Hello Tony!
Many thanks for your valuable comments! I will write back results when I
will have some ;) This box is under preparation for production, I was
making low-level tests and some kernel/system tunings. Hopefully will
have some practical experiences (with this box) soon.
Thanks again!
On Mon, Jan 16, 2006 at 09:00:28PM -0500, Carlos Alperin wrote:
Another request make me test my t1 card, which has no quality problems, but
all that I get is:
[EMAIL PROTECTED] zaptel-1.2.1]# ./zttest
Opened pseudo zap interface, measuring accuracy...
99.987793% 99.987793%
On Monday 16 January 2006 09:02, [EMAIL PROTECTED] wrote:
i tried to setup realtime voicemail recently with 1.2.1
but couldn't get it to work. no matter what i do. it still
looks for config in the voicemail.conf file. (BTW realtime
sip extensions works fine)
here's the voicemail line in
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Take a look at this:
http://bugs.digium.com/view.php?id=0006256
I opened this bug!
Thanks
Mimmus
___
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Hello,
I am writing a program based on Astersik Manager which needs to put calls on
hold and to redirect them to others extensions.
I haven't funded any action able to do this.
Is there a way to put calls on hold using Asterisk Manager Actions?
Amaury
Yes,
But I'm reading on the opposite side, or always I loose 1 sample?
[EMAIL PROTECTED] zaptel-1.2.1]# ./zttest -v
Opened pseudo zap interface, measuring accuracy...
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample
Hello.
I am in a strange situation. I have two asterisk. Asterisk A makes a
call for asterisk B by IAX. Asterisk B recives the call and delivers
it to Openser by SIP. The problem is openser printing this in the screen:
ERROR: parse_to : unexpected char [] in status 5: David sip: .
Did someone made or implement an script to
move all the info from the standar cdr format to a PostgreSQL or MySQL?
Were going to start moving all the
last two years of cdr soon, and really I never take this point on my schedules.
Thanks,
Carlos Alperin
From:
[EMAIL
Carlos Alperin a écrit :
Did someone made or implement an script to move all the info from the
standar cdr format to a PostgreSQL or MySQL?
Here's what I did:
To create a CDR table:
create table cdr (
accountcode varchar (30) NOT NULL,
src varchar(64),
dst
Is there a key sequence to stop a call as its ringing, before the call is
answered?
The * key stops a call after it is answered, but I'd like a way to cancel the
call during the ringing phase.
/Obelix
This message was sent
Jean-Michel Hiver a écrit :
Carlos Alperin a écrit :
Did someone made or implement an script to move all the info from the
standar cdr format to a PostgreSQL or MySQL?
Here's what I did:
Actually let me give credit where is due: I didn't *do* this, I probably
found it somewhere on the
As you can see on your link, the reference on the voip-info web site is for
Centos 4.1
I recommend you to compile asterisk instead of running the rpm install. That
is going to give you a more accurate idea of what is going on with the
modules that you need to be able to run asterisk.
All that
Hi Karsten,
I have the same problem. MusicOnHold sounds awful. The PRE-1e does not have this
problem. I have two identical systems (hard-/software). One system has the
problem the other does not. I thought i could be timing problem or interrupt
conflict. But we could not find out the problem.
Merci, Jean databases are not my speciality.
This save me a lot to read.
Regards,
Carlos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: Tuesday, January 17, 2006 4:52 AM
To: Asterisk Users Mailing List - Non-Commercial
Hi Armin,
thx for your feedback, but what do you mean with Did
you load the card with config for DID on that port?
I have loaded the modules with:
modprobe capi
modprobe kernelcapi
modprobe divacapi
modprobe divas
and then loaded divactrl like this:
divactrl load -f ETSI
I suppose that this
Hi everybodyI'm trying to make Asterisk 1.2.1 run under VMWARE and Suse
9.2.I use ZTDUMMY module for timing and ZTTEST gets an average precision of
98,4 %.Is there any way to improve it?Best regardsMauro
Zanin
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build_tools/make_version_h
include/asterisk/version.h.tmpif cmp -s include/asterisk/version.h.tmp
include/asterisk/version.h ; then echo; else
\ mv
include/asterisk/version.h.tmp include/asterisk/version.h ; \fi
rm -f include/asterisk/version.h.tmpif cmp -s
.cleancount .lastclean ; then echo
On Tue, 17 Jan 2006, richard Coco wrote:
Hi Armin,
thx for your feedback, but what do you mean with Did
you load the card with config for DID on that port?
