiki for Realtime.
If the Wiki is telling you can
Bonjour,
what about a goto?
Christian Gatti
Visual Online S.A.
Sunday, January 9, 2005, 4:42:23 PM, you wrote:
Hello,
is it possible to wait indefinitely (i.e. until user hangs up) somewhere
in the dialplan? I tried Wait(-1), but it doesn't work.
Ah grmpf :), CVS head is the lastest version ? so ?
cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot co asterisk
asterisk-addons ??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: dimanche 9 janvier 2005 16:58
To: Asterisk Users
I've found, when upgrading from earlier releases that do not support
realtime (e.g., 1.0.1), you must first make install from the asterisk
directory before attempting to build asterisk-addons.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Sun, 9 Jan 2005, Serge
Hi all,
My office recently purchased an InterTel Axxess system with the IPRC
card for VoIP. To our suprise, this card allows the InterTel endpoints
and MGCP endpoints to work, but not SIP clients. I was really
expecting to get a SIP softphone working with this setup, but that
appears to
I am using a combo of static files and Asterisk
Realtime configuration. This section works fine when a
static file:
---
[from_pstn]
;Voipgate
exten = 4507,1,Goto(from_pstn,s,1)
exten = s,1,Macro(dial-ext)
exten = s,2,Hangup
---
But, when I drop it
Console:
*CLI dial [EMAIL PROTECTED]
No such extension '650' in context
'from-sip'
Extentions.conf
[from-sip]
switch = Realtime/@realtime_ext
extconfig
realtime_ext = mysql,asterisk,extensions_table
res_mysql.conf
[general]
dbhost = 127.0.0.1
dbname = asterisk
Quick update on my
issues, Voicemail doesn't pickup also. It just drops the
line..
Thank you
Chris Tuska
--
Hello All,I have Cisco 7960's,
Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out
no issues but when someone calls in the phone
I am looking for some in India to buy VOIP phones from. Please get in touch
with me off the list on [EMAIL PROTECTED]
Sorry for the off topic mail I am just having such a hard time finding any
voip phones in India that I got desperate and didnt know which list to post
this on.
--
regards
Shoval Tomer wrote:
Hi all.
Can anyone comment why shouldn't we use FC 3 for an * production system?
Depends. If you have already chosen FC3, then I would assume that you
are comfortable with its limitations (services are community rather than
vendor based, there is a fair bit of
Joseph wrote:
I would like to know more about your solution.
My solution involves a patch to app_queue that essentially makes it call
SetGroup() on any channels it creates to call queue members (agents),
and call GetGroupMatchCount() before calling a member to see if they
should be considered
Robert Jackson wrote:
Another possible scenario is to specify the context to call the agent
when using AgentCallBackLogin. This way you can have one set of
behaviors for reaching an agent at an extension and another set for
simply reaching the extension outside of an ACD context.
Yes, that
I hva ean HFC-S card in a box that I'm trying to get to work with
bristuffed Asterisk 1.0.3. The box is an Athlon64 running a RHEL rebuild
with a plain vanilla 2.6.10 kernel. I tried both APIC and NOAPIC mode.
The installation went ok and does give output that seems correct
SPAN 1: CCS/ AMI
The Asterisk Documentation Project is proud to present, The History
of the Zapata Telephony Project as it relates to the Asterisk PBX.
Written by Jim Dixon, the founding father of the Zapata telephony project
(http://www.zapatatelephony.org) which started a revolution in
computer telephony.
This
Dear List members-
I am trying to configure ASTCC (Asterisk calling card application) but
having a hard time to configure it properly. My project deadline is
approaching and couldn't figure out how to make ASTCC functional. Here
are some details what I have done so far.
1) I have installed
Hi,
Well I found this in the wiki
which just looks like a good match, does anyone know how to and where
thisthe chan_vpb.c isso that I can change it and also how to add the
wait on the dialplan
Asterisk vpb channels: USA Caller ID (Hits: 2219)
(Relevance: 11.329)
(:biggrin:) If you
Hi,
I need to setup a
demo for asterisk and need some help here please. The demo is connecting to
Asterisk a Cisco 7970 SIP (ver. .0) and a SIP client on HP iPAQvia a
wireless hotspot. I need to configure both with the same extension with a shared
line like in Cisco CallManager. This way
Hello Every
one
I need to enable
Asterisk to route external land line calls to the PSTN. Regarding our
environment, we have Cisco CallManager (3.3.4) to which IP phones are connected.
