On Thu, 2005-10-20 at 22:52 -0700, snacktime wrote:
When you say software for the gateways, you mean you integrated with
third party software that connected to one of the processing networks?
Do you remember the name of the network that was used? I guess there
could be a network that checks
Please move all discussions about this service provider to the
asterisk-biz list.
Thank you!
/O
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Ted Cabeen wrote:
In August and September of last year, there was some discussion of
changing the Voicemail and Record applications to send back CNG RTP
packets during recording to prevent inbound calls from dropping when
they assumed a disconnect after 30 seconds of no RTP frames.
Was
On 10/20/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
On Thu, 2005-10-20 at 22:52 -0700, snacktime wrote: When you say software for the gateways, you mean you integrated with third party software that connected to one of the processing networks? Do you remember the name of the network
What about SMS or LEC billing? The second may be more difficult, but there
are many solutions with SMS, even if it is just transferring contact
information (instead of hiring operators to typing in data with the keypad).
The first can be used alone or in conjunction with credit cards and direct
AltusIt's in the transcoding - http://www.voip-info.org/wiki-Asterisk+dimensioning has some notes on oh323 v.s. chan_h323 (chan_h323 is just pass through) - someone says there that you won't be able to run more than
20-25 decent quality calls before asterisk dies when transcoding and H323 are
hi
does any body know a CLI SIP CLIENT
for LINUX except Linphone?
thanks in advance
The information contained in this electronic message and any attachments to
this message are intended for the exclusive use of the addressee(s)and may
contain confidential or privileged information. If you
On Thu, 2005-10-20 at 23:38 -0700, snacktime wrote:
Ok that makes sense. There was an extra layer of checking done before
the authorization was sent to the processing network. The bank
mandated the extra checking which trintech had to perform.
well technically it was visa, mastercard and
I've been thinking of using yate
http://yate.null.ro/pmwiki/index.php/Main/H323ToSIPSignallingProxy to do
this. Any thoughts or experiences?
Darren Wiebe
[EMAIL PROTECTED]
Rob Lith wrote:
Altus
It's in the transcoding -
http://www.voip-info.org/wiki-Asterisk+dimensioning has some notes
On Thu, 2005-10-20 at 23:37 -0700, Justin Newman wrote:
What about SMS or LEC billing? The second may be more difficult, but there
are many solutions with SMS, even if it is just transferring contact
information (instead of hiring operators to typing in data with the keypad).
The first can be
I'm looking for a good way to do this as well.
Can someone recommend a merchant service that works
well with asterisk systems?
I have heard systems that have you spell your name
letter by letter, system beeps to prompt each letter,
then runs a speech to text routine that then enters
the data.
Mine stopped working sometime back in Feb. I just made the changes
so everything points to fwdOUT.net now, but it still seems to fail.
Using a sniffer, I see packets going out, but none coming back. I
have a firewall, but 4569 has been opened, and I'm not seeing denys
on the firewall
On 10/21/05 07:42 Mojo with Horan Company, LLC said the following:
Sorry, didn't answer your other question. I don't know why you couldn't
put both W and w together in a Dial command. You don't really want
customers starting a recording, but they're not likely to figure out
how, right?
This should give you a guide.
http://www.oinko.net/astrecipes/index.php?n=102
I have been using both H323 and OH323 with no big problems since * 0.7.
The only thing you notice is an added need for restarting * on busy
machines.
Bye
l.
On Thu, 20 Oct 2005 14:23:53 +0200, Carlos Arnt [EMAIL
Hello Waldo,
if you use AddQueueMember plus a fake queue_log registration, you can tell
who the agent was, not just from what terminal she was connecting from. It
is then possible to report who was available at a certain time, or see
agents logging on and off, going to pause, measuring the
On Fri, 2005-10-21 at 02:14 -0500, Jay Austad wrote:
Mine stopped working sometime back in Feb. I just made the changes
so everything points to fwdOUT.net now, but it still seems to fail.
Using a sniffer, I see packets going out, but none coming back. I
have a firewall, but 4569 has
Hi all,
I've setup an asterisk gw (yyy.yyy.yyy.yyy), connected via IAX2 with another asterisk (xxx.xxx.xxx.xxx) which acts as a centrex, and via SIP for
incoming/outgoing calls with a provider (zzz.zzz.zzz.zzz).