I have loaded the modules with:
modprobe capi
modprobe kernelcapi
modprobe divacapi
modprobe divas
and then loaded divactrl
Louis-David and list,
We were having the exact same issue using mpg123: distorted sound, clicks,
etc.
First, we are using Gentoo, and didn't realise the ztdummy.ko module wasn't
loaded by default. This is required for the timing.
mpg123 still refused to work correctly (sounds like 2-bit audio
Hi,
we are trying to setup a prototype Asterisk machine for our call center
(15-20 users).
We are encountering some difficulties in finding a 'good' softphone
(SIP/IAX).
Suggestion/experience?
Is there some product available for Windows with modifiable code?
Is there some freelance developer
Matt,
Today, I'm working on a proposal for 150 seats PBX replacement.
Competitors are using Aastra Matra, Alcatel or Cisco IPBX.
Do you think I could name some of those call centers (those 100 using
vicidial) to prove Asterisk is a safe choice ?
If positive, is there a way I could get in touch
We are looking for SIP trunks for our * pbx for our business. Being able
to port our numbers is an absolute requirement. teliax can do it, but I am
unsure of the others.
Anyone have experiences (good, bad) with the above mentioned providers to
share? Eg reliability, quality, etc.
-Dan
[EMAIL PROTECTED] wrote:
On Monday 16 January 2006 09:02, [EMAIL PROTECTED] wrote:
Put in voicemail.conf searchcontexts=yes
and do not forget to stop and start *.
Reload may not do.
benchev
That's not a solution, but just a workaround.
1.2.1 has a bug where it always uses an empty context
Hello,
I can give you some contacts off-list, and of course you can use my
company as a reference as well. I don't have any contacts in France,
but I do in Spain and Greece if that is OK.
We have a basic French translation of the web clients with no finished
images, but we do have fully
Point taken.
At $1300 per month it really isn't worth it.
PaulH
- Original Message -
From: Tim Litwiller [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 17, 2006 4:41 PM
Subject: Re: [Asterisk-Users] I
TITLE:
Cisco IP Phones SYN Flood Device Reload Vulnerability
SECUNIA ADVISORY ID:
SA18479
VERIFY ADVISORY:
http://secunia.com/advisories/18479/
CRITICAL:
Less critical
IMPACT:
DoS
WHERE:
From local network
OPERATING SYSTEM:
Cisco IP Phone 7900 Series
On Sunday 15 January 2006 12:23, Kerry Garrison wrote:
I have an install with the Digium TDM2400 with the EC module and even
though Digium techs have spent well over 10 hours tweaking and tweaking the
call quality is so bad we are ready to chuck it. I think that you were on
Is this FXS or FXO
I want to create a VoIP solution to allow many members of a closed community
to talk to each other (one on one) via soft phones. In many ways, what I
want is not unlike Skype, except that it would allow for relative anonymity
and be open only to a select group. The system should support as many as
Hi All,
I am a newbie to VOIP and after some problems I was able to install Asterisk. If
I start Asterisk I could find Asterisk Ready at the end and I am thinking
that Asterisk is started successfully. Later after changing my Extensions.conf
and ser.conf nothing works, I could still see the
Paul Klipp a écrit :
I want to create a VoIP solution to allow many members of a closed community
to talk to each other (one on one) via soft phones. In many ways, what I
want is not unlike Skype, except that it would allow for relative anonymity
and be open only to a select group. The system
On Tue, 2006-01-17 at 14:46 +0100, Paul Klipp wrote:
I want to create a VoIP solution to allow many members of a closed community
to talk to each other (one on one) via soft phones. In many ways, what I
want is not unlike Skype, except that it would allow for relative anonymity
and be open
Hi Manoj,
you have to configure a IAX endpoint while you have sent a SIP
configuration, with no phone defined. Maybe you could send IAX.CONF so we
could se what is the issue...
Ciao
Mauro
___
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Pisac wrote:
But I'm dissapointed with all this minor needless problematic
changes which needlessly spending my time. I will realy double rethink
in the future about upgrading any tuned system to new Asterisk release.
It was _never_ documented that you could skip a numeric parameter for
Dan - try Avaya 4610/4620 (WML) and Alcatel iptouch (XML) as well.