E1 terminated on a couple of As 5300's which are controlled by a soft switch
(Cisco PGW200 Call Control).
I had a *lot* of trouble with echo when I first set * up! I'm using an
x101p clone.
The solution I found was to change 1 line in one of the zaptel source files and
recompile.
The file is zconfig.h and I uncommented:
#define AGGRESSIVE_SUPPRESSOR
I've got 2 x101p cards in and I've got them on
Hi Mike,
The solution I found was to change 1 line in one of the zaptel source files
and
recompile.
The file is zconfig.h and I uncommented:
#define AGGRESSIVE_SUPPRESSOR
Great - i'll give this a try.
What settings do you have in zapata.conf then?
Thanks
--ian
Anyone know in the current zaptel drivers and stable asterisk what the
parameters are to receive caller ID in the UK over BT lines?
Thanks
Looked at the Wiki and bugs.digium but more confused, perhaps someone
can help me
John
___
Asterisk-Users
On Sun, 9 Jan 2005, Remco Barende wrote:
I hva ean HFC-S card in a box that I'm trying to get to work with bristuffed
Asterisk 1.0.3. The box is an Athlon64 running a RHEL rebuild with a plain
vanilla 2.6.10 kernel. I tried both APIC and NOAPIC mode.
The installation went ok and does give
Sirs,
I have a question about CreateConnection (CRCX) at MGCP.
For example, I have the phone number 5220107 and want to make a call for
it using MGCP through a media gateway. How can I proceed?
I know the command I must send to the media gateway should be like this:
CRCX trans_id endpoint MGCP
On Mon, 2005-01-10 at 07:51, Walid Azab wrote:
Hi,
I need to setup a demo for asterisk and need some help here please.
The demo is connecting to Asterisk a Cisco 7970 SIP (ver. .0) and a
SIP client on HP iPAQ via a wireless hotspot. I need to configure both
with the same extension with a
Sirs,
I have a question about CreateConnection (CRCX) at MGCP.
For example, I have the phone number 5220107 and want to make a call for
it using MGCP through a media gateway. How can I proceed?
I know the command I must send to the media gateway should be like this:
CRCX trans_id endpoint MGCP
I need to setup a demo for asterisk and need some help here please. The demo
is connecting to
Asterisk a Cisco 7970 SIP (ver. .0) and a SIP
client on HP iPAQ via a wireless hotspot. I need to configure both with the
same extension
with a shared line like in Cisco CallManager. This
way if
For some thime now I have been trying to get some sense out of Caller ID
on PSTN lines in AU and have been getting no where. Now, at last, I
seem to be getting something, even though it is an error message (line 5
below).
All and any guidance would be welcome.
-- Starting simple switch on
In the real world (or at least in my world) we use undersubscribed
internet connections that come with a service level agreement (SLA) that
guarantees that the jitter, delay, and packet loss with be within
defined parameters in the service agreement.
With most DSL and Cable you will not get a
Do not top post.
Trim the posts - there were 3 list signature blocks
On Sunday 09 January 2005 22:47, Damon Estep wrote:
In the real world (or at least in my world) we use undersubscribed
internet connections that come with a service level agreement (SLA) that
guarantees that the jitter,
Hi,
They are using equipments from Cirpack.
I dont understand :
Solved it by adding progress_ind
setup enable 3 on the voip peer.
Where should i add this parameter ?
Thanks for your help,
AB
Oswaldo Arratia
oarratia at workersequity.net
Tue Jan 4 09:59:12 CST 2005
What type of equipment
Has anyone found an inexpensive EM trunk card that will play with *?
Looking for an interface to a legacy electromechanical PBX that's able
to pass answer supervision. Docs on the X100P card would be helpful, we
could probably pull EM out of that. Any ideas?
As far as I can see the problem is with the Read() function.
