I can call my Broadvoice DID from a outbound caller-id blocked phone,
and BV happily delivers the CID to Asterisk (and then on to my IP phone
display.) I've tested with the *67 prefix from a PSTN phone to make
sure it was supposed to be blocked. The number is always correct, but
sometimes
On 24/05/05, Anton Krall [EMAIL PROTECTED] wrote:
Ive also tested yac... Problems there.. How to tell asterisk what ip to send
the info to...
Yes. I currently use a simple lookup in an AGI to map extension number
to IP. Ideally, though, I'd like the listener to 'log on' to an
extension to
On Wed, May 18, 2005 at 07:07:02AM +0100, Matthew Walster wrote:
I've just migrated Asterisk from my old Gentoo system to an Ubuntu system,
copied across all the /etc/asterisk files and now it fails to work. After
brief looks, I find that it can't access:
/var/log/asterisk/messages
On Mon, 23 May 2005, Kanuri, Seshu (Company IT) wrote:
FireFly is the best of the IAX softphones. Other softphones do not work
as good as FireFly. DIAX has many bugs still. DIAX Softphone disconnects
with Windows DLL errors everytime there is a problem in the call like
Asterisk Channel Not
Hi
I hope that better late than ever
On Mon, May 16, 2005 at 10:24:33AM -0400, Joel Duffield wrote:
Hi All
I had asterisk running on a xercom
Xorcom, please :-)
install, I upgraded the box to a full
debian install and now asterisk is not starting from on boot. I can start
asterisk from
Rod, thanks for your feedback:
In article [EMAIL PROTECTED],
Rod Bacon [EMAIL PROTECTED] wrote:
Tony, I have managed to compile both versions (on separate servers,
obviously), and have them working. My question is specifically related
to which one do I choose?.
You have the luxury of choice
Hi Seshu,
DIAX has many bugs still.
DIAX Softphone disconnects
with Windows DLL errors everytime there is a problem in the call
like
Asterisk Channel Not available etc.
Can you provide more details and not just talk about 'many bugs'?
I have no feedbacks about such kind of problems in DIAX.
[EMAIL PROTECTED] wrote:
Hi, how I can connect Astrisk to my iptel account???
[iptel.org]
type=friend
host=iptel.org
fromuser=my_account_name
secret=
nat=yes
We've had problems as well when the friend in sip.conf
was named the same as the domain. Call it something
else (iptel-out?)
On Mon, 2005-05-23 at 23:19 -0700, Johnathan Corgan wrote:
Is caller ID blocking implemented by sending the cid information anyway,
but with a bit that says don't give to end user? I guess BV would be
ignoring this bit.
That is it exactly. There is a privacy bit that is set and odds are
Hi,
Citeren Anton Krall [EMAIL PROTECTED]:
Let mek now what you need Florian and Ill send it offlist.
| Seems to me Im been displayed both... How can I control it?
|
|No way to know that without more in-depth knowledge about your
|configuration (i.e.dialplan, what channel have you
Ian Pattison wrote:
Hi all,
Yesterday, in an attempt to take back my phone room, I pulled everything apart
as far back as the demarc and rebuilt it. In the process of putting things back
together I accidentally connected my incoming lines to my FXS ports and my
phones to my FXO ports. I
Hi all,
I used oh323 support from inaccess. It work very well.
I would like to test h323 integrated support.
This my problem when I test it :
I cannot heard any thing in both way.
The test is : SIP -- Asterisk -- H323
This is th debug trace from h.323 :
--
Hello,
I want to record the message of both parties during a
dial.
My extensions.conf at the line where dial is looks
like this:
exten =
s,803,Dial(SIP/arjankroon2,30,rR)
My Sip.conf look like this:
[arjankroon2]
;Turn off silence
suppression in X-Lite (Transmit Silence=YES)!
Hi,
Is there a way to hear what your telco has to say (Early B3 connects)
using zaphfc (zaptel)?
All suggestions are welcome ;-)
Gr. Steven
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Hello,
I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found
at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send
a register staement (nothing in thertereal log). With the 1.0.3.81
version, the phone register properly.
Is ther any know bug with the SW
I subscribed to the dundi mailing list, but so far I have not got a
single message from there. Is there a message archive?
I want to setup DUNDI. I have a peering agreemrent, but what is next?
I copied from the wiki all parts, but still I am a little bit lost. Has
anybody setup DUNDI?
bye
Hi everybody I am a newbie here...
When an * works as a SIP proxy, and the UA sends rpid header, the
asterisk does not send this header forward...
Is it possible to set this up somehow??
