Hello list,
Does anybody managed to reboot GrandStream phone with sip notify
sip_notify.conf section peer
It seems that I need to send a sys-control Event but i suspect that's not
enaugh my phone just answer me a CSeq: 102 NOTIFY.
Cheers
Laurent
hello
any one please tell me if there is a way to define a
range of users in sip.conf
suppose i want to create 1000 user from 500 to
5000999 with no password from
thanks
Kamran
Start your day with Yahoo! - make it your
Hi Guys,We have just set up Asterisk 1.0.7 with (CVS Head) for a realtime enviorment using MySQL Asterisk Addons.I have populated the sip_buddies table with the same information that is came from our
sip.conf, however registration seems to fail for the softphone we have set up.Does anyone have
hi
i gues the error is in this line
include/asterisk/strings.h:232: parse error before
`va_list'
can anyone help me please. how can i fix
this?
much thnks.
chris.
- Original Message -
From:
chris
To: asterisk-users@lists.digium.com
Sent: Tuesday, July 26,
in iax.conf devi anche mettere questa riga per ogni fwd:
register = FWDNumber:[EMAIL PROTECTED]
Bruno.
kswail wrote:
Hello,
I am trying to setup my Asterisk box to accept calls from the FWD network.
I've followed all the config advice / samples I've found on the web.
Making calls to
Hello,I
have few questions about Asterisk.I installed Asterisk from CVS on
FreeBSD and I made cvsup 2 days ago.1.I couldn't find Asterisk version
using "asterisk -V" command.How can I to find version
information?2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV
E/F (FXS)onit.I
On Fri, Aug 05, 2005 at 01:17:05PM +0800, 163 wrote:
Thanks a lot for you help first.
I tried to load the drivers, but failed.
[EMAIL PROTECTED] voicepet-single-x100p]# /sbin/modprobe zaptel
modprobe: Can't locate module
Jason ha scritto:
Could someone assist me in configuring this phone. It is saying in
the CLI that its registered and saying its capabilities are recieved
but i got no dialtone on the phone. Thanks
are you using chan_skinny or chan_sccp?
Sergio
Well thanks Stefan, for the help but when I am executing the AGI script I am
getting the errors as below:
Aug 5, 2005 3:29:44 AM net.sf.asterisk.util.impl.JavaLoggingLog info
INFO: Received connection.
Aug 5, 2005 3:29:45 AM net.sf.asterisk.util.impl.JavaLoggingLog error
SEVERE: Unable to create
Hi,
I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm
having problems with Caller ID. I have run clidtest, and it seems happy
enough, returning:-
server clidtest # ./clidtest /dev/zap/1
Number: 041222, Name: MOBILE
(that number's fake.) However, I'm not getting
What do I put in voicemail.conf to let me send another user a voicemail
from inside Comedian? I've CVS-HEAD, and the instructions are a bit
ambiguous on the voicemaill.conf.sample. Advanced option 5 is the only on I
don't have, and a very important one to have, indeed.
Chris Coulthurst
Hello !
I 'd like to connect Cisco IP phones to *. (7940 7960)
Shall I use SIP or SCCP. Which approach provides better support
of features of the Cisco IP phones ?
Thanks !
Johann
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Yes it does apply. Near the top of your sip.cfg file, you should have lines
like this:
alertInfo voIpProt.SIP.alertInfo.1.value=ring-answer
voIpProt.SIP.alertInfo.1.class=4/
alertInfo voIpProt.SIP.alertInfo.2.value=internal
voIpProt.SIP.alertInfo.2.class=5/
alertInfo
Well thanks Stefan, for the help but when I am executing the AGI script I
am getting the errors as below:
If you want to retrieve sip peers from Asterisk you won't do this via an
AGI as I explained. You will just run the main() method of the Manager
class I sent you in my last mail as an
Hi Guys,
We have just set up Asterisk 1.0.7 with (CVS Head) for a realtime
enviorment using MySQL Asterisk Addons.