When this function times out due to no user input, the extension is
terminated
and the call is hung up. Maybe this was the intended behaviour, but I
can't see
how its of any use.. Surely this is not the most desirable behaviour ?
Lane wrote:
Hi, again.
I've spent a week trying to get asterisk to work on FreeBSD unix, with some
success. Everything works until I plug the box into the TELCO line and then
the line goes off-hook and stays that way.
I wasn't able to get the zaptel stuff working under 5.3, but that has
more to
I had the same problem and I see that this was addressed in udev-043 :
http://lwn.net/Articles/111858/?format=printable
(search for zaptel)
FC3 has udev-039-8.FC3 installed by default. If you run up2date, an
update to udev-039-10.FC3.6 is available that fixes this problem. Also
the typical zaptel
When calling to the PSTN (outside VOIP or *) then you will not be able
to supply the name of callerID even if you have a PRI. The only thing
you can provide is the number and the receiving switch of the call is
the one responsibble for attaching a name to the phone number thru
SS7. If you have a
If you are using a VOIP provider and they are using just PRI (and not
SS7, or they didn't configure their SS7), and change the callerID to
the number you supply, then the name should show up, to whatever it is
listed with the local phone company, i.e. if your phone number is
5551234 and verizon is
OK here it goes..
Caller ID is two parts or actually three:
Part 1 Number only
Part 2 Number + Name
Part 3 Whole lotta stuff (also known as ADSI)
Here is the US, I cannot speak for other countries.
When party A places a call to Party B. Party A's Telco picks up the
number, either from a
Alexander Lopez wrote:
Most Telcos do not receive the Name as part of the data in the call
through the tandems b/w Telcos, they opt rather to do the lookup in the
LIDB themselves.
Just for the sake of completeness: most telcos do not would imply that
some telcos *do*.
You also say they opt to
Caller Name is stored in a SCP. It is a TCAP transaction. The receiving
switch via SS7 recieves
the calling party number in the ISUP message of the SS7 datastream. It is
normally in the IAM mesasge. Then a TCAP CNAME query is launched from the
called switch thru
the STP's to a SCP which has
There should be no quotes after the : in the cisco SIPmacaddress.cnf files.
Change it from:
# Line 1
line1_name: Scott
line1_authname: scott
line1_password: scott
# Line 2
line2_name: Scott1
line2_authname: scott1
line2_password: scott1
To:
# Line 1
line1_name: Scott
line1_authname:scott
In my six years of SS7 work I have never seen the calling name generated by
the calling switch and passed via the SS7 network. Normally and of all the
installations
that I seen, it is done by the called switch via a TCAP query to a SCP
database.
Tom c.
- Original Message -
From: Brian
Is the TCAP DB part of the LIDB collective (no Borg pun intended)??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
Chandler
Sent: Sunday, January 09, 2005 6:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; C F
Subject: Re:
TCAP is a transaction application. The CNAME, LIDB,800,.LNP and AIN
database COULD
be in the same SCP, but in most cases it is not. LIDB database are used for
calling card, operator
services, etc. These are all seperate databases stored for use in an SCP
connected to STP's.
So is there a
Title: Message
Can anyone post a
set of instructions on how to install this card?
I have
RH9Linux and the latest Asterisk.
I have triedto
install the card using instructions from Quicknet and the card works for a
little while and then Asterisk issues a message about the driver missing a
Thanks. I was always under the impression that they were all separate
tables in the same DB and that they were collectively called 'The
LIDB!!'
For my and the others here could you describe the function of the
different DBs?
I now understand the CNAME, I thought I knew the LIBD, I can guess on
Caller id is not in the configs that are supplied with zaphfc from
bristuff, and it's not mentioned in the wiki.
Is it possible to use callerid with zaphfc/bristuff?
Thanks!!
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I made this change in my sip.conf file, I
removed allow=gsm, allow=alaw and now everthing works great.