As I can see in the current chan_sip.c, only callerid can be set up in
some circumstances, but rpid
Hi,
zapata.conf:
prindication = passthrough
best regards
Klaus
--
Klaus-Peter Junghanns
Am Dienstag, den 24.05.2005, 10:34 +0200 schrieb Steven Lam:
Hi,
Is there a way to hear what your telco has to say (Early B3 connects)
using zaphfc (zaptel)?
All suggestions are welcome ;-)
Gr.
Please I have combed the Archive to no avail on this problem protocol
control problem in oh323.
I'm receiving calls from CISCO AS5300 - Asterisk - Zap Channel. The
calls clears the remote location but drops on my own end. Please what
could be
wrong. I have included the oh323.conf and log
Hi!
I'm using FastAGI (agi://) to make some calls. To do the dialing i use EXEC
DIAL Zap/g1/
But how can I make answer supervision with FastAGI? DIAL command won't return
until call is finished.
Thanks in advance!
--
Daniel
___
Hello
I was wondering why the asterisk processes where
growing, from one process at launch time to more than twenty after some
weeks and I was wondering if all resources won't be eated at last (with
2% of memory by process make the sum)
I know this question has already been
Hi list,
I am trying to install a TDM22B in my * box. I am
using a Pentium III with SuSE 9.2 (2.6.8-24-default) and * 1.0.7. I installed
the TDM22B (no IRQ conflicts), recompiled zaptel, libpri and asterisk. I
modified a file (cant remember which one) to create my devices entrien
in
Hi
Can any one tel me what is [EMAIL PROTECTED]
Thx
Q
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Title: Message
Google
is your friend:
http://www.google.com/search?sourceid=navclientie=UTF-8rls=GGLD,GGLD:2003-36,GGLD:enq=%22What+is+Asterisk%40Home%22
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
QuintinSent: 24 May 2005 11:41To:
Hello all!
First of all, this is my first post to the list.
Ive tried to find my answers in the forums and by Googling , but no luck.
My apologies if this question has been answered before.
Ive set up an Asterisk box with four local SIP
users. The Asterisk box uses a SIP provider for
Hi;
Using [EMAIL PROTECTED] and it working well in network but when
can not logged in over internet although the server is reachable
Does anybody has any idea?
Thanks
Betul
Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini
In article [EMAIL PROTECTED],
Bjorn Ove Kristiansen [EMAIL PROTECTED] wrote:
I've set up an Asterisk box with four local SIP users. The Asterisk box uses
a SIP provider for placing external calls and receiving incoming calls as
well. In other words, there's no PSTN or ISDN lines attached to
Asterisk @ Home This CD will install
everything you need to get your Software PBX going.
Its a complete ISO CD that brings
together the OS (CentOS 3.4)
Asterisk Software version 1.0.7 stable
AMP Asterisk Management Portal
Web GUI
FTP
TFTP
Plus many more items.
Every
Go give idefisk a try (and let us know how you like it).
http://www.asteriskguru.com/idefisk_beta.html
A new version will be online tomorrow.
Joachim.
Terry H. Gilsenan wrote:
I disagree,
I use iaxphone from here:
http://www.sokol-associates.com/IaxPhone.htm
I have found it to be more
Hi there
I have been successfully using the asterisk command MYSQL to read
information from a MySql database, but was wondering if there was any way of
WRITING data (ie user input data) to the database ???, looking through the
parameters of the MYSQL command it seems as though this function isn't
I having been DIAX without problems.
Only one bug I can find and thats related to using Large screens and
window sizing.
it only happens @ the moment on one windows machine which hasn't been
service pack'd.
Now if there was a way to add a free g729a codec to it :-)
Gary
On Tue, 24 May 2005
If I put an incoming PSTN call on hold, and then dial another extension,
which busy, I want to be able to use the hook-switch to cancel the
enquiry call, so that I can go back to the original incoming call.
Currently if I do this the incoming call gets connected to busy tone and
I lose control of
Thanks you for your interest in our project.
Effectively we are looking to set up a small functioning demo site so
that we can show some investors as to the potential and future of VoIP
digital communications.
The ultimate goals of the service is establish presence that can grow into
a large
Has anyone had any echo issues with the Digium TDM02B FXO card? I
purchased a clone wildcard on E-Bay for $8.00 and have had horrible
echo issues I'm assuming it is because of impedance issues? Just
wondered what people's takes were on the TDM02B FXO and echo?
I just added 2 Digium T100 cards. When my * box boots, it found them and configured them.When I enter genzaptelconf, it comes back with the following error:
line 13: Unable to open master device '/dev/zap/ctl'Unloading zaptel hardware drivers:Removing zaptel module: rmmod: module zaptel is not
Sorry about last posting, typo...
I just added 2 Digium X100P cards. When my * box boots, it found them and configured them.When I enter genzaptelconf, it comes back with the following error:
line 13: Unable to open master device '/dev/zap/ctl'Unloading zaptel hardware drivers:Removing zaptel
Vikram Rangnekar wrote:
static ! Get your carpets washed and use static guard on it.
