= NULL;
break;
*/
Best Regards,
Miroslav Nachev
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How can catch Ringing Event that is coming from ISDN channels and
then to play rining media in case that PSTN is not sending such media.
The same question for the other states like busy, etc.
Any help or suggestions?
Best Regards,
Miroslav Nachev
in Caller ID, but the phone (GXP-2000) must understand and
recognize this.
Best Regards,
Miroslav Nachev
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to the original codec for each party of conference group. So, if you
would like to to conference with software coding you will need of very
power computer.
Best Regards,
Miroslav Nachev
S Thank you all for your explanations related to my question.
S I have one follow-up question though.
S
and 64 simultaneous channels;
- PCI with 64, 128 and 256 simultaneous channels.
In the future the device will support Audio and Video Codecs and
Windows OS.
Best Regards,
Miroslav Nachev
S Hi All,
S Is it correct to say that by design, asterisk wont make use of any cards
S hardware
Hi,
MK ... 2) Buy 4 Carrier Access Channel Banks for $100 ...
From where can I buy Carrier Access Channel Banks for $100? Any URL
or related info?
Best Regards,
Miroslav Nachev
MK Shoval,
MK Interesting Mention. I agree, most people don't have CO exp. And I
MK wish daily I had
degree C
Humidity:5-95% RH, non-condensing
It is possible to be added 3rd Ethernet port for DMZ or other
purposes.
Best Regards,
Miroslav Nachev
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).
Jumper selectable to choose the source: power
connector (jack) or WAN port.
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Miroslav Nachev
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will be between $400
and $1000. Using this MMT you can use 50 MHz CPU for Asterisk.
--
Best regards,
Miroslavmailto:[EMAIL PROTECTED]
Thursday, November 25, 2004, 7:14:50 PM, you wrote:
SL On Nov 25, 2004, at 7:58 AM, Miroslav Nachev wrote:
Hi,
To clarify Xscale, I mean
Dear Bartosz,
Try this: http://www.junghanns.net/asterisk/page17.html
quadBRI PCI ISDN EUR 600,-
Best Regards,
Miroslav Nachev
BJ Hello,
BJ I am looking for 4 port ISDN BRI card.
BJ I have checked wiki and found one, but they do not show prices
BJ for that card. Can somebody
Dear Juan,
We have some success using H.323 as common protocol (OpenH323), but
there are some problems with Fax (T.38) and Multiply Codecs (when you
set more than 1 codec. For example G.729 and G.711).
Best Regards,
Miroslav Nachev
JVG Has somebody ever tried to integrate
:
[EMAIL PROTECTED]
Maybe this is because of point 3.
Best Regards,
Miroslav Nachev
ac hi all
ac as we have brought sample to test in singapore and
ac have start deployment , if you need may be can email
ac to me.
ac regards
ac alex chua
ac --- Miroslav Nachev
comparing with the 1.0 version where is string. When we replace
callerid variable with cid.cid_num the problem was solved.
Best Regards,
Miroslav Nachev
m Hi all,
m I'm trying to compile asterisk-oh323-0.6.3b but I got some comiling errors
m just on start. Can someone tell me the steps
/res_features.c - ast_park_call
We need of this events for new CDR Prepaid systems.
Best Regards,
Miroslav Nachev
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. Also they required min. 10 pcs for ordering.
Best Regards,
Miroslav Nachev
BoAML On Sun, 24 Oct 2004 18:11:44 -0600, Joseph
BoAML [EMAIL PROTECTED] wrote:
Has anybody tested any gateways from ACT
BoAML Last time those gateways came up in a conversation it was concluded
BoAML that efforts
. The prices for regual (big) quantities is $60 per port.
Best Regards,
Miroslav Nachev
BoAML On Mon, 25 Oct 2004 12:37:33 +0200, Miroslav Nachev
BoAML [EMAIL PROTECTED] wrote:
Unfortunately the Mediatrix products are very expensive.
BoAML just one example. my point
Dear Alex,
From where you found this device for $165? I found that the List
Price of this device is $220. Can you send me the URL or some
contacts?
Best Regards,
Miroslav Nachev
AB Hi,
AB We choose the Mediatrix 2102 with 2 analogue and 2 ethernet ports.
AB Cost: £89.99 (roughly
Hello Joseph,
I tested their products and the Phones are OK. They are better than
GrandStream. The Gateways schould be improved.
