Dear Christian,
Thank you for your
interest, unfortunately your link did not help me, I think I might not
understand what you mean. Do you know if Asterisk and Snom can provide Call
Completion or did you use another server?
We have currently around
125 Snom-190 installed, but are not
i found that here Method is REGISTER
char * pszNonce= dcd98b7102dd2f0e8b11d0f600bfb0c093;
char * pszCNonce = 0a4f113b;
char * pszUser = Mufasa;
char * pszRealm = [EMAIL PROTECTED];
char * pszPass = Circle Of Life;
char * pszAlg = md5;
char szNonceCount[9] =
Listas wrote:
You should use Firefly Third party edition
I'm using it in IAX2 and it works fine
Thanks for the hint. I am still not there ;-(
iax.confincludes:
register = 611:password@voip.elmit.com ;firefly
[611] ; Firefly 3rd party
type=friend
host=dynamic
disallow=all
On 26/12/2004, at 12:42 PM, Jean-Yves Avenard wrote:
Hello
Didn't find any information in the wiki. Regex only refers to the
dialing syntax
Thank you all for your answers, it seems to work in most cases...
However I have something like this:
exten = _8[89]XXX,1,Dial,Zap/4/1414${EXTEN:1}
exten
If I replace the whole lot with:
exten = _88XXX,1,Dial,Zap/4/1414${EXTEN:1}
exten = _88XXX,2,Dial,Zap/3/1414${EXTEN:1}
exten = _88XXX,3,Dial(SIP/jya-home-out/${EXTEN:1},60,r)
exten = _88XXX,4,Dial(IAX2/freshtel/${EXTEN:1},60,r)
exten = _89XXX,1,Dial,Zap/4/1414${EXTEN:1}
I tried to connect my * to IAXtel, but i always get this errors.
chan_iax2.c:5849 socket_read: Registration of 'mnetwork' rejected:
Registration Refused
On dial a iax number i get:
chan_iax2.c:5526 socket_read: Call rejected by 69.73.19.178: No authority
found
chan_iax2.c:5528 socket_read:
Thanks for your input all who have responded.
I would like to set up a SOHO PBX with *, VoIP in, and ISDN phones and
Analogue phones internally. What hardware would people here suggest?
Seth
-Original Message-
From: Dorn Hetzel [mailto:[EMAIL PROTECTED]
Sent: Saturday, December 25,
I tried to connect my * to IAXtel, but i always get this errors.
chan_iax2.c:5849 socket_read: Registration of 'mnetwork' rejected:
Registration Refused
On dial a iax number i get:
chan_iax2.c:5526 socket_read: Call rejected by 69.73.19.178: No authority
found
chan_iax2.c:5528
On Thu, 23 Dec 2004, Jerry Geis wrote:
Did you do a make config in the zaptel source directory?
THat works for me.
Oops! No, didn't do that. Will have a got at it. Thanx!
Remco
Jerry
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Noticed the following on the CLI:
Received HUP signal -- Reloading configs
== Parsing '/etc/asterisk/extconfig.conf': Found
== Parsing '/etc/asterisk/manager.conf': Found
snip
How do I track down the source of the HUP?
There are no indications in any log files and I do not have any
cron
On Friday 10 December 2004 03:41 pm, Eric Wieling aka ManxPower wrote:
Adi Linden wrote:
I admit that this might be some very basic question... How do I obtain
Asterisk 1.0.3 from CVS? Does '-r v1-0' get me 1.0 or 1.0.3?
-r v1-0 will get you the latest 1.0.x CVS. Basically it will get you
Dear All,
Did anybody get success in configuring voice modem for asterisk?
I am having Intel AMR modem by AC'97 and running on as a softmodem.
Thanks
/Harshal
--
There are 10 kinds of people in the world: Those who understand binary
and those who don't...
At 06:24 AM 12/25/04, you wrote:
On Sat, 25 Dec 2004, John Bittner wrote:
Anyone having any problems with CVS-HEAD-12/24/04-15:59:15
and ALERT_INFO
I have a system setup with polycom phones configured to auto
answer on internal calls. When we upgraded to the latest CVS
the auto answer stopped
On Tuesday 21 December 2004 10:36 pm, Tracy R Reed wrote:
I am having a hell of a time with transfers.
First the Snom issues:
The transfer button on the Snom 220 does not work. I have read about
Use the soft button!
--
Steve Szmidt
They that would give up essential liberty for temporary
Any suggestions?