Thank you everyone for the replies. After doing some testing, it has
been determined that it was the phone that was the cause of the user
being shocked. We could relocate the phone, switch to a power
Well I've been trying to get faxing over regular h323 to work, and the
sending part hasn't been working too well. And since t38modem has t38
support it seems like it'd be more reliable for faxing so I thought it
might make sense if there was some sort of t38modem channel in
asterisk. How to
Anyone have a practical experience/knowledge relative to why a 5ESS
central office switch would require a w in the Dial statement to
handle analog pstn-fxo calls?
I fully understand what w is doing, just trying to better understand
why a 5ESS doesn't accept dtmf a little quicker then it
Does anyone know of a way to play a random sound file in a given directroy.
Ideally I want to incorporate the random play with the Asterisk 'Read'
command, however any other suggestions would be good.
Cheers
Ed
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Below posting seems like a FoIP success story. I trawled the list but
found no practical guidelines, (only references to
http://www.voip-info.org/tiki-index.php?page=FoIP
which also does not have practical guidelines)
on setting up FoIP functionality using T.38.
I am aware that faxing over IP
I needed a system to allow my windows clients to hit print and
eventually result in a fax being sent via asterisk. I used smbfax and
modified it to allow the phone number to be supplied by means of an
email and it all works fine
...except I do not know how to read off the TxFAX application
Of course Asterisk needs a T.38 facilities in various forms - gateway,
termination, RTP-IAX, etc. Its just that t38modem is of no help
whatsoever in achieving that.
I am implementing T.38 for Asterisk, but I keep getting diverted by
higher priorities.
Regards,
Steve
Matthew wrote:
Well
AGI script in Perl (or whatever) should do it easily.
Get a random number between 1 and n then index into a list of files.
Then either play the file from the AGI script or just return it's name in a
variable for later playing.
/edg
--On Tuesday, May 24, 2005 2:01 PM +0100 Marshall,
Maybe this marks me as a real newb but where do I set the number of
rings that a phone has before it sends it to voicemail?
Also for some odd reason when I ring an extension attached to my
sipura 2100 ATA it takes it about 12 seconds to start ringing after I
dial it (sits there with dead air on
Hi all.
I am unable to answer calls coming into asterisk
over PSTN. (UK)I want to have a call answered by my TDM400P/FXO module and
forwarded to a sip phone.
When I make a call from the PSTN to the BT line
installed on my FXO module the sip phone rings however, when i pick up
thecall using
thanks to all for your help
i've keeped busy myself reading tons of documentation at
http://www.voip-info.org/
and
http://idsn.jolly.de
and between pratical hw/sw experiences and theory ... i'm stuck.
my configuration should be the following:
hardware idsn nt1 input isdn card on asterisk
Rich Adamson wrote:
Anyone have a practical experience/knowledge relative to why a 5ESS
central office switch would require a w in the Dial statement to
handle analog pstn-fxo calls?
I fully understand what w is doing, just trying to better understand
why a 5ESS doesn't accept dtmf a little
I'm trying to compile spandsp and Asterisk but the patch makefile hunks
don't even resemble any parts of the asterisk Makefile. I am using the
latest version of both. Has anyone else run into this problem?
-Mark
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On May 23, 2005 02:44 pm, Ian Pattison wrote:
Yesterday, in an attempt to take back my phone room, I pulled everything
apart as far back as the demarc and rebuilt it. In the process of putting
things back together I accidentally connected my incoming lines to my FXS
ports and my phones to my
The patch is not that much.
I just opened it up - and then typed in the patch information into the
Makefile manually.
I have to do this for another application, so it wasn't that hard to do.
Even if i tried to copy / paste it would complain (the make process) on
the app_txfax.so : app_txfax.c
On May 22, 2005 04:46 pm, todd wrote:
Thanks for the info: do you know if the g729 codec for Asterisk is limited
to only the G.729A version. Is there G.729 B available for Asterisk?
Does it matter?
g729 has fixed-point and floating-point implementations. The data coming out
the back end is
Hi!
Has anybody tried this yet?
http://www.kapanga.net/IP/benefits.cfm
A softphone that claims to support T.38
regards,
klaus
Matthew wrote:
Well I've been trying to get faxing over regular h323 to work, and the
sending part hasn't been working too well. And since t38modem has t38
support it
Hi,
some of my sip fones which have several external numbers assigned
are not reachable after a certain timespan. Instead of the fone the
Voicemailbox is trigger in busy mode. After a reboot if the sip-fone
the problem goes away for some time. Ive seen this problem with Sipuras
and Grandstreams.