--
Best regards,
Miroslavmailto:[EMAIL PROTECTED]
Monday, October 25, 2004, 3:11:44 AM, you wrote:
J Has anybody tested any gateways
Dear Olle,
I can say that Emil Ivov has very good knowledge on IPv6 too. You
can use it.
Best Regards,
Miroslav Nachev
OEJ Marc Blanchet wrote:
- no asterisk does not work over ipv6.
- ipv6 port won't be as easy as I would like to...
- I'm currently working on it. Had
Hi,
Is there any experience with Yoda VoIP Devices Asterisk?
Best Regards,
Miroslav Nachev
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Miroslav Nachev
Miroslav Nachev wrote:
SU and exactly how does that get the FAX into the T.38 channel? :-\
Using G.711 or implementing T.38 in Asterisk or adjusting Asterisk
to OpenH323 T.38. From our expirience Asterisk detect that the line is
with Fax data. The problem is what next
Dear Steve,
I can't understand from your mail can I use SpanDSP or not?
Today we try this fax-modem:
http://www.openh323.org/t38.html
The problem now is that we can't start it with HylaFAX.
Best Regards,
Miroslav Nachev
SU Hi Miroslav,
SU It sounds like you don't
Hi,
We try to send Fax through IP Network but without success. The
other party use NetCentrex SoftSwitch and our communication protocol
between us is H.323 (OpenH323). The error that the other party receive
is: bearer capability not imoplemented.
Is it possible to send Fax using
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Pedro Howat
Rodrigues
Sent: 19 October 2004 15:53
To: Miroslav Nachev; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Fax over IP doesn't works
Hi ,
I tried this a lot
Hello Steve,
There is another way to send fax if you use Fax Machine on some FXS
port.
--
Best regards,
Miroslavmailto:[EMAIL PROTECTED]
Tuesday, October 19, 2004, 7:50:52 PM, you wrote:
SU So what changes with T.38? You still need spandsp to interwork with the
Hello Steve,
In the project OpenH323 the T.38 is supported. The easyest way is to
correct Asterisk logic with OpenH323.
--
Best regards,
Miroslavmailto:[EMAIL PROTECTED]
Tuesday, October 19, 2004, 7:38:19 PM, you wrote:
SU Michael Loftis wrote:
Just my $0.02
Hi,
How can I get the Fax Status of transmited document - complete,
error, etc.?
Regards,
Miro.
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Hi,
I would like to make a simple application with address book which
to dial the numbers and to transfer the call to the caller before the
called party is answered. How can I do that?
Regards,
Miro.
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Hi,
http://www.costcentral.com/searchresult.php?keyword=PAP2searchin=1
Mfg Part # Stock Price
-- -- --
PAP2 No $49.86
PAP2-NA Yes $49.76
Best regards,
Miroslavmailto:[EMAIL PROTECTED]
Sunday,
?
Best Regards,
Miroslav Nachev
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Hi,
I am looking for GSM to BRI ISDN Gateway. Any help?
--
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Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel:(+359-2) 983-32-62
Mobile: (+359-88) 897-31-95
E-Mail: [EMAIL PROTECTED]
[EMAIL PROTECTED]
http://www.space-comm.com
Post address:
P. O. Box 941
Hi,
Can I use HT-486 as VoIP Gateway together with Asterisk?
Are there any success experiences?
--
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Dear Pavel,
Go to http://www.voip-info.org/wiki-Asterisk
or search in google
Best Regards,
Miroslav Nachev
Hello guys,
I'm new to asterisk and I have some problems -
would you help me please.
I installed following configuration: Linux server
with Asterisk + Digium X100P
everytime.
Can you give me some ideas/suggestions how to solve this case?
Best Regards,
Miroslav Nachev
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Hi,
C18 I suggest you go the channel bank route.
Can you be more detailed? Any URL? What is this and how to do it?
On Wed, 18 Aug 2004 10:16:01 +0200
Miroslav Nachev [EMAIL PROTECTED] wrote:
Hi,
We have a case where we need of 16 x FXS, 12 x FXO and
1 x E1. To
do this using
OK, but there are just FXS ports. What about FXO ports?
Miroslav Nachev wrote:
Hi,
C18 I suggest you go the channel bank route.
Can you be more detailed? Any URL? What is this and how to do it?