I have populated the sip_buddies table with the same information
that is came from our sip.conf, however registration seems to fail for
the
softphone we have set up.
Does
Please stop asking the same questions over and over.
On Monday 25 Jul 2005 02:46, Balgansuren.B wrote:
Hello,
I have few questions about Asterisk.
I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago.
1.I couldn't find Asterisk version using asterisk -V command.
How can I
Johann Steinwendtner wrote:
Hello !
I 'd like to connect Cisco IP phones to *. (7940 7960)
Shall I use SIP or SCCP. Which approach provides better support
of features of the Cisco IP phones ?
SIP will cost you an extra $100 per phone to license the SIP software.
But the SIP has been
I have configured /etc/asterisk/zapata.conf, but
now Asterisk refuses to start:
Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:5976
mkintf: Unable to get parameters
Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:9478
setup_zap: Unable to register channel '1-15'
Aug 5 10:47:29
I think that you are wrong:
Here is the /proc/zaptel/1 file(seems to be correct, and log
Messages too indicates the initialization
is correct.):
Span 1: WCT1/0
"Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS/CRC4 RED
1 WCT1/0/1 Clear
If you see your /proc/zaptel/1 you will see a RED
Hi
I already have an swtich stmt in my extensions.conf
switch=Realtime/[EMAIL PROTECTED]
Even i tried with one you send, but same error.
please if you are using realtime do me a favour by sending all configuration you are using.
Thanks
vinod malani
On 8/5/05, yusuf [EMAIL PROTECTED] wrote:
Hi
what is the upload speed on B?
Looks to me as you have bandwidth problem!
Martin Kronstad wrote:
Hi!
Problem:
I can’t hear what the people at Location B i saying, they hear me but I
do not hear them. They can call, I can call. Just no sound.
My current setup is:
Hello!
I use FastAGI very frequently [meaning 30 channels IVR is made in it]
and sometimes I find, that there is a message like:
Jul 29 09:38:55 VERBOSE[896] logger.c: == Auto fallthrough, channel
'Local/[EMAIL PROTECTED],2' status is 'CHANUNAVAIL'
Jul 29 09:38:55 VERBOSE[893] logger.c:
Hi there,
this is my queues.conf, I'm using todays CVS:
[599]
joinempty = yes
musiconhold = default
strategy = roundrobin
servicelevel = 60
wrapuptime = 0
maxlen = 0
timeout=15
announce-frequency = 15
member = SIP/381
member = SIP/300
At first call 381 rings, if you hang up and call again you
You might have better luck posting this question on Asterisk-Dev (on
how to disable checksum etc).
On 8/5/05, Jon Whitear [EMAIL PROTECTED] wrote:
Hi,
I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm
having problems with Caller ID. I have run clidtest, and it seems
Hello there,
I'm new to PHP AGIs and I'm having problems with a particular script
that has a include_once statement on it. If I remove that stament,
the script runs until the section of the code that depends on the
include and then returns. If I include that statement, the script does
not
I don't post here often but I read with interest all the postings. - I've
been on a lot of mailing lists, but this one is by far the most interesting.
I've been doing a lot of work with 'tftp' loading Cisco 79xx phones with
firmware, configs. for asterisk, etc.
And I see where a
IPManager - Asterisk Configuration Tool has been released with IPSwitchBoard
Overview
IPManager is a configuration tool for Asterisk. It gives you an easy way of
configuring Asterisk to perform maintenance and creation of the following:
. SIP Extensions can be configured very easy with
On Friday 05 August 2005 14:04, Leo Burd wrote:
Hello there,
I'm new to PHP AGIs and I'm having problems with a particular script
that has a include_once statement on it. If I remove that stament,
the script runs until the section of the code that depends on the
include and then returns.
Has anyone had any luck getting USB ISDN devices to work with Asterisk? I have
bought a DayTrek miniVigor 128 and would like to get it to work with CAPI or
mISDN. Has anyone every successfully done something like this?