Chris Tuska
[general]disallow=allallow=gsmallow=ulawallow=alaw;
My PSTN Service
provider[Sipmedia]disallow=allallow=gsmallow=ulawallow=alaw
Hi ,
I'm fairly new user but as far as I understood Asterisk architecture, and
to have something working with your ASTCC installation, you should
do as follow:
Scenario
- You have a pstn provider, say for example voipjet (i have no interest
with them, but it's straight forward to setup a trunk
- Original Message -
From: Alexander Lopez [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, January 09, 2005 5:55 PM
Subject: RE: [Asterisk-Users] Little confused about Caller ID
Thanks. I was always under the
In the real world (or at least in my world) we use undersubscribed
internet connections that come with a service level agreement (SLA)
that
guarantees that the jitter, delay, and packet loss with be within
defined parameters in the service agreement.
[...]
In the real world (or
Robert Rozman wrote:
Hi,
I have Siemens combiset - it can gateway GSM phone to normal analog phone.
It has output where I can connect regulat analog phone.
How can I connect to combiset with Asterisk - via fxo or fxs ?
It is my understanding that an FXS device generates dial tone, and a FXO
Without going into detail of what the query is actualy called, when
the called switch make the query to find out the name, does it ask it
from the originating switch? or it askes it from the switch that is
responsibble for servicing the number?
The difference would be if, I have a PRI from
He has a similar story:
http://lists.digium.com/pipermail/asterisk-users/2005-January/082034.html
On Sun, 9 Jan 2005 20:14:05 -0500, C F [EMAIL PROTECTED] wrote:
Without going into detail of what the query is actualy called, when
the called switch make the query to find out the name, does it
Try flash oprator panel
On Fri, 07 Jan 2005 10:48:47 +0100, lokotes [EMAIL PROTECTED] wrote:
Hi,
is it possible to transfer an incomming call to another ext. without
answering? I'm not talking about (un)conditional redirection but
functionality, when calee can each time decide whether answer
[EMAIL PROTECTED] wrote:
Hi,
I have Siemens combiset - it can gateway GSM phone to normal
analog phone. It has output where I can connect regulat analog phone.
That would be an FXS connection.
How can I connect to combiset with Asterisk - via fxo or fxs ?
You would need to connect the
On Sun, 2005-01-09 at 12:39 +0100, Philipp Ebneter wrote:
[snip]
I ran into a similar problem: I also have a eicon 4bri and tried to
install it on a dell server with redhat as 3.
The problem I have is that I always got a error message when doing
modprobe capi. The module is compiled in the
These comments are based on domestic SS7. International SS7 works different
and
when you internetwork between domestic and international SS7 everything
changes.
Sorry about long post..
TC.
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Sorry, in the post I wrote:
I used to work with an Avaya Difinity G3. We had PRI, which gave us
incoming CallerID (Name Only),
It should read:
I used to work with an Avaya Difinity G3. We had PRI, which gave us
incoming CallerID (Number Only),
Sorry for the mistake.
And to Mr Tom C. thanks a
Thanks,
So what are the fresholds of the jitter, delay, and
packet loss I should be asking my ISP for?
robert
--- Damon Estep [EMAIL PROTECTED] wrote:
In the real world (or at least in my world) we use
undersubscribed
internet connections that come with a service level
agreement (SLA) that
Hi,
I have the following requirements I'd like to implement with asterisk:
1. Asterisk notifies interested PC's on the network that there's an
incoming call so that the telemarketing app can bring up the customer
automagically
2. If a telemarketer makes a call and the customer isn't there and
Thanks,
So what are the fresholds of the jitter, delay, and
packet loss I should be asking my ISP for?
robert
On a T1 you should expect an SLA that states;
Latency - 80ms round trip latency between your end and the ISP core
routers within a few thousand miles, SLAs typically only cover
Hi Steve -
Is * a proper tool to provide a SIP-MGCP gateway? Am I even asking
for something that makes sense?
Sure. I think * is perfect for what you describe. In fact, I don't
really know of another tool that could bridge SIP and MGCP.
If so, where's the best from the ground up, assume I
Oh yeah, the other thing I wanted to ask was about getting the
telemarketing app to dial. Currently the application sends a dial string
to a modem on the user's desktop pc to initiate the dialling, but I
would like it to call asterisk to do the dialling directly. I guess the
order of events would
So while raising the ire of the group by being associated with a
telemarketer, you show that you really just need to familiarize yourself
with asterisk and the features available to you. So far you haven't ask
a question that requires even a moderate user a moment of thought.