Phil Menico
XTEND Communications
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, January 16, 2006 7:55 PM
To: Asterisk-Users@lists.digium.com
Subject:
Hi all,
I am going to be building an Asterisk system to replace the current
aging (aged) Nortel Meridian system in a travel agency. There is
already a voice T-1 in place and currently there are about 20 extensions
in use. I would want to move up to about 25 extensions immediately and
about
[EMAIL PROTECTED] wrote:
Hi All,
I am a newbie and trying to install Asterisk from instructions given
in http://www.voip-info.org/tiki-index.php?page=Asterisk+RPM. We have
Centos 3.3 so
I downloaded rpm's from
ftp://ftp.linuxsys.com/pub/LSE/packages/CentOS-3.4/asterisk-1.0.9/
and tried
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
All,
- 1 call transfer -
Call comes in for uip200. can trasnfer it just fine.
- 2 call transfer -
Call comes in - then a second call comes in
I can longer transfer either call? I can toggle between them but not
transfer.
Does
Steven Ringwald wrote:
On Mon, 2006-01-16 at 22:30 +0100, Tamas wrote:
/var/log/dmesg:
...
CPU: L1 I Cache: 64K (64 bytes/line), D cache 64K (64 bytes/line)
CPU: L2 Cache: 512K (64 bytes/line)
mtrr: v2.0 (20020519)
CPU: AMD Athlon(tm) 64 Processor 3000+ stepping 02
Using IO-APIC 2
i have not checked your configs, but Destination Unreachable is a
message out of the scope of Asterisk. Please check your network
configuration, first you should be able to ping the asterisk box from
the client and viceversa.
Regards
On 1/17/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi
its funny, please tell us where we can see your sip.conf and the
relevant extensions.conf to see how are you registering and dialing.
Regards
On 1/17/06, david.castro [EMAIL PROTECTED] wrote:
Hello.
I am in a strange situation. I have two asterisk. Asterisk A makes a
call for asterisk B by
You could keep your phones if you used a Norstar SIP Handset Gateway from
http://www.citel.com/products/handset_gateways/
Each handset gateway allows you to convert up to 24 Norstar handsets from the
proprietary Norstar signalling protocol to SIP.
Other handset types are also supported.
You are right, only outgoing calls!
I found lines that you mentioned, but I do not understand where is
difference? In current chan_zap.c I read:
if (!IS_DIGITAL(ast-transfercapability)) {
set_actual_gain(p-subs[SUB_REAL].zfd, 0, p-rxgain, p-txgain, p-law);
} else {
You could probably save some money by building the server from scratch
rather than buying a dell. I would at least buy a 2 port T-1 card, just
cause you're better off keeping only one card in the system and its only
a few hundred dollars more for the 2 port. This will make it easier if
you
yes, but the UA should have compatible codecs, otherwise Asterisk will
stay doing transcoding (a lot of CPU usage). Also remember that doing
re invites imply that the two UA should be able to communicate each
other without help, this turns complicated when both are behind NAT.
Regards
On
Hi all,
we use OpenSER together with Asterisk.
All SIP users registers with OpenSER and asterisk is doing the voicemail
thing.
We use the Asterisk RealtimeArchitecture for voicemail users and SIP peers.
The database table for the sip peers is a view from the OpenSER subscriber
table.
The MWI
Tzafrir Cohen wrote:
On Tue, Jan 17, 2006 at 03:48:44PM +0100, Kib Eki wrote:
If I use a rawplayer like this:
#!/bin/sh
while(true) do
for name in $@; do
cat $name ;
done
done
BTW: 'while(true)' is is csh syntax that accidentally works in sh. In sh
it spawns a subshell for the true.
BTW:[2]
Voipjet can not port your number. Nor will they respond to requests
for problems to be fixed in a timely manner.
On 1/17/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
We are looking for SIP trunks for our * pbx for our business. Being able
to port our numbers is an absolute requirement.
Does anyone know where the order of the wires on this connector can be found?
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
I'm using both Teliax and VoipJet for my home asterisk (low call volume).Teliax had been 100% reliable yet.With VoipJet I had some down time, delay problem... But they improve there service over the time.As VoipJet is cheaper it is my default long distance carrier. When I got problem with VoipJet
Has anyone gotten announce-holdtime in queues.conf to work? Doesn't seem to
matter what combination of options I use, I can't get this particular setting
to do what the docs say.
Thanks.
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
I have been having problems dialing into Verizon conferencing using my *
system.
If i dial using a POTS line directly, the dtmf codes for the conference
room are recognized with no issues, however, verizon doesnt recognize the
keys when i press them as being a valid code.