You can start by looking at the WiKi pages:
http://www.voip-info.org/wiki-Asterisk
Hi,
We try to start TDMoE but the result is that the Asterisk and the
Network are crashed.
Are there some successful stories with TDMoE? Any help will be very
useful.
Best Regards,
Miroslav Nachev
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Hi,
We try to start TDMoE but the result is that the Asterisk and the
Network are crashed.
Are there some successful stories with TDMoE? Any help will be very
useful.
Best Regards,
Miroslav Nachev
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Hi,
We try to start DTMoE but the result is that the Asterisk and the
Network are crashed.
Are there some successful stories with DTMoE? Any help will be very
useful.
Best Regards,
Miroslav Nachev
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Hi,
It is good that the SER is used for SIP Proxy. But in this case how
to use the PBX capabilities of Asterisk like IVR, VoiceMail, DialPlan,
and etc.?
Best Regards,
Miroslav Nachev
--- Kurtz [EMAIL PROTECTED] wrote:
Why is it that the wiki indirectly recommends SER
is with DSP channels for coding and transcoding and
the 2nd board is with FXO and FXS ports.
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel:(+359-2) 983-32-62
Mobile: (+359-88) 897-31-95
E-Mail: [EMAIL PROTECTED]
[EMAIL PROTECTED]
http
Hi,
In the old Asterisk CVS we need of ZAPTEL and ZAPATA sources. Now
in the Asterisk Download instructions ZAPATA is missing:
http://www.asterisk.org/index.php?menu=download
Do we need more of ZAPATA?
What is the role of ZAPATA?
Best Regards,
Miroslav Nachev
COSMOS
.
How can I start using all DTMF features using Cisco Phone?
Best Regards,
Miroslav Nachev
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PM, you wrote:
NG Hello,
NG On Wed, 2004-08-04 at 04:35, Miroslav Nachev wrote:
Hi,
When we use BudgeTone where the DTMF is set to via RTP (RFC2833)
all the DTMF functionality of Asterisk is working OK. When use Cisco
7960 the transfer is working OK, but when I try to remote pick-up
Hi,
I start the fax capabilities of Asterisk, but I don't know how to
detect that the sent fax status - complete, error, etc.
Any ideas?
Best Regards,
Miroslav Nachev
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Hi,
I have MicroNet VoIP Gateway SP5014 with 2 x FXO, 2 x FXS
Ethernet ports. One friend say that the Config Menu of this GW is very
similar ot WellTech Config Menu.
How to start this GW together with Asterisk?
Best Regards,
Miroslav Nachev
Dear Chris,
We are interesting of this and would like to work together with
you.
Best Regards,
Miroslav Nachev
While I was away on vacation, buried deeply in another thread (New
Asterisk bounty: SIP simultaneous), Olle E. Johansson raised a question
which is close to my heart
Dear Yoram,
This question is not for this discussion. The right places are
Users or/and Dev.
Which kind of hardware PCI Card or Standalone Gateway?
By the way your hardware is compliant with the following main
requirements, then your hardware can work with Asterisk:
1. Registrar
.
Please give me some suggestions.
Is there any regular price?
--
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel:(+359-2) 983-32-62
Mobile: (+359-88) 897-31-95
E-Mail: [EMAIL PROTECTED]
[EMAIL PROTECTED]
http://www.space-comm.com
Post address:
P. O
Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel:(+359-2) 983-32-62
Mobile: (+359-88) 897-31-95
E-Mail: [EMAIL PROTECTED]
[EMAIL PROTECTED]
http://www.space-comm.com
Post address:
P. O. Box 941,
1000 Sofia,
Bulgaria
Office address:
ap. 9, fl. 4,
11
gets blocked. This can be done through the tones coming
from the PTSN, which in this case, are:
Frequency : 425 Hz
Level : 10 dBm0
Signal (ms)/Pause (ms): 320, 320
Max. Duration : 30 s
--
Best Regards,
Miroslav Nachev
COSMOS Software
:
SIP User ID:
Authenticate ID:
Authenticate Password:
NAT Traversal:
Yes, STUN server is:
Use NAT IP: 193.200.15.141
--
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel:(+359-2) 983-32-62
Mobile: (+359-88) 897-31-95
E-Mail: [EMAIL
Hi,
Is there any program for ZAPTEL FXO with which I can debug the
signals that are coming from PSTN (Tones, Voltages, Ampers, etc.)?
