Hello guys,
I have searched high and low, but not found any information about
rules of using DTMF in SIP INFO method. Cisco has described something with
Signal=, but it look like this feature is dependent on implementors?
The problem is chan_sip.c cannot correctly translate received DTMF
digits,
Andrew Joakimsen wrote:
Includes are recursive
Make a context with just all the internal extensions, and then make
contexts for all the outbound calls and another group of contexts just
as you are doing (admin, sales, etc)
Thank you, Just the answer I was looking for!
Ken
Hello,
We are planning to buy following Hardware for Asterisk TestBed. Please let me know if
this seems fine to you.
1. IP Phones ( 5 in number) CISCO 7940/7960, SNOM 200, Pingtel xpressa
2. Wildcard T100P interface card, that will connect Asterisk server to
Our Nortel Switch SL-100
3.
Hi.
The implementation is correct, I can use sip info
method to get all the DMTF, *,# included (eg voicemail
works great with sip info dtmf)
the line atoi(buf) is needed 'cause buf is a char, and
we need a int value to do the comparisons below that line.
and I don't see why they get set to 0
Just for integration, look here
http://lists.digium.com/pipermail/asterisk-users/2003-July/016464.html
basically sip info dtmf are:
Event encoding (decimal)
_
0--90--9
* 10
# 11
A--D
Hi all,
thank U all for your very fast response. I want to clarify that my
question was not regarding about the fasibility of
Voip over satellite in general but especial the behavior of the Asterisk
PBX on a long delay path. We just successfully tested
H323 Voip with a Innovaphone IP
Tarun Banka wrote:
Hello,
We are planning to buy following Hardware for Asterisk TestBed. Please let me know if this seems fine to you.
1. IP Phones ( 5 in number) CISCO 7940/7960, SNOM 200, Pingtel xpressa
2. Wildcard T100P interface card, that will connect Asterisk server to
Our Nortel
Hey..
Has anyone played around with Asterisk on the Itanium2, Opteron or
Athlon64??
Can Asterisk (or Linux for that matter) actually make good use of a
64bit system??
Later..
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64bit is not really needed for Asterisk. (maybe a 4GB+ cdr database...
but you can run that on a seperate platform)
The Opterons' SMP performance might be useful though... cypromis and
others mentioned this before here...
Someone even sent in a patch which made Asterisk compile on a 64bit
unfortunately you can't do outgoing calls with LineJACK. If you have to
place outgoing calls, then buy some FXO VoIP Gateway (Micronet,
Audiocodes) or digium hardware.
Um, why? is the LineJack not a FXO _and_ FXS interface?
I'm curious as to why Quicknet's hardware receives such chilly
Hi All.
When I run modprobe zaptel I get the message that the zaptel.o was
compiled for kernel version 2.4.20-4GB while this kernel version is
2.4.20-4GB-athlon. And fails.
When I run modprobe wcfxo I get the message that the zaptel.o was
compiled for kernel version 2.4.20-4GB
On Sun, Oct 12, 2003 at 09:01:40AM -0400, Andrew Kohlsmith wrote:
unfortunately you can't do outgoing calls with LineJACK. If you have to
place outgoing calls, then buy some FXO VoIP Gateway (Micronet,
Audiocodes) or digium hardware.
Um, why? is the LineJack not a FXO _and_ FXS
I was incorrect in my citation of credit in the below email. Properly the
credit goes to John Todd for the Asterisk config examples. His excellent
article is at:
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=1
Sorry for the goof-up.
Robert
My config that works for number 1
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I need some help figuring out how to solve the following scenario:
If an external call comes in from my provider (src cid 12345678) via chan_zap,
into Asterisk which then is told to forward the call to a different external
number (87654321),
Hello Brancaleoni,
Sunday, October 12, 2003, 4:39:32 PM, you wrote:
BM Hi.
BM The implementation is correct, I can use sip info
BM method to get all the DMTF, *,# included (eg voicemail
BM works great with sip info dtmf)
BM the line atoi(buf) is needed 'cause buf is a char, and
BM we need a
What devices do you plan to use? PSTN line in USA and IP phones in
Nepal? Would this be for one user or a large office?
Regards,
Andrew Joakimsen
Envision Studio
http://envisionstudio.net
888-210-8063
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Has anyone experienced low volume with a X100P FXO card? If so, can you
offer some suggestions to improve the volume and quality of the
connection?
I have adjusted the txrx gains without much success.
Kevin
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Thank you, it helped.
I started * with asterisk -vvvgc and I can see all STDERR
messages
Serge
From: Florian Overkamp [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problems with AGI scripts in Perl and Java
Date: Sat, 11 Oct 2003
This will be for the large office. a call generated by
the office in USA will have to be forwarded to Nepal.
and at first its PSTN line only as VoIP is illegal in
Nepal. we are planning to do VoIP also once it is
declared legal.