Thanks,
Christoph
___
is it possible to register asterisk in a sip proxy as
if it were a terminal (like a cisco ATA)? how?
Thanx
Jenna ;)
___
1GB gratis, Antivirus y Antispam
Correo Yahoo!, el mejor correo web del mundo
I've done this using SPA-2000, SPA-2000 can generate polarity
reversal signal, The pay-phone detects call answer and hangup by
revesal signal.
also the pay-phone must be supported polarity reversal detection.
Anyone got any suggestions? I need to know what piece of hardware I
I'd like to officially reclaim the Features in a GUI thread ;)
Asterisk Hackers, Admins, and general digital phreakers of old,
After careful consideration, the ARTCP project will probably have to be
split into two major sections, both distros or at least maps for a system to
be designed as well
There is error correction in TFTP. Its done at the application layer and not
the transport layer.
TFTP uses two UDP ports for control and data transfer, this is probably
where there are problems with NAT devices.
The control connection is ;
client - sport dynamic(x) - server dport 69
client
Outta intersted - why mysql?
If postgresql not a better option?
I would happily contribute to postgres work (and am indeed starting to work
on something similar atm, schemas written, etc) - but at the end of the day
mysql still does not cut it inmo.
No offence to mysql developers, etc.
Side note, I've jumped to a different name, as ARTCP defines more the
control program portion than an entire distro.
ARTP -Now- AstCD (Asterisk Complete Distribution)
Obviously the name would change to something a little more memorable once
the project is in a release phase
Sherwood
I personally prefer MySQL-MAX. I curently run *RT in a large production
environment comprised of more than 1K users, with MySQL-MAX as my backend.
Also, it's a point of I've spent so much time working with MySQL that I
don't want to have to jump systems. It's fit the needs of the VOIP provider
I
On Friday 05 August 2005 09:11, Chris Thompson wrote:
Outta intersted - why mysql?
If postgresql not a better option?
This is an old argument which works both ways just fine.
--
List Manager
Network Voice Communications, Inc.
netwvcom.com
___
Has anyone been able
to successfully integrate the Avaya AUDIX voicemail system with Asterisk?
I have been
diligently investigating converting our small (Ontario, Canada) office to
Asterisk, and ditching our Avaya PBX. However, our head office (New
Jersey, USA) maintains our AUDIX
I have an Asterisk serving 15 people using the X-Lite soft-phone.
Currently they all register to the internal IP address of Asterisk
(192.168.1.110). I only use VoIP internally. External calls go PSTN.
I'd like to arrange it so that they register to our external WAN
address (port forwarded to
Hello,
I am using asterisk with a HFC-Card which is connected to the internal S0 of a
Siemens Hi Path 3000. When asterisk receives a TEI check request an active
call to the PSTN ends. Does someone know this problem? I tried
bri-stuff.0.1.0-RC4a, bristuff-0.2.0-RC8h and bristuff-0.2.0-RC8m.
vinod malani wrote:
Hi Guys,
We have just set up Asterisk 1.0.7 with (CVS Head) for a realtime enviorment using
MySQL Asterisk Addons.
1.0.7 is NOT CVS HEAD!
1.0.7 is STABLE and RealTime doesn't work on STABLE!
-Matthew
___
Asterisk-Users
Yes you can.
In sip.conf you must edit:
register = user in SIP proxy:password in SIP proxy:AUTH-ID in SIP
proxy@IP of SIP proxy/local peer in asterisk where you answer the call
and you must define a peer for the SIP proxy:
[SIP-proxy]
type=peer
context=where you have the peer for answer
Help is on the way:)
This is quite simple to achieve on Sipura units. There is a parameter
called
Dial Tone: [EMAIL PROTECTED],[EMAIL PROTECTED];10(*/0/1+2)
It defines the frequencies and duration of the tone. The 10 you see
near the end is the duration. Simply change it to
Background:
We are currently implementing an Asterisk based solution for
a customer to enable teleworker phone access. We have connected an Asterisk box
running SUSE 9.3 with an AVM Fritz PCI ISDN card installed running CAPI to a Tenovis
box. Softphones using SIP (referred to as SIP
Hi Stefan,
I have all the necessary files for the code to be executed. The
fastagi-mapping.properties file is also correct. But still I am getting the
error for
No script configured for agi://
The IP address is correct and as well as the agi file name. Does it make a
difference
Hi!