Initiate a call,
I have an asterisk with a TE110P configured as T1 which is behind a PSTN
gateway. This gateway has an E1 to PSTN and a T1 to asterisk. This T1 is
configured as Network and * as CPE.
Every call I receive in E1 gateway is directly switched to asterisk using
T1. Remember E1 is alaw. Both E1 and
Hi,
I'm currently working on something like that. Still under dev. but should
have more developpement this winter.
I've got a 'dumb' dialer (no prediction) written in perl (POE) that
monitor asterisk and launch calls. Something like Shadydial, but more
simple to understand/customize. This part
Its still fails!
[EMAIL PROTECTED] apps]# patch apps_makefile.patch.new
patching file Makefile
Hunk #1 succeeded at 42 with fuzz 2 (offset -7 lines).
Hunk #2 FAILED at 73.
1 out of 2 hunks FAILED -- saving rejects to file Makefile.rej
On Fri, 2005-01-07 at 22:08, Jim Radford wrote:
Basically
On Sun, 2005-01-09 at 00:12 +, Bill Seddon wrote:
rmdir will only remove empty folders and --ignore just prevents error
messages being displayed.
Run the command: rm -rf *
in the asterisk root folder and then execute rmdir
Firstly, this won't work, since there are a couple of . files
hi all,
its my first time to post here, im in the process of building asterisk
based telephone system (just small). i already installed asterisk
server, i just wanted to test 2 sip softphones to get working before i
move on, is it possible to have 2 softphones talk to each other without
any
I have been having this exact problem with a Tatung dual EMT-64 server as
well.
I have been trying to get a TE410P running and all looks great, driver
loads, runs ztcfg OK, etc. but no interrupts are ever processed.
One additional piece of info that I have not seen in this thread is that I
am
Greetings,
I have a VoIP provider that uses H323. Basically they provide a
proprietary win32 client for you to login with a username and password.
I've ran ethereal through it and have sort of figured out how it works,
but I know very little about H323.
What I want to do is, to have asterisk
Make sure you has a span defined for each port on the TE410P. With out
signaling it would not take interrupts.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl H.
Putz
Sent: Monday, January 10, 2005 12:38 AM
To: Asterisk Users Mailing List -
Good day all
We got a Wildcard TE110P
I installed linux,zaptel,libpti and asterisk
I coped over my zaptel.conf and zapata.conf from a previous E100P config
But when I try to start asterisk it gives error not bying able to load
zap channles:
== Parsing '/etc/asterisk/zapata.conf': Found
Jan 10
On Mon, 2005-01-10 at 16:00, Altus Snyman wrote:
Its still fails!
[EMAIL PROTECTED] apps]# patch apps_makefile.patch.new
patching file Makefile
Hunk #1 succeeded at 42 with fuzz 2 (offset -7 lines).
Hunk #2 FAILED at 73.
1 out of 2 hunks FAILED -- saving rejects to file Makefile.rej
Yep,
Kavit Munshi wrote:
Hi,
Has anyone in australia got asterisk running on FreeBSD? how would i
pass the opermode=AUSTRALIA parameter to the wcfxs.ko module as kldload
doesnt let you pass parameters to the module like modprobe in Linux.
I tried to get the sysctl variable using sysctl -a it might
You are using a PRI based config for POTS lines. It will no worky. Post
your zap*.conf files.
I'll take a look at them for you..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Altus
Snyman
Sent: Monday, January 10, 2005 1:24 AM
To: asterisk
Subject:
On Mon, 2005-01-10 at 01:33 -0500, Alexander Lopez wrote:
You are using a PRI based config for POTS lines. It will no worky. Post
your zap*.conf files.
I'll take a look at them for you..
How do you plug analog lines into a T1/E1 card?
A better guess is either the driver for the card
Sorry, its late here in Miami. 2:00AM. I thought it was a TDMXX card.
Read too quickly. Maybe its time to go to sleep.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Monday, January 10, 2005 1:42 AM
To: Asterisk Users Mailing
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