--- Andrew Joakimsen [EMAIL PROTECTED]
wrote:
What devices do you
I'd like some feedback on potentially submitting a request (and probably a
patch too) to change the way the AGI command GET DATA works.
Right now, # terminates the entry, which is then returned with the #
stripped off the end. What I'd like is to allow user configurable
termination digits, which
Can a call be kicked out of a queue if it reaches a specific timeout?
I don't see an obvious way to do this in either queues.conf or
extensions.conf any pointers or patches to do this? smile
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http://bugs.digium.com/bug_view_page.php?bug_id=195
hunt the bugs, someone else was doing one also.
James Sizemore wrote:
Can a call be kicked out of a queue if it reaches a specific timeout?
I don't see an obvious way to do this in either queues.conf or
extensions.conf any pointers or
On Fri, 2003-10-10 at 16:50, rnc Info Lists wrote:
I am getting the following messages that seem to be coming from Asterisk.
In the system there are no ZAPTEL cards installed. I did uncomment ztdummy
in the Makefile in /usr/src/zaptel before running make install. Any
ideas on how to get rid
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Tarun
BankaSent: Sunday, October 12, 2003 4:17 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Is this
Hardaware Enough for Asterisk ?
Hello,We are planning to buy following Hardware for
Hi All
I am having this problem when setting up a H323 call.
Can anybody tell me what is going on?
Thanks
Serge
--
NOTICE[245776]: File chan_oh323.c, Line 1293 (oh323_write): H323:1637:
Format changed from 4 to 8.
WARNING[245776]: File frame.c, Line 76 (ast_smoother_feed):
Hi!
I'm considering giving the Grandstream BudgeTone-102 phones a try.
You might also want to take a look at the Swissvoice products
(www.swissvoice.net). As far as I have been able to judge from the web
site those phones offer considerably more for the same price.
Next to that I'd be very
Hi,
I need some help with my sip phones. I have a Xten softphone
and a Budge Tone 101 from Grandstream.
If Im connected from my LAN all is fine but from the
Internet I connect the phone but I dont have the sound.
Asterisk SLI show me this when I try to call my voicemail:
are you using Frame Relay? how big are the packets? I don't think you would
be using ATM over a satellite VSAT modem. Also don't be fooled, satellite
modems don't speak IP. There is normally an IP edge device that makes you
believe it is IP.
The 500 ms delay is the speed of light. You need to
Dream on about the 100 ms or less. Once you get to the satellite, it is the
same time regardless of where you are going on the foot print of the
satellite. Speed of Light does not understand American speed limits. Of
for that matter Europeans. The speed of light is constant. Just pick up
I am not sure I understand the comments but please allow me to simplify.
1) 1xT1 in the T400P goes to the Telco provided T1 connection.
2) 1xT1 in the T400P goes to the Channel Bank.
3) The channel bank breaks up the FXO or FXS analog.
I would suggest you stay away from a Bank Channel to receive
Are you using NAT? Is nat=yes in your
sip.conf? canreinvite=no, reinvite=no ?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Chris Hariga
Sent: Sunday, October
12, 2003 10:42 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No sound
with
I've been trying to use the AGI get_data function for some time now, and
can't get it to work. Today I reinstalled a clean system with Red Hat
8.0 (I had been using RH9, but was told * had problems with RH9) and
downloaded the latest Asterisk CVS to install. I then downloaded and
installed
is
your SIP phone behind a NAT? is* behind a NAT?
Uriel
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Chris
HarigaSent: Sunday, October 12, 2003 10:42 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] No sound
with SIP Phones on the
Ok.. Let me pose a question regarding this configuration.
Lets say you have the ISP bring in a full T1 and they split it half
voice half data. They would usually do this in a channel bank on
site... So in this scenrio... You have the Channel Bank from the ISP
where they split the channels. Then
Ok.. Let me pose a question regarding this configuration.
Lets say you have the ISP bring in a full T1 and they split it half
voice half data. They would usually do this in a channel bank on
site... So in this scenrio... You have the Channel Bank from the ISP
where they split the channels.
Do you r really need more CPU power for Asterisk? I'd think in a
larger system you'd go with multiple servers this would allow for
redundancy
--- WipeOut [EMAIL PROTECTED] wrote:
Hey..
Has anyone played around with Asterisk on the Itanium2, Opteron or
Athlon64??
Can Asterisk (or Linux
Ive read and experienced the echo problems with the
X100P. Is Digium
going to fix the problem or refund our money?
I want to see this work because myself and
other small companies out there use analog lines. I would trade up to T1 but that requires me to
have at least 9 lines. If I did
I have a Cisco 7940
when you call in from outside and dial the Cisco phone extension I get
this
Read_channel ## vpb/1-3: Setting record mode, bridge = 0
WARNING[18451]: File chan_sip.c, Line (sip_write): Asked to
transmit frame
type 8, while native formats is 4 (read/write = 4/4)
==
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