The bandwith is not the problem, uploadspeed is about
400 kbits.
I think I found the solution, I need to have a Proxy in
the middle, or set up a IAX2 client and server at each end
I will be testng this next week.
BR Martin Kronstad
What is the
upload speed on B?
Looks
Thanks Greg,
Not too much skill yet. I will be
doing first time.
What is cheapest available in cisco
or polycon.
Any other company that is a little cheaper.
Varun
- Original Message -
From: [EMAIL PROTECTED]
Date: Friday, August 5, 2005 10:34 am
Subject: RE:
Hard phones.
Varun
- Original Message -
From: Jason Walker [EMAIL PROTECTED]
Date: Friday, August 5, 2005 10:35 am
Subject: RE: [Asterisk-Users] ip phones
Soft phones or hard phones?
There are many free VOIP soft phones out there.
-Original Message-
McQuiggan, Mark xt46480 wrote:
Has anyone been able to successfully integrate the Avaya AUDIX
voicemail system with Asterisk?
Haven't tried it, but sounds doable.
At worst case, I would like our Asterisk users to be able to bounce to
an AUDIX mailbox for voicemail storage. At best, I
its kind of difficult to say if we dont know what the included php script has.
i think that the wrap function that Christoph propouse it may work for
debuggin purposes, but i dont think it will solve the problem. Until
you tell us, or show us, the content of the scripts we will be doing
our best
agree with all written below - additionally use php -l to lint/check the
syntax of the file (and the include)
if needed - do a include_once 'bleh.php || die some message;
to see if thats an issue.
my $0.02
- Original Message -
From: Moises Silva [EMAIL PROTECTED]
To: Asterisk Users
-- Goto
(macro-record-enable,s,12) -- Executing DBget("Local/[EMAIL PROTECTED],2",
"RecEnable=RECORD-IN/8521") in new stack -- DBget:
varname=RecEnable, family=RECORD-IN, key=8521 -- DBget:
Value not found in database. -- Executing SetVar("Local/[EMAIL PROTECTED],
Hello List!
I have a problem that has been posted to the list more than once, but so far
I have not been able to find a solution searching the archives and Google.
The problem is with Asterisk audio files not being played to the x-lite
client.
I have an out-of-the-box [EMAIL PROTECTED]
I am using Asterisk CVS from last week and have been using Realtime SIP
for a couple weeks now without any problems. Yesterday I decided to turn on
Realtime IAX but I am having problems dialing to my long distance providers
like Voicepulse, Sixtel or Nufone. I get the following:
--
The key seems to be listing the 10 digit
extensions dialplan in a context other than the context they are defined in in
sip.conf, correct?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dbruce
Sent: Thursday, August 04, 2005
6:55 PM
To: Asterisk
Users Mailing
Hi,
We have just done an upgrade now when ever the
console displays a single line as below
Zaptel: Master Changed to TE4/0/1
Zaptel: Master Changed to TE4/0/2
Zaptel: Master Changed to TE4/0/1
Zaptel: Master Changed to TE4/0/1
The asterisk r
Show alls the lines been Hung Up
[ New Beta Version - Beta Extended ]
A new version of IAX Phone Pro Beta is available. A few bugs have been
fixed and the beta has been extended until October 12, 2005 (the date of
AstriCon 2005). You can download either a new install (be sure to
un-install the old version) or just a new
In file included from include/asterisk/utils.h:26,
from term.c:32:
include/asterisk/strings.h:232: parse error before `va_list'
include/asterisk/strings.h:232: warning: function declaration isn't a
prototype
make: *** [term.o] Error 1
pls advise on how i can fix this,
I'm seeing all sorts of problems and it's probably more of my lack
of experience than anything else. I have a BT100 running 1.0.6.7
code. When I go to the status page it says it's not registered
(hmm, that's not good). I also can't get dial tone but I can dial!