The problem is that outgoing calls are hanguped after three rings if they are not
I have done the mfc r2 setup as described in the wiki but
when I try a call
Oct 21 10:02:24 WARNING[14127]: channel.c:1913 ast_request: No channel type
registered for 'UniCall'
Oct 21 10:02:24 NOTICE[14127]: app_dial.c:764 dial_exec: Unable to create
channel of type 'UniCall'
== Everyone is
Lenz,
Thanks for the response. I agree with you. However, I have a couple
of questions:
1) How to do a fake queue_log registration
2) One of the needs I have is to be able to generate the calls
received or made by an agent in real time. I figured I could do this
by querying the CDR, but
hello list,
i have a dial plan
exten =12345,1,Dial(sip/100sip/101sip/102,30,rt)
so when calls come to 12345 all the phones 100, 101,102 rings.
if any person receives the call it connects the call with no problem. but
in the CDR it does not show who receied the call.
it shows
hello list,
in my asterisk i have blind transfer and attendent transfer.
when call Z which is a public call through Capi(BRI) is received by user A he can see the Caller ID of Z and
if user A blind transfer the call to user B, user B can see the caller ID of user Z but
when user A attendent
hi
i heard some talk about something in zaptel is currently incompatible
with 2.6.13.
is this so?
if so, will this be fixed soon?
thanks
roy
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Hi Waldo,
about how to create fake queue_log entries, the answer is quite simple:
see http://www.oinko.net/astrecipes/index.php?n=100
I instead doubt that you can use the CDR for real-time logging, as CDR
data usually gets written when the call ends. Of course you can hack with
it, but
Im trying to write a small php script wich will connect to oracle DB and
get me some information.
Ive already have a script wich perfectly works with apache.
But I cant launch it from asterisk
I think its becouse i should have some global enviroments
ORACLE_BASE and others
I have [EMAIL
I'm using zaptel on FC4 with 2.6.13. and it works good.
Bob.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd
Karlsbakk
Sent: Friday, October 21, 2005 10:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On Fri, Oct 21, 2005 at 10:44:08AM +0200, Roy Sigurd Karlsbakk wrote:
hi
i heard some talk about something in zaptel is currently incompatible
with 2.6.13.
is this so?
if so, will this be fixed soon?
zaptel 1.0.9.2's release notes has something about 2.6.13 kernels.
--
Tzafrir Cohen
I can't even call out anymore! That stopped working probably 3 or 4
days ago. My iax looks pretty much the same as yours. I always get
circuits busy. :(On 10/20/05, Lilantha Karunaratne [EMAIL PROTECTED] wrote:
Guess we all have the incoming problem!
When I make a call out form
Thanks for the update. I can make calls
out to 1-800 numbers in the US
but IAX to IAX between the 2 offices doesnt work which runs purely on
IP. Thats the problem were having.
Cheers!
Lilantha
From: Blake Krone
[mailto:[EMAIL PROTECTED]
Sent: Friday,
Hi List
What are the project status of func_odbc.c? Is it part of *? Maintained
by some body else? Deprecated? Obsolete? Only alpha/beta state?
I'm building a custom prepaid solution on top of *, and therefor I want
* to check the users account before initiation the dial command. After
Hi all,
I've setup an asterisk gw (yyy.yyy.yyy.yyy), connected via IAX2 with another
asterisk (xxx.xxx.xxx.xxx) which acts as a centrex, and via SIP for
incoming/outgoing calls with a provider (zzz.zzz.zzz.zzz).
The problem is that outgoing calls are hanguped after three rings if they are
not
Tom Tune wrote:
It happens on everything I have tried. I do not have any vlans or qos
configured. Even on a generic 5 port with only the phone and asterisk
server connected.
I am beginning to think that it is a Microsoft style mutual upgrade
scenario.. can't use the phones without the switch.
i heard some talk about something in zaptel is currently incompatible
with 2.6.13.
is this so?
if so, will this be fixed soon?
zaptel 1.0.9.2's release notes has something about 2.6.13 kernels.
I was thinking more about the CVS HEAD version
roy
On 21 Oct 2005, at 06:42, Andreas Mavrides wrote: mine is 339Mine was working yesterday, but it was not supporting the GSM codec, which was supported for the free-to48-states outbound service when it was running.How is your asterisk configured? Do you permit the GSM Codec?Tim.
no i only allow ulaw
- Original Message -
From:
tim
panton
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, October 21, 2005 1:15
PM
Subject: Re: [Asterisk-Users] Goiax.com
DID not working anymore?