I remember something a
Hi,
I answer to my own posting...
On Sun, Jan 15 2006 Karsten Wemheuer wrote:
Hi,
I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When
I activate music-on-hold on a SIP-to-SIP connection, the music sounds
like in a fast-forward play mode. On the *-console I can see
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Douglas Garstang
Has anyone gotten announce-holdtime in queues.conf to work?
Doesn't seem to matter what combination of options I use, I
can't get this particular setting to do what the docs say.
Hi All
I have some grandstream phones registered to my asterisk and all internal,
external, voicemail services etc are working very well.
I am not sure that it is a problem more so an annoyance. If someone dials my
extension number or external DDI while I am already in a call rather than
When using asterisk real time, every time somehting
occurs in asterisk it goes to the DB. If the DB isnt
up natrually it dosent know what to do. So yes this
behavior is perfectly normal.
Dovid
(Sorry about the spelling mistakes)
--- Dov Bigio [EMAIL PROTECTED] wrote:
Hello,
After 2 weeks and
VoipJet has been great to me for dial time.
Nufone.net is where I get my inward dialing for my VoIP. Also good
experience so far.
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, January 17, 2006 4:45 AM
Hi,
emails to astricon.net seems to bounce (at least for me)
I need information about proper authorized Asterisk training in the
Miami, FL area and the possibility of later DCAP testing.
Thanks,
--
---
Erick Perez
Linux User 376588
http://counter.li.org/
When I read variables in AGI scripts, I see only the follwing 13 variables
agi_request
agi_channel
agi_language
agi_type
agi_uniqueid
agi_callerid
agi_dnid
agi_rdnis
agi_context
agi_extension
agi_priority
agi_enhanced
agi_accountcode
beside these, I found following variables documented on
in Sip.conf make sure to add NAT=yes. Also I reccomend
reading the new book that came out. (Dont have the URL
if some one else can please post it). The book will
help you learn the basics of asterisk and also answer
many of your questions. Also check out the wiki.
Dovid
--- Ever Zalazar [EMAIL
Hello!I am new to Asterisk, AMP, Linux...did I say all?.. I just installed Asterisk, and for my needs it is working great. In my AMP I see the nwebmail but I can´t get into it. When I place my login and password, comes with the following message: "An internal error has occured. Please
Ok.. but I don't use Real Time at all.
I just use cdr_mysql. It would be smarter if it simply ignored MySQL outages
or at least just logged, but without stopping.
Regards
dov
- Original Message -
From: Dovid Bender [EMAIL PROTECTED]
To: Dov Bigio [EMAIL PROTECTED]; Asterisk Users
When making a call through voicepulse, I can hear one ring,
then the ring tone changes slightly and it continues forever.
I think the first ring actually goes trough. If I hang up and
try again it works normally. Any clues?
Regards,
David Koski
[EMAIL PROTECTED]
In extconfig.conf I have:
voicemail = odbc,asterisk,voicemail_users
sipusers = odbc,asterisk,sip_users
sippeers = odbc,asterisk,sip_users
Asterisk version is 1.2.1
When asterisks starts, I don't saw any SQL queries in my mysql log.
First, when a user calls his own mailbox, I saw a sql querie
Is call waiting enabled on your extension(s)? Mine behaves this way, but
that's how I want it. *70 enables and *71 disables (I think). I think call
waiting is disabled by default, so someone would have had to enable it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hello,
Please try olle on [EMAIL PROTECTED] (or look for steve sokol's email.)
Zoa
Erick Perez wrote:
Hi,
emails to astricon.net seems to bounce (at least for me)
I need information about proper authorized Asterisk training in the
Miami, FL area and the possibility of later DCAP testing.
Hello.
Asterisk A is version 1.2.1.
Asterisk B is version 1.0.9.
If I call by IAX from Asterisk A to B, and after that, Asterisk B
call by SIP to Openser, the call works.
The invite message from Asterisk to openser by Sip is:
U 2006/01/17 17:50:49.261265 10.2.11.50:5061 - 10.2.11.50:5060
Hi,
is there anybody having a copy of an old DCAP test just to take a look?
TIA
Giorgio Incantalupo
Erick Perez wrote:
Hi,
emails to astricon.net seems to bounce (at least for me)
I need information about proper authorized Asterisk training in the
Miami, FL area and the possibility of later
On Tue, 2006-01-17 at 09:44 -0700, Wiley Siler wrote:
VoipJet has been great to me for dial time.