I'm afraid I'm lost any good
Hi,
Sherwood McGowan wrote:
I personally prefer MySQL-MAX. I curently run *RT in a large production
environment comprised of more than 1K users, with MySQL-MAX as my backend.
Also, it's a point of I've spent so much time working with MySQL that I
don't want to have to jump systems. It's fit the
Ok I can register with BV fine (as far as I can tell from asterisk -
see below). I am able to make outgoing calls but all incoming calls
get a fast busy.
I have opened and forwarded the following ports to my pbx:
5060-5063 UDP + TCP
69 UDP (BV claims they need this)
1-2 UDP
I tried
We do extension by extension is our dialing plan because we have a wildcard at
the end trapping all unused extensions and playing a this extension is not in
use message and forwarding users into our IVR. It depends on individual
circumstances which works better. We have 300 DIDs for our sip
Hi all,
When I was making calls from an IP phone, through a X100P, to PSTN,
the following error was encountered.
-- Executing Dial(SIP/25086937-aa6c, Zap/1/91713545) in new stack
-- Called 1/91713545
-- Zap/1-1 answered SIP/25086937-aa6c
Aug 6 00:12:53 WARNING[3983]: chan_zap.c:4717
I'll do what I can. This is all I can say about it. We haven't even had the
first meeting of contributors yet, but I'm sure we will do what we can. The
idea is that this is STABLE, and since I don't use PostgreSQL at all, I'm
sticking with what I know to be sure of stability.
I'll take the
thanx for the reply.
i tried it, and now asterisk is doing something.
but the problem is that instead of sendind a
REGISTER message to the SIP PROXY, it is sendind an
OPTIONS
message, and the PROXY responds with 404 NOT FOUND
ihave in my sip.conf file:
register = 7771::[EMAIL
I am looking for a recommendation on either a Compaq/HP or IBM server for a
100 user Asterisk Server. Unfortunately because of customer constraints I
cannot go with Supermicro, etc.
Syed Akbar
Alico Systems Inc
www.alicosystems.com
Tel: 562-436-1510
VoIP Newbie wrote:
Hi all,
When I was making calls from an IP phone, through a X100P, to PSTN,
the following error was encountered.
-- Executing Dial(SIP/25086937-aa6c, Zap/1/91713545) in new stack
-- Called 1/91713545
-- Zap/1-1 answered SIP/25086937-aa6c
Aug 6 00:12:53
On Aug 5, 2005, at 11:20 AM, Tarpo, Louie wrote:
We do extension by extension is our dialing plan because we have a
wildcard at the end trapping all unused extensions and playing a
this extension is not in use message and forwarding users into
our IVR. It depends on individual
Jenna Cole wrote:
thanx for the reply.
i tried it, and now asterisk is doing something.
but the problem is that instead of sendind a
REGISTER message to the SIP PROXY, it is sendind an
OPTIONS
message, and the PROXY responds with 404 NOT FOUND
ihave in my sip.conf file:
register =
Probably complaining about the dialed number. You say you are dialing
the pstn - and I assume in north america.
What is the number 91713545 supposed to dial? Last time I checked
pstn calls were either 7 or 10/11 digits.
perhaps you forgot to strip the 9 off?
Perhaps the pstn is returning an
Hi list:
I have a client that needs to connect a Asterisk PBX with a TE110P
of Digium and one Nortel Option 11.
Actually the Nortel Option 11 have a AMI E1 card. With it the have
problems of clock sync.