On 21 Oct 2005, at 06:42,
Just thought I'd let everyone know that a new revision has popped out from
Digium: Rev J. I
don't have an I board in front of me to compare with,
so I can't tell you what's different (besides a bunch more text on the back).
It looks like
there is a PE-68624 chip near each RJ-45
Olle E. Johansson wrote:
Please move all discussions about this service provider to the
asterisk-biz list.
Thank you!
It's a free service. It belongs on this list.
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I have put in the extension.conf
exten = 135,1,Dial(SIP/[EMAIL PROTECTED])
and nothing in the sip.conf, the extensions can dial each other fine.
But whe i put the sip.conf:
[Oswel]
type=friend
regexten=135
username=135
secret=1234
callerid=Osie
host=192.168.1.35
canreinvite=no
disallow=no
On Fri, 2005-10-21 at 12:11 +0200, Roy Sigurd Karlsbakk wrote:
i heard some talk about something in zaptel is currently incompatible
with 2.6.13.
is this so?
if so, will this be fixed soon?
zaptel 1.0.9.2's release notes has something about 2.6.13 kernels.
I was thinking more
I have put in the extension.conf
exten = 135,1,Dial(SIP/[EMAIL PROTECTED])
exten = 125,1,Dial(SIP/[EMAIL PROTECTED])
and nothing in the sip.conf, the extensions can dial each other fine.
But when i put the sip.conf:
[Oswel]
type=friend
regexten=135
username=135
secret=1234
callerid=Osie
Hi, all
Is there a way in extensions.conf to exclude context?
For example, I have contexts A, B and C.
I want something like this:
[A]
some extensions
[B]
some extensions
include=A
[C]
some extensions
include=B
excludeA
If I only do include=B in [C], it will automatically include [A] in
Hello for all
I have the [EMAIL PROTECTED] configured and everything is ok , but i am a
portuguese speaker and i want to configure the festival to say in
portuguese,
Somebody can help me with this problem ?
Thanks or all
Lucio
___
Lucio,
Se puder ajudar de alguma forma...
Abs,
Fábio Sakai
DGX - Digital Express
Suporte CosmoCall
[EMAIL PROTECTED]
+55 11 3049.8109
-Mensagem original-
De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Lucio de Aquino
Marinho
Enviada em: sexta-feira, 21 de outubro de 2005
Hi,
Philip Fleischer wrote:
I have done the mfc r2 setup as described in the wiki but
when I try a call
Oct 21 10:02:24 WARNING[14127]: channel.c:1913 ast_request: No channel
type registered for 'UniCall'
Oct 21 10:02:24 NOTICE[14127]: app_dial.c:764 dial_exec: Unable to
create channel of
[EMAIL PROTECTED] napisał(a):
At this moment we work with build 12 (Vers. 1.0.0 - Build 12). We have
other version (build 15), but this firmware don't work fine.
So when I do firmware update, the phone should work?
--
Best regards,
Bartosz Piec
___
I'm trying to use * CVS HEAD. Compilation is done without problem and when I
started * I got these messages
[res_config_odbc.so] = (ODBC Configuration)
res_config_odbc loaded.
[chan_h323.so] = (The NuFone Network's Open H.323 Channel Driver)
== Parsing '/etc/asterisk/h323.conf': Found
==
Igor Briski napisał(a):
U: target
P: password
Thank you both Igor and Karim.
Finally I've found the Telnet Commands Guide document where it is all
described (along with username/password). Thank to Rafael Gonzalez!
This document can be found at:
If a call comes in thru an agent channel (be it from queue or a direct dial
Agent/100) and he puts the call on hold,
there is no MOH, if i dial the extention the agent is logged on to and he puts
the call on hold there is MOH.
Is this solveable? i'm using 1.0.9 at the moment.
Kind regards,
Same here. This is running [EMAIL PROTECTED] which might help
[EMAIL PROTECTED] ~]# amportal start
SETTING FILE PERMISSIONS
Permissions OK
STARTING ASTERISK
/usr/sbin/safe_asterisk: line 42: 12051 Segmentation fault
${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk
How can i change the language in asterisk?
Thanks for all
Lucio
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To
I'm trying to use * CVS HEAD. Compilation is done without problem and when I
started * I got these messages
[res_config_odbc.so] = (ODBC Configuration)
res_config_odbc loaded.
[chan_h323.so] = (The NuFone Network's Open H.323 Channel Driver)
== Parsing '/etc/asterisk/h323.conf':
Anybody got any documentation/experience on the subject?