For me too. I'm using it right now to terminate all my calls. Works very
well.
Nufone.net is where I get my inward dialing for my VoIP. Also good
experience so far.
Thanks,
Wiley
Hy List,
i´m having a big problem, with my new Asterisk 1.2.1 Server i cannot
register with my SIP-Providers. With my old Asterisk Server i hadn´t
such problems.
Here is the relevant part of my new sip.conf:
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
Hi,
On Mon, Jan 16 2006 Louis-David Mitterrand wrote:
Hello,
Using asterisk-1.2.1 I am trying to convert my music-on-hold files from
.wav to alaw:
% sox moh.wav -r 8000 -c 1 moh.al resample -ql
The file sounds fine when listened with:
% sox mox.al -t ossdsp /dev/dsp
Hi all,
Do you monitor call quality ?
If positive, how do you proceed ?
Which issues (echo ? call interruption ?) do you prevent with such
monitoring and which conter-measures do you engage when a problem occurs ?
Cheers
Olivier
___
--Bandwidth
I've been using Teliax for about 2 months and it's been great. Great
quality, features. Prices are a bit on the high side, many say this is
because they have great service. However, I haven't seen that; I emailed
and left a voicemail 2 days ago, I have yet to receive a reply. I'm not
confident
the oreilly Asterik book can be found at:
http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip
Mit freundlichen Gren
With kind regards
Klaus Peras
Dovid Bender schrieb:
in Sip.conf make sure to add NAT=yes. Also I reccomend
reading the new book that came out. (Dont have
Hi,
I would like to see if during a call a new voicemail was recorded. I want to
send a SMS to mobile phones if someone recorded a message on our voicemail
system.
I can use VMCOUNT to see if there are new messages in the Inbox but this will
result in new SMS being sent even if the caller
I have used virtual machines in the pastt o run
asterisk and it didnt run to well. The problem is that
the ZTDUMMY uses hardware for the timing. The problem
is that the hardware isnt real so natrually you wont
get acurate timing.
Dovid
--- Mauro Zanin [EMAIL PROTECTED] wrote:
Hi everybody
I'm
here a little list of sites about that
http://www.marko.net/asterisk/archives/0209/0583.html
http://home.cogeco.ca/~camstuff/agi.html
http://www.voip-info.org/wiki-Asterisk+AGI
- Original Message -
From: Innocent Evil [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Dov Bigio a écrit :
Ok.. but I don't use Real Time at all.
I just use cdr_mysql. It would be smarter if it simply ignored MySQL outages
or at least just logged, but without stopping.
What would be even nicer would be for * to buffer it for a while before
it starts dropping cdrs...
here a little list of sites about that
http://www.marko.net/asterisk/archives/0209/0583.html
http://home.cogeco.ca/~camstuff/agi.html
http://www.voip-info.org/wiki-Asterisk+AGI
- Original Message -
From: Innocent Evil [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday,
Hi all,
I've been working with * for a long time now, but only with analog FXS/FXO
systems.
I am venturing towards setting up a box in Germany now and I believe that
requires a Fritz card? Do I even have to use the Fritz cards? Why not a
Digium card
We have 2 ISDN lines ( -- 6 handsets)
On Tue, 2006-01-17 at 10:39 +, scott wrote:
Hi All
I have some grandstream phones registered to my asterisk and all internal,
external, voicemail services etc are working very well.
I am not sure that it is a problem more so an annoyance. If someone dials my
extension number or
On Tue, 2006-01-17 at 10:51 -0600, yrving rivas wrote:
The log files don´t help me very much.
The errorlog should be helpful for your webserver. I am unsure where
those are by default, but it sounds like this is a 500 series error
code, possibly from a failed cgi. My guess is that you would
Miami was in December. Check the website for upcoming locations. As
far as the test good luck.. Its digital and not given out. all I can
tell you is to study all aspects including voip standards and such.
brian
On Tue, 2006-01-17 at 18:06 +0100, Giorgio Incantalupo wrote:
Hi,
is there
I use Cisco 7940's and 7960's over here, loaded with SIP, and they do very
nicely. Also, the SCCP channel for * is under heavy development, and may
offer a future option to convert in that direction, too (SCCP, or skinny, is
their native tongue, not SIP). We got our phones from John Putnam at
Hi
I wrote a small patch to netcat to work with unix domain sockets to
enable me to communicate with an asterisk daemon through the
unix-domain socket /var/run/asterisk/asterisk.ctl .