They can change the AMI CARD to a PRI CARD, te questions are:
1) Which
Hello:
I have problems sending dtmf signal to an ATA186 my configuration is:
ATA186 -- asterisk -- ATA186 -- FXS to FXO Converter -- PSTN
The ATA186 are set to send dtmf RFC2833, but it seems that the ATA186 can't
generate dtmf so I can dial to a PSTN number.
Is there a setting that can fix
The next question is, was your call successful? I see you dialed an 8
digit number. Is that what's required on your line?
MARK.
Eric Wieling aka ManxPower wrote:
VoIP Newbie wrote:
Hi all,
When I was making calls from an IP phone, through a X100P, to PSTN,
the following error was
if i remove that line, asterisk stop sendind the
OPTIONS message to the SIP PROXY, but it's still NOT
sending the REGISTER message.
i would alse need to register more than one number
--- Eric Wieling aka ManxPower [EMAIL PROTECTED]
escribió:
Jenna Cole wrote:
thanx for the reply.
i tried
Switch to IAXCOMM and use an IAX extension. Problem
solved.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin
KronstadSent: Friday, August 05, 2005 7:03 AMTo: 'Asterisk
Users Mailing List - Non-Commercial Discussion'Subject:
[Asterisk-Users] Asterisk -
On Thu, 4 Aug 2005, Joseph Rothstein wrote:
Background:
We are currently implementing an Asterisk based solution for a customer to
enable teleworker phone access. We have connected an Asterisk box running
SUSE 9.3 with an AVM Fritz PCI ISDN card installed running CAPI to a Tenovis
box.
Why does the system have to be based on a linux distro? I think
that's the wrong way to go. It's one thing to create a linux distro
around a popular piece of software, but it's another to create
software that can only be used as an entire linux distribution.
If I were you I would take an
I'm testing my asterisk system and the realtime backend. My Asterisk
build is rather aged, 03/18/2005 CVS. I have successfully moved Sip
peers and Voicemail boxes to the realtime database backend and this
works very well except for MWI. I don't seem to be able to get MWI to
work when I store the
??? i dont understand.
On 8/5/05, Siegel, Joerg [EMAIL PROTECTED] wrote:
Ich bin am 9.8. wieder im Hause!
Mit freundlichen Grüßen,
Jörg Siegel.
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Hi,
I've read through
the archives, and wanted to get an updated opinion on the Uniden UIP200
phone. Seems like there were a lot of opinions that it was a good phone,
but there were a few items that people were waiting for firmware updates for,
but that was in 2004.
I'm going to be using
Hello
I have a Grandstream GXP2000 with latest
firmware. When I use it holding the handpiece I don't hear any echo -
neither does other end. However, if I use it handsfree, the other end
notices echo when they speak - ie their voice is echoy. I hear their voice
being a bit echoy.
Is this
On Fri, 5 Aug 2005, Angus Comber wrote:
Hello
I have a Grandstream GXP2000 with latest firmware. When I use it holding the
handpiece I don't hear any echo - neither does other end. However, if I use it
handsfree, the other end notices echo when they speak - ie their voice is
echoy. I
This known as is 'acoustic echo' or 'room reverb' and involves mathematics that
is quite a bit different from that used when cancelling regular 'reflected
electrical signal' echos, as the signal is being acousically distorted as it
echos around the room. On many handsfree handsets it doesn't
How does one go about programming a Cisco 7914 sidecar to be used as a
busy lamp field?
Thanks
Craig Bruenderman
Network Advocates, Inc.
300 Envoy Circle
Suite 300
Louisville, KY 40299
Main: 502-412-1050
DID: 502-992-5929
Fax: 502-412-1058
Mobile: 502-548-1100
On Fri, 5 Aug 2005, Craig Bruenderman wrote:
How does one go about programming a Cisco 7914 sidecar to be used as a
busy lamp field?