I'm trying to get it working, but the documentation I have lacks any
information on what should be installed on the server side.
--
Igor Briški - [EMAIL PROTECTED]
___
--Bandwidth and
iHello,
I know this topic has been discussed a lot, but I just wanted to
add in my $0.02 worth about preventing Goiax from being abused.
First, a few things I did which could have raised red flags were
restrictions in place.
1. When I first heard about Goiax, I immediately signed up
Just see this bug:
http://bugs.digium.com/view.php?id=5489
On Fri, 2005-10-21 at 07:23 -0600, Rich Adamson wrote:
I'm trying to use * CVS HEAD. Compilation is done without problem and when
I
started * I got these messages
[res_config_odbc.so] = (ODBC Configuration)
res_config_odbc
Just see this bug:
http://bugs.digium.com/view.php?id=5489
On Fri, 2005-10-21 at 07:23 -0600, Rich Adamson wrote:
I'm trying to use * CVS HEAD. Compilation is done without problem and when
I
started * I got these messages
[res_config_odbc.so] = (ODBC Configuration)
res_config_odbc
This is the information I got from Swissvoice support, I didn't tried yet, but
if it can helps.
How to use an external phone book
IP10S phone supports access to Cisco Phone Book but not all functionalities.
The IP10 uses his own interface to access to the Phone Book.
If you want to connect
Hi,
I have noticed that when a SIP redirect is sent back to Asterisk by a
SIP peer, that Asterisk will (quite appropriately) do a
Dial(LOCAL/redirect-number) in the context of the original callee.
It also forks the CDR, which is excellent. Sadly, under these
circumstances, I need to alter the
yes, its irrelavent what the channels within a channelized T1 do, but
with
a pri is more complicated FWIW forget about PRI in Canada, no one seems to
want to offer it. With channelized you need a drop and insert channelbank,
fxs ports on the channels for extensions, and another T1 out from it
I am looking for a little help. Last night I brought my first * box
into production use. The configuration is as follows:
PRI/VoIPJet --- * --- Nortel
Everything is working except I want to make all local numbers and 911 go
out over the PRI line.
I thought adding these lines to fromnortel
This seems like an over complicated way to solve fraud. I think a small
one time fee (or yearly fee) to sign up will prevent more abuse than
anything.
Anyone who would complain about $10 or $20 to sign up for a lifetime (or
per year or whatever) is a leech if they can't understand WHY there
Hy,
any one of you know a way to enter an extensions in a context as soon as
the phone is hooked up? I would like to find a way to alert the internal
users that a new message is present in voce mail, so as soon as they take
the phone to place a call the system alerts for the presence of new
Jason Becker wrote:
Had to do some digging to find out what you were talking about - I guess
you are referring to the section Using native Asterisk format_mp3 for
Music on Hold* found here:
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf
Some of the comments suggest that
Hi i checkout today the latest cvs but after that i
get coredumps.
Somebody know?
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*** THIS IS A RESEND - I got a deletion notice since I used a naughty
word previously (replaced it with complain :) ) so wasn't sure if it got
out ***
This seems like an over complicated way to solve fraud. I think a small
one time fee (or yearly fee) to sign up will prevent more abuse than
Altus,
Just looking over the voip-info wiki
http://www.voip-info.org/wiki/Asterisk+dimensioning it seems you have hit
the limit of h323.
about 1/3 way down won't be able to run more than 20-25 decent quality
calls
Shawn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi,
Your first post got through just fine. Just FYI.
Jayson.
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To UNSUBSCRIBE
[EMAIL PROTECTED] wrote
It's a free service. It belongs on this list.
Olle is right. Even if it is a free service it does not belong here.
This forum is for any Asterisk related user issues, not some DID issue
of one of a hundred such service providers.
Take it off this list.
Seshu Kanuri
Hello,
Just setup an asterisk server and Amp help me install their
portal and Asterisk. The server is up and all the hardware is
loading. Problem is with incoming calls. All calls go to the first
extension on the system and it still does not ring it goes straight to
voicemail. The
Sorry to do this, but I think I may have gotten lost in the shuffle of other
posts, and this prob is rather troubling.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: Thursday, October 20, 2005 7:11 PM
To: 'Asterisk Users
Kanuri, Seshu (Company IT) wrote:
[EMAIL PROTECTED] wrote
It's a free service. It belongs on this list.
Olle is right. Even if it is a free service it does not belong here.