What does your dialplan entry in extensions.conf look like?
Here's one that has been working:
exten =
_1700NXX,1,Dial(IAX2/mnetwork:[EMAIL PROTECTED]/[EMAIL PROTECTED])
I tried yours and at the moment i got this one.
exten = _1700NXX,1,Dial(IAX2/iaxtel/${EXTEN:2})
They have a free version coming out that raises the limit to 8 gig I
believe.
Gary
Does Oracle have a decent-featured free version of their db software? That
was my original point, and where MS SQL 2005 is quite in the lead (limited
only to 1GB of RAM, 4GB DB, and 1 CPU).
-Michael
On 21 Dec, 2004, at 13:54, [EMAIL PROTECTED] wrote:
This is for a small business (restaurant and catering). We want to move from POTS
to VoIP to save on the phone bill. Currently we use four lines for
voice + one fax going into a Lucent Partner system PBX.
Right now I'm considering two
Any suggestions?
What does your dialplan entry in extensions.conf look like?
Here's one that has been working:
exten =
_1700NXX,1,Dial(IAX2/mnetwork:[EMAIL PROTECTED]/[EMAIL PROTECTED])
I tried yours and at the moment i got this one.
exten =
On Tuesday 07 December 2004 04:18 pm, Matt Darnell wrote:
On Tue, 7 Dec 2004 12:58:11 -0500, George Herndon
[EMAIL PROTECTED] wrote:
On Dec 7, 2004, at 8:48 AM, [EMAIL PROTECTED]
wrote:
ken ,
i too have a comdial analog pbx. i'm running a seperate vm system and
would like to
Hi,
I have a problem
with asterisk behind an IX66 router. Outgoing calls are OK at this time, I say
this because I have occasions when outgoing calls fail. For incoming calls I
consistantly get the following error message:-
Dec 26 15:44:10 NOTICE[23533]: chan_sip.c:7183
handle_request:
Hi,
I seem to remember from the TelAppliant HowTo PDF, that you actually
have to create an individual entry in sip.conf for each Number, just as
you do for each SipGate number.
Regards,
Marc
Steve Beaumont wrote:
Hi,
I have a problem with asterisk behind an IX66 router. Outgoing calls are
OK
I have not been able to find anything that relates to this problem. The agents
are using Cisco phones.
Calls goes into a queue. but once an agent picks it up it cannot be
transferred. However if they call directly to the agents extension it's not a
problem transferring calls.
It sounds like
Marc,
I'm not sure what you mean. Are you suggesting that I enter a sip entry,
E.g. [0870xyzabc] for the telappliant provided PSTN number ?
Best regards
Steve B
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Marc Storck
Sent: 26 December 2004 16:12
Just upgrade to most recent CVS and use RealTime
Extensions. Works fantastic and is built-in to asterisk core code.
-Matthew
- Original Message -
From:
Gabriel
Afana
To: asterisk-users@lists.digium.com
Sent: Saturday, December 25, 2004 3:36
AM
Subject:
Hi;
Any idea of how to have different ringing tone on
called party for different caller-id by means of "Alert-Info"
header.
Regards
Mohammad
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The qop method should only be copied from what you receive from the
server. If the server doesn't
send it, don't send it back.
Your pszURI should be the same as the Request-URI
Compare your code with the routine build_reply_digest in chan_sip.c
Kamran Ahmad wrote:
i found that here Method is
Thanks... my problem is solved now. But i don't know why it don't work
befor. :-)
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Rich Adamson
Gesendet: Sonntag, 26. Dezember 2004 16:19
An: Asterisk Users Mailing List - Non-Commercial Discussion
On Sat, 25 Dec 2004 10:43:00 -0500, Dorn Hetzel
[EMAIL PROTECTED] wrote:
On Sat, Dec 25, 2004 at 04:27:52PM +0800, Ronald Wiplinger wrote:
Greg Hill wrote:
I am looking for a small device with four FXO and one WAN connection.
Simple, so that the cleaning woman can make a hardware reset if
On Sat, 25 Dec 2004 17:07:35 +0800, Dinesh Nair [EMAIL PROTECTED]
wrote:
we're evaluating the use of a Lucent APX8100 E3/SS7 to SIP gateway for
use in conjunction with asterisk, serving something like 4000+ lines.
does anyone have experience with the APX8100 and it's integration with
SIP
Brian Capouch [EMAIL PROTECTED] writes:
God, I'm sure everyone on the list must be thinking, Oh, why oh why
didn't *Greg* write Asterisk instead of Mark; he seems so very much
smarter. . .