In case that I have to do this program which is the closest entry
point of the ZAPTEL software?
Best Regards,
Miroslav Nachev
COSMOS
files, which could
help us tracing the actions and seeing which action is completed and
which not.
Seeing the actions sequence will help us to establish and solve the
problem we have. We count on your help for the solution of this
problem.
Best Regards,
Miroslav Nachev
COSMOS Software
mailto:[EMAIL PROTECTED]
Thursday, July 1, 2004, 6:28:52 PM, you wrote:
SC On Thu, 2004-07-01 at 11:00, Miroslav Nachev wrote:
Hi,
We have our own algorithm handling (dial plan) the calls and
different events. When we receive an external call (from FXO
,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel:(+359-2) 983-32-62
Mobile: (+359-88) 897-31-95
E-Mail: [EMAIL PROTECTED]
[EMAIL PROTECTED]
http://www.space-comm.com
Post address:
P. O. Box 941,
1000 Sofia,
Bulgaria
Office address:
ap
.
How can I do that?
Also is it possible the manner of dialing plan to be different
depending of the caller using Caller ID?
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel:(+359-2) 983-32-62
Mobile: (+359-88) 897-31-95
E-Mail: [EMAIL PROTECTED
,
Miroslav Nachev
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Hi,
I am looking for Call Generator for PRI ISDN and BRI ISDN signals.
From where I can found some cheap or 2nd hand call generator
(tester/analyzer)? Maybe PCI based will be cheaper than standalone
solution.
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd
Dear Michael,
MD why not use asterisk with QaudBRI and/or E100P ?
Because I have to be sure that I am Euro ISDN compliant. My target
is Bulgaria which is in Europe.
Best Regards,
Miroslav Nachev
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf
Hi,
I can't find anywhere on the Asterisk web the license terms for
commercial use of Asterisk software. Do I have to pay something
(and how much) if I want to use the Asterisk in our IP PBX solutions?
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel
I have problems with answering of FXO when FXS line is
open
; interfaces for internal analog phones
signalling=fxo_ks
threewaycalling=yes
; interfaces for external PSTN line
signalling=fxs_ks
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises
Hi,
I am interesting is there any way to use Cisco DSP Modules with
Linux?
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel:(+359-2) 983-32-62
Mobile: (+359-88) 897-31-95
E-Mail: [EMAIL PROTECTED]
[EMAIL PROTECTED]
http
to take
(redirect) the call and answer to it using my SIP telephone. Could you
give me any ideas how to set up this configuration - how can I catch
the incoming call.
Thank you in advance.
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel:(+359-2) 983-32-62
telephones.
Is there any ready solution for this case we could use and how much
it will cost?
Thank you in advance.
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel:(+359-2) 983-32-62
Mobile: (+359-88) 897-31-95
E-Mail: [EMAIL PROTECTED
Dear Scott,
The idea is to be used new SIP phones instead legacy phones. With
this the network cables (UTP/FTP-5) will be used instead one cable
network for the Computers and other cable network for the Phones.
Best Regards,
Miroslav Nachev
If there is already an existing phone
Hi,
Is it possible to use SER (www.iptel.org) toghether with Asterisk?
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel:(+359-2) 983-32-62
Mobile: (+359-88) 897-31-95
E-Mail: [EMAIL PROTECTED]
[EMAIL PROTECTED]
http://www.space
Hi,
I would like to find some way for hardware coding instead software
(using the Host CPU). Are there any PCI boards just with codecs (DSP)
or other way?
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel:(+359-2) 983-32-62
Mobile: (+359-88) 897
Hi,
I have Debian Linux with kernel 2.6.6. The all packages compiled
except ZAPTEL where I have the following error:
voipgw:/usr/src/zaptel# make
make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory `/usr/src/linux-2.6.6'
CC [M] /usr/src/zaptel/zaptel.o
directory `/usr/src/linux-2.6.6'
make: *** [linux26] Error 2
--
Best regards,
Miroslavmailto:[EMAIL PROTECTED]
Wednesday, June 2, 2004, 6:17:12 PM, you wrote:
FB On Wed, 2004-06-02 at 14:56, Miroslav Nachev wrote:
I have Debian Linux with kernel 2.6.6. The all
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