This forum is for any Asterisk related user issues, not some DID issue
of one of a hundred such service
On Fri, 2005-10-21 at 15:51 +0200, René Enskat [Teamware GmbH] wrote:
Hi i checkout today the latest cvs but after that i get coredumps.
Somebody know?
Yes confirmed. It's pbx_dundi.so that causes it.
If you noload = pbx_dundi.so in modules.conf it'll run.
If you need dundi then...
--
Hello,
Ok, so where does Goiax related traffic belong? Should Goiax have its own
mailing list? I would tend to think so, but lacking such a specific list,
this list is probably a place where many users and potential users will be
subscribed.
Jayson.
- Original Message -
From: Kanuri,
I think:
http://bugs.digium.com/view.php?id=5489
On Fri, 2005-10-21 at 15:51 +0200, René Enskat [Teamware GmbH] wrote:
Hi i checkout today the latest cvs but after that i get coredumps.
Somebody know?
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Hi,
Here are a few more thoughts for ways to cut down on abuse. These pretty
much center on making it more difficult for abusers to create accounts.
Everyday users are also subject to these rules, but they shouldn't really
matter if the purpose behind them is to cut down on abuse which could
I am noticing that Zaptel will sometimes not have enough time to load all
the way at bootup. If load by hand 'service zaptel start' then it loads just
fine. I am running zaptel 1.2.0Beta1 on FC4 2.6.13-1. Since this is
inconsistent I have not found a way around this yet.
-However -
On this
On Fri, 21 Oct 2005 10:25:59 -0400
Paul [EMAIL PROTECTED] wrote:
Kanuri, Seshu (Company IT) wrote:
[EMAIL PROTECTED] wrote
It's a free service. It belongs on this list.
Olle is right. Even if it is a free service it does not
belong here.
This forum is for any Asterisk related
Hi,
Is it required to use an MTP on the Cisco callmanager, when integrating
with asterisk (using h323) ?
I am working on a project where the goal is to interconnect Cisco
Callmanager (version 4) clouds together, using either SIP or IAX between
multiple * servers. Basic setup will be:
PHONE -
If the phones are in the same subnet (I mean, all are in the local lan), try
setting canreinvite=yes in one xlite client and the phone,, and give it a
try+
Hope this help.
Juan.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Tod Detre
Enviado el:
Hi there,
I'm having trouble with queue_log. Whenever somebody makes a
queue call, I always get multiple rows in queue_log- several ENTERQUEUE
records. Then when somebody picks up, I get one CONNECT and several
ABANDONs. Anybody got any idea why this could be? My queues are all set
to ring
On Thursday 20 October 2005 19:18, Tom Tune wrote:
My Cisco 7960g SIP phones share an annoying feature: Anything I plug into
their 2nd Ethernet or PC port loses connection every thirty seconds or so.
I did a ping -t and can see regular drops. I do not have access to Cisco's
tech support
Very late response to this thread, but I have ordered a VoiceBlue 4 port
from the distributor in England last week. I was actually surprised at how
inexpensive it was, we budgeted $5K for this and it came in at about $3K Cdn
including tax and shipping. We are switching over to a GSM network
Scott Wolfe wrote:
I am noticing that Zaptel will sometimes not have enough time to load
all the way at bootup. If load by hand 'service zaptel start' then it
loads just fine. I am running zaptel 1.2.0Beta1 on FC4 2.6.13-1. Since
this is inconsistent I have not found a way around this yet.
On Wed, Oct 19, 2005 at 09:16:48AM -0400, Dave Wise wrote:
I am using a * w/a PRI for the TDM interface to telco.
I am running Asterisk CVS-HEAD-05/29/05-03:59:44
All was working well until I needed a SIP ATA to be unlisted.
in sip.conf, on the account I used:
restrictcid=yes
I am
utils.c at line 194 is broken.
Commenting the following fixed it for me
/*hp-hp.h_addr = hp-buf;
if (inet_pton(AF_INET, host, hp-hp.h_addr) 0)
return hp-hp;
return NULL; */
-Original Message-
hi, did you solve this problem, which i exactly have?
lokotes wrote:
Hi,
Background:
I'm running 2x * boxes.
Box A has a registered user which dials a number. The connection is
sent to Box B which acts as pstn gateway (sangoma 1xE1 card).
Problem:
On Box A before executing Dial() command I
Hello!