First: Greg is right, but he's also wrong. Yes, the schema is broken,
from a theoretical point of
On Sun, Dec 26, 2004 at 12:16:24PM -0600, James Taylor wrote:
On Sat, 25 Dec 2004 10:43:00 -0500, Dorn Hetzel
[EMAIL PROTECTED] wrote:
On Sat, Dec 25, 2004 at 04:27:52PM +0800, Ronald Wiplinger wrote:
Greg Hill wrote:
I am looking for a small device with four FXO and one WAN connection.
Tom Ivar Helbekkmo wrote:
However, this attitude that anything Mark does must be right, and can
not be questioned, is seen disconcertingly often in this forum. It is
counter-productive, and should be discouraged.
My intent was to tweak Greg a bit for what seemed to be condescending
attitudes
I had the same problem with snom 190 phones.
Using the transfer with # instead of Transfer Button on the phone worked
for me.
In my configuration REFER was not send, so the transfer with the button on
the phone did not work.
Guido Hecken
-Ursprüngliche Nachricht-
Von: steve szmidt
On Sat, Dec 25, 2004 at 11:12:22PM -0500, Dorn Hetzel wrote:
I'd like to get VM_CALLERID to include number in addition to name
since often when calls come from cell lines or various other,
the name is just a city, state and the number would be more
usefull. Is there a way to get the number in
On Sun, 26 Dec 2004 14:17:59 -0500, Dorn Hetzel
[EMAIL PROTECTED] wrote:
On Sun, Dec 26, 2004 at 12:16:24PM -0600, James Taylor wrote:
On Sat, 25 Dec 2004 10:43:00 -0500, Dorn Hetzel
[EMAIL PROTECTED] wrote:
On Sat, Dec 25, 2004 at 04:27:52PM +0800, Ronald Wiplinger wrote:
Greg Hill wrote:
I
On Sun, 26 Dec 2004 14:17:59 -0500, Dorn Hetzel
[EMAIL PROTECTED] wrote:
On Sun, Dec 26, 2004 at 12:16:24PM -0600, James Taylor wrote:
On Sat, 25 Dec 2004 10:43:00 -0500, Dorn Hetzel
[EMAIL PROTECTED] wrote:
On Sat, Dec 25, 2004 at 04:27:52PM +0800, Ronald Wiplinger wrote:
Greg Hill wrote:
I
Anyone know why this
is happenning at asterisk start up. I understand that this ahs something to do
with using external SQL database for asterisk configuration. I have not worried
too much about this side of asterisk to date as I have been strugling with the
configuration from plain text
On Sun, Dec 26, 2004 at 02:22:43PM -0600, James Taylor wrote:
On Sun, 26 Dec 2004 14:17:59 -0500, Dorn Hetzel
[EMAIL PROTECTED] wrote:
On Sun, Dec 26, 2004 at 12:16:24PM -0600, James Taylor wrote:
On Sat, 25 Dec 2004 10:43:00 -0500, Dorn Hetzel
[EMAIL PROTECTED] wrote:
On Sat, Dec 25,
Dorn Hetzel wrote:
(a) there are definitely analog DID implementations out there.
not saying they're pretty, but they exist...
(b) are you really sure it's cheaper with only 4 channels to
do a T1? including local loop?
As far as I know, Asterisk/Zaptel does not support analog DID
Dorn Hetzel wrote:
On Sat, Dec 25, 2004 at 11:12:22PM -0500, Dorn Hetzel wrote:
I'd like to get VM_CALLERID to include number in addition to name
since often when calls come from cell lines or various other,
the name is just a city, state and the number would be more
usefull. Is there a way to
On Sun, 26 Dec 2004 14:26:48 -0600, Eric Wieling aka ManxPower
[EMAIL PROTECTED] wrote:
Dorn Hetzel wrote:
(a) there are definitely analog DID implementations out there.
not saying they're pretty, but they exist...
(b) are you really sure it's cheaper with only 4 channels to
do
Eric Wieling aka ManxPower wrote:
I seem to recall a bug regarding this. Are you using 1.0.3, 1.0.x CVS
STABLE, or CVS-HEAD. The problem is that what was listed in the
voicemail.conf.sample as the default e-mail message was, in fact, not
the default e-mail message. Uncommenting out the
Hi All,
I have the following scenario, it may already have been
answered elsewhere, but I cant find the solution.