Only then I noticed that reading the code is done very differently than
a typical network protocol: it expects
Hi I want to authenticate the asterisk users from the LDAP directory server not from the sip.conf.I tried to use the astirectory-1.2 . But i am not able to configure it properly. If somebody
used it then please help. In the res_ldap.conf file i made the following entries. I am using my normal
I thought about using a softphone for my callcenter. I was using x-light
to do some testing, but I always had quality issues. One computer would
have static, the other echo. I think those problems were in the
soundcard or or other computer hardware. For the cost to fix those
problems on the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I believe you can use the externnotify to accomplish this...
Sean
Koopmann, Jan-Peter wrote:
Hi,
I would like to see if during a call a new voicemail was recorded. I want to
send a SMS to mobile phones if someone recorded a message on our
Ben Fried wrote:
Just to be clear, I have inbound and outbound faxes working with my
TDM400, by going the iaxmodem and hylafax route. No need for a
separate modem or an x100p card.
Be
Ah, that's interesting. Can you provide some details on how you set it up?
Phil
Thank you all for the responses.
Matt, when you say they (gnuDialer) have been focusing on other areas,
what do you mean by that?
The reason I'm asking, 2 features that I'm looking for that vicidail
doesn't support:
1. Scheduled call back
2. The ability to detect an answering machine. Although I
Thats usualy a phone specific feature, however you could setup a macro
that will screen the calls before you answer them.
On 1/17/06, Obelix [EMAIL PROTECTED] wrote:
Is there a key sequence to stop a call as its ringing, before the call is
answered?
The * key stops a call after it is
Hi,I have built another Asterisk box using one ISDN HFC-S card and Bristuff-0.2.0-RC8p. But this time it behaves very strangely. Asterisk simply hangs and in logs I receive something like this:
--NOTICE [1197]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1
--NOTICE [1197]: PRI
What version of asterisk you using?
On 1/16/06, Kerry Garrison [EMAIL PROTECTED] wrote:
Wish I could help, I can tell you I have never been disconnected from a
call on my system that has been running non-stop for months with an
SPA-3000.
All of my settings are documented here:
Koopmann, Jan-Peter wrote:
I would like to see if during a call a new voicemail was recorded. I want to
send a SMS to mobile phones if someone recorded a message on our voicemail
system.
I can use VMCOUNT to see if there are new messages in the Inbox but this will
result in new SMS being
On Tuesday, January 17, 2006 4:13 PM Pisac wrote:
You are right, only outgoing calls!
I found lines that you mentioned, but I do not understand where is
difference? In current chan_zap.c I read:
if (!IS_DIGITAL(ast-transfercapability)) {
set_actual_gain(p-subs[SUB_REAL].zfd, 0, p-rxgain,
I use Cisco 7940's and 7960's over here, loaded with SIP, and they do very
nicely. Also, the SCCP channel for * is under heavy development, and may
offer a future option to convert in that direction, too (SCCP, or skinny, is
their native tongue, not SIP). We got our phones from John Putnam at
One question, if I change chan_zap.c, what should I type to compile and
install only that module, and not whole asterisk again.
I tried
gcc chan_zap.c -o /usr/lib/asterisk/modules/chan_zap2.so
but I'm getting error during compiling.
___
--Bandwidth
i cannot take any responsability if your computer explodes or
something, but im attaching you a patch that you may want to give a
try.
regards
On 1/17/06, Reto Kortas [EMAIL PROTECTED] wrote:
In extconfig.conf I have:
voicemail = odbc,asterisk,voicemail_users
sipusers =
by the way, you MUST create peer contexts in sip.conf.
For example, if you have peer 10 in your db, you should write at least
[10]
type=peer
in sip.conf, so it will attempt to load the rest of the info from the
DB. Hhehe, i know is kind of useless for now, but if it works, i can
make it to read
You could always set the pager field to the email address of the
person's phone. This would send basic information about the call.
Aaron
Koopmann, Jan-Peter wrote:
Hi,
I would like to see if during a call a new voicemail was recorded. I want to
send a SMS to mobile phones if someone
Hello all,
I have a minor annoyance and I think there is a better solution...
I have a dialin menu, with options 1 and 2.
I need the ability to do a zaptel flash transfer, but to get the proper
extensions to work, I need to do an include of the context which
contains those extensions. Is
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