I have a question regarding time-based
includes in the dialplan. How are boundary conditions handled? And
is there a definitive, documented procedure for how to handle overlapping
time includes? For example, if I want to have day/night service from
8 A.M. to 5 P.M., there are two ways I
Um, this is a bit of a grey area, isn't it? Don't people post on -users
about Broadvoice or insert VoIP provider here not working as well? There's
really no hard and fast rules defined about what is -users worthy or not,
except:
-If you are talking about code, it belongs on -dev
-If you are
I have a TDM22B (latest rev), Sipura 2000, and Cisco ATA 186. Running
CVS HEAD from about a week ago.
Calls made from a SIP device on either the cisco or sipura work fine.
Call made from an analog phone hooked up to one of the FXS ports on the
TDM22B sound fine, but any DTMF entered after the
Disregard my previous post. I was thinking the uplink to your telco was via
SIP.
Ray
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On Wed, Oct 05, 2005 at 09:14:09PM +0100, Kevin Walsh wrote:
Do you mean something like VoiceMailMain(${CALLERIDNUM})?
Yes, that works nicely. Thank you!
--
Mason Loring Bliss [EMAIL PROTECTED] Cthulhu fhtagn!
http://blisses.org/awake ? sleep : random() 2 ? dream : sleep;
Hi all.
I've just begun learning *, and as my dialplans and macros have
gotten more complex I started wishing for a way to more easily follow
the flow of various Goto, GotoIf, Dial, and etc. commands, especially
when trying to use priority n and labels rather than numbered
priorities.
On Friday 21 Oct 2005 15:26, Jayson Smith wrote:
Hello,
Ok, so where does Goiax related traffic belong? Should Goiax have its own
mailing list? I would tend to think so, but lacking such a specific list,
this list is probably a place where many users and potential users will be
subscribed.
Hi!
Can anyone help a newbie with this problem. Everything
works except my incoming SIP. Get this in the log
Oct 21 14:04:25 VERBOSE[2696]:
10 headers, 0 lines
Oct 21 14:04:25 DEBUG[2696]: # Testing 83.140.41.62 with 192.168.1.0
Oct 21 14:04:25 DEBUG[2696]: Target address
On 10/20/05, Dave Cotton [EMAIL PROTECTED] wrote:
On Thu, 2005-10-20 at 03:31 -0400, [EMAIL PROTECTED] wrote:
SuSE Linux Enterprise Server 9
Asterisk 1.2.0 beta1
I am trying to build and install Asterisk on SuSE. I started with a
fresh full installation of SuSE.
The last lines of
I received some postings back, as I was trying to do the same thing.
it' is a problem with Kernel 2.6... 2.4 works fine .. this is the summary
I got from reading the posts before.
I hope that helps... I dont have the ability to go DOWn in kernel to 2.4..
I'm going to poll the group one more
On Fri, 2005-10-21 at 09:27 -0700, [EMAIL PROTECTED] wrote:
I received some postings back, as I was trying to do the same thing.
it' is a problem with Kernel 2.6... 2.4 works fine .. this is the summary
I got from reading the posts before.
I hope that helps... I dont have the ability to go
Is it required to use an MTP on the Cisco callmanager, when
integrating
with asterisk (using h323) ?
As of CCM 4.X, Cisco indicates that any H.323 trunk that will support
MoH/Transfer/etc need MTP resources. Annoying.
I am working on a project where the goal is to interconnect Cisco
Hi All:
I have a situation to be resolved.
Assume that one location call center with 150 agents.
I have two asterisk servers to serve those 150 sip phones. The servers
are connected to PSTN as 4 T1/PRI for each.
I have a few questions,
Can sip phones login to both servers for the call
When I make a call out form the 1st to GoIAX and into my 2nd box,
I see this Oct 21 13:46:39 NOTICE[4948]: Rejected connect attempt
from 204.13.233.114 which in other words mean that the call comes
in but nothing happens to it.
Not quite. It means YOU rejected it. You need to have a section
oops, typo!
http://www.voip-info.org/wiki/view/Asterisk+dimensioning
-Original Message-
From: Shawn Porter [mailto:[EMAIL PROTECTED]
Sent: Friday, October 21, 2005 10:08 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] how many oh323
Altus,
On Fri, 2005-10-21 at 09:39 -0700, Tielin Xu wrote:
Hi All:
I have a situation to be resolved.
Assume that one location call center with 150 agents.
I have two asterisk servers to serve those 150 sip phones. The servers
are connected to PSTN as 4 T1/PRI for each.
My question is why do you
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