I already have a PBX and would like to start implementing
asterisk. I have ordered a 4 port card from the asterisk store (2 port FXS and
2 port FXO) and am waiting for it
I am not sure if it is the right list for the post.
Please excuse my lack of expertise, if it is a bad
post.
Is there anyway to detect the originating network
identity of the call in Asterisk? For example, if the
Asterisk gets a call from Cingular Network, is there
anyway to find out that the
I just checked the code and didn't see any way to change the pager email
body or subject like you can the regular email notification - only the from
string. If you wanted to hack it, the messages are in apps/app_voicemail.c
on or about line 1033.
Steve
Dorn Hetzel wrote:
Now the regular
Hello,
The asGUIclient suite has a predictive dialer component to it(VICIDIAL) and
it can function well on multiple Asterisk servers at once using a single
MySQL server backend. It performs on par with several mid-level commercial
dialers that we have compared it to(Nobel, TripleP, DataTel,
focus on npa-nxx (area code-prefix)
if the call is coming from a non-ported number, then
http://telcodata.us/docs/queries.html may help you --
see the example files..
there are also a couple other sites out there.. but i've
found this one to be my favorite thus far.
-m
On Sun, 26 Dec 2004, oi
Regards,
Michael Di Martino
Director of MIS
The telx Group
Office: 212 480 3300 X.2022
Cell: 646 207 6603
[EMAIL PROTECTED]
--
Sent from my BlackBerry Wireless Handheld
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sip phones
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I have read on the wiki that deadlocking is a problem when using the Manager API.
I have the manager api running. I am making hundreds of telnet connections from a remote server (running windows) to the asterisk server via telnet / manager api.The remote machine is telnetting in to the manager
mattf wrote:
Hello,
The asGUIclient suite has a predictive dialer component to it(VICIDIAL) and
it can function well on multiple Asterisk servers at once using a single
MySQL server backend. It performs on par with several mid-level commercial
dialers that we have compared it to(Nobel, TripleP,
On December 26, 2004 07:40 pm, Michael Di Martino wrote:
Regards,
Michael Di Martino
Director of MIS
The telx Group
Office: 212 480 3300 X.2022
Cell: 646 207 6603
[EMAIL PROTECTED]
--
Sent from my BlackBerry Wireless Handheld
We're impressed. Really we are.
Thanks mattf I'll try/test VICIDIAL this monday ;)
[OT]
Steve ... It 's used on VERY agressive telemarketing
campaings, PDialer isn't only anonying call, you can show
the CID as you want, in order to be called by
your customers ... with * you can manage your CID
as you want if you have IAX or
That's good to get a general idea, but number portability only tells you
which carrier has the block. It does not let you know about specific
numbers :-{
Lyle
- Original Message -
From: Matt Klein [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
http://www.illuminet.com/docs/lidb/
-
I believe there are more instances of the abridgment
of the freedom of the people by gradual and silent
encroachments of those in power than by violent and
sudden usurpations. - James Madison
On
More specifically, see the data sheet about lidb:
http://www.verisign.com/stellent/groups/public/documents/data_sheet/001944.pdf
You could go that route, or get a switch, or... there's a variety of other
options. But if you're looking for a full number lookup, you're looking
for lidb access..
Yes, it can be used to make annoying telemarketing calls. But in our case,
we created it initially to use to call back our client base for reminder
calls on system maintenance. Previously we had just given agents a list of
clients to call each day manually, which was very inefficient and didn't
This was pretty much fixed several months ago, but on a heavily loaded
system you can get manager API pauses(output just stops and then floods
out all at once) of upto 20 seconds if your Asterisk system is under very
heavy load. Even in those cases the system will not deadlock, so the manager
API
Ronald Wiplinger wrote:
I reloaded my asterisk and found some red lines flushing by. When I
stopped it I see:
WARNING[21481]: cahn_zap.c:9773 setup_zap: Ignoring signalling
WARNING[21481]: cahn_zap.c:9773 setup_zap: Ignoring echocancelwhenbridge
WARNING[21481]: cahn_zap.c:9773 setup_zap:
Something of an update...
At the recommendation of a consultant I called in last week, I've now
switched to an AVM Fritz! PCI card, using CAPI. At this stage I'm only
using the one card so I don't need the patches to run multiple cards
yet.
Upon loading the modules (capi/capifs and fcpci), I
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