This can be done with SCCP only. CHeck the wiki.
Thanks
Craig Bruenderman
Network Advocates, Inc.
300 Envoy Circle
Suite 300
Louisville, KY 40299
Main:
Andres wrote:
Help is on the way:)
This is quite simple to achieve on Sipura units. There is a
parameter called
Dial Tone: [EMAIL PROTECTED],[EMAIL PROTECTED];10(*/0/1+2)
It defines the frequencies and duration of the tone. The 10 you
see near the end is the duration. Simply change
how many channels using codec g729 can be done by an
internet bandwidth to 512kb dedicated.
__
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___
I have a pair of snom 360s at a customer and they were giving me Low Memory
errors. The distributor suggested updating the firmware. I did that, to the
one just below 4.0 (which wasn't released yet). One of the phones is still
giving the Low Memory error every 3-4 days. The other one had a
how many channels using codec g729 can be used by an
internet bandwidth to 512kb dedicated.
__
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Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
___
keep approx. 32kb per channel..
-Original Message-
From: [EMAIL PROTECTED]
Sent: Fri, 5 Aug 2005 11:16:32 -0700 (PDT)
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] how may channels
how many channels using codec g729 can be done by an
internet bandwidth to 512kb
The reason the system is going to be a linux distro is because it will be a
complete out of the box asterisk system ready to be installed. Just like
[EMAIL PROTECTED], only much much more integrated and having more features.
As far as Linux not being a popular server platform? Maybe I missed
how many 'register =' I can have in
sip.conf___
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To UNSUBSCRIBE or update options visit:
Hi everyone.
I need to get CallerID to route incoming calls, but i keep getting this
on the CLI for the callerid
=
-- Starting simple switch on 'Zap/1-1'
Aug 5 13:18:50 ERROR[2756]: callerid.c:260 callerid_feed: fsk_serie made
mylen 0 (-85)
Aug 5
Tim P wrote:
[EMAIL PROTECTED]:mypass:[EMAIL PROTECTED]/2068660133
You need to add the number to the back so you can route it with asterisk.
Ok I can register with BV fine (as far as I can tell from asterisk -
see below). I am able to make outgoing calls but all incoming calls
get a fast
My E1 has 10 lines from my telco, 10 lines are blocked and 20 are idle.
I guess those 10 blocked are my lines(channels). Also reading this:
http://voip-info.org/tiki-index.php?page=Asterisk+MFC+R2
i come to this lines:
...
cas=110-124:1101
The 4 characters after the colon in the cas
Kris Boutilier wrote:
This known as is 'acoustic echo' or 'room reverb' and involves mathematics that is quite a bit different from that used when cancelling regular 'reflected electrical signal' echos, as the signal is being acousically distorted as it echos around the room. On many handsfree
http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption
On 8/5/05, Innocent Evil [EMAIL PROTECTED] wrote:
keep approx. 32kb per channel..
-Original Message-
From: [EMAIL PROTECTED]
Sent: Fri, 5 Aug 2005 11:16:32 -0700 (PDT)
To: asterisk-users@lists.digium.com
On Fri, 2005-08-05 at 14:09 -0400, Craig Bruenderman wrote:
How does one go about programming a Cisco 7914 sidecar to be used as a
busy lamp field?
In the sccp.conf file,
o As a Line:
You can assign a line to the button/lamp which is really neat.
The lamp is green when you are on the line,
Robbie:
I fought with echocancel and various parameters for a long time with
little luck. Then I uncommented AGGRESSIVE_SUPPRESSOR and DISABLED
the Fax/tone detection in in zconfig.h since we're not faxing via
Asterisk. Recompiled and all echo disappeared.
Hope that helps.
-Rob
--
We have experienced some Snom firmware issues, although the are not
related to the symptoms you describe. We found that the sidecards will
not power on unless the 360 host phone is running the latest firmware rev.
Cory Andrews
Purchasing / EVP
VOIPSupply.com
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