On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote:
Hello,
I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found
at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send
a register staement (nothing in thertereal log). With the 1.0.3.81
version, the phone
On May 21, 2005 01:22 pm, Julius Igugu wrote:
You need to instal the module 'libipt_ipp2p.so'
No, you only need that for the P2P stuff; rc.tc works just fine (minus the p2p
reprioritization) without it.
I agree with the earlier poster, it looks like you have a kernel that has no
concept of
Brett, Gary wrote:
Hi there
I have been successfully using the asterisk command MYSQL to read
information from a MySql database, but was wondering if there was any
way of WRITING data (ie user input data) to the database ???, looking
through the parameters of the MYSQL command it seems as
On May 20, 2005 11:52 am, Francois Lambert wrote:
http://www.abptech.com/mainpages/products/citelGateway.html
Have a look at these guys. They do have a gateway for Avaya and Nortel
and they say, it is certified with Asterisk.
I've seen these before but still haven't got any pricing info for
exten = 1234,1,Dial(SIP/1234,Number_of_Sec_for_Ringing,tr)
Tim P wrote:
Maybe this marks me as a real newb but where do I set the number of
rings that a phone has before it sends it to voicemail?
Also for some odd reason when I ring an extension attached to my
sipura 2100 ATA it takes it
On May 22, 2005 04:43 am, Jean-Christophe Heger wrote:
Depending on you bandwidth, you might not need QoS. Priority could be
enough.
Provided that the ISP doesn't just strip the bits anyway. I know many who do
this, especially since kids tend to set their traffic to high priority
because
On May 20, 2005 09:42 am, Geoff Manning wrote:
I am trying to prove a concept of call routing before we move towards
development of a production system. I need to have calls routed coming into
a call center based on DNIS. What type of syntax is needed in the
extensions.conf file and how can I
Good day all
I need some help
What is the device= in capi.conf
How will it look for a 4 port card?
I have a section for each msn...but how do I tell witch msn is witch
port because in the extensions.conf I dail [EMAIL PROTECTED]
Please give me a example of more than one file
Thanks
Altus
Now for the fun one - change ring pattern?? Like distinctive ringing? Is
this supported by asterisk or the end-point
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Umair Bari
Sent: Tuesday, May 24, 2005 10:20 PM
To: Tim P; Asterisk Users Mailing List -
On May 22, 2005 03:20 am, chawki hammoud wrote:
My ISP has the internet connection set-up where 8
people share the bandwidth. Would the script still
help boost my voip calls?
You can only control your own bandwidth. Your ISP would have to be willing to
help you in this matter.
-A.
On May 20, 2005 07:21 am, Phill Wolf wrote:
I wish the card (or Asterisk) waited for the dialtone before dialing.
Sometimes the phone company gives a dialtone within 1 second; other times
it takes 2 seconds and occasionally 3 seconds or longer. Crufting a large
number of w's into the dialplan
Anyone have a practical experience/knowledge relative to why a 5ESS
central office switch would require a w in the Dial statement to
handle analog pstn-fxo calls?
I fully understand what w is doing, just trying to better understand
why a 5ESS doesn't accept dtmf a little quicker then
Hi, im working with Gentoo as you toldme every user HAs a number but i
just cant find it...
i do not have any special hardware i only have VoIP,
can some one please helpme providing some example IAX and SIP users
where i can see their phone numbers?
bye and thanks!
On 5/24/05, Samy Antoun
Rich Adamson wrote:
Anyone have a practical experience/knowledge relative to why a 5ESS
central office switch would require a w in the Dial statement to
handle analog pstn-fxo calls?
I fully understand what w is doing, just trying to better understand
why a 5ESS doesn't accept dtmf a little
You will have to add the 4 or so lines to the make file manually.
Chris
- Original Message -
From: Mark Ratering [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, May 24, 2005 7:51 AM
Subject: [Asterisk-Users] spandsp issue
I'm trying to compile spandsp and
Shhh... Be very very quiet. I've noticed CID delivery of restricted
CIDs on several providers. I don't want this to be turned off.
-Original Message-
From: Johnathan Corgan [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 24, 2005 1:19 AM
To: Asterisk Users Mailing List -
You can do nearly anything you want with AGI. Writing to databases,
sending instant messages, swapping log files, updating a web-site or
ordering pizza.
-Original Message-
From: Brett, Gary [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 24, 2005 6:37 AM
To: 'Asterisk Users Mailing
On Tue, 24 May 2005, Stankiewicz Michael wrote:
i'm quite sure that sw side is correct and, again, i've got doubts on
the cables, since your two responses (thanks a lot to Alex and Emanuele
{ciao}) are opposite.
i've made 2 cables, following the schemes one here:
I would like yac to have the hability of running any external command line
for example, opening a http:// url or running any command line.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Peter Bowyer
|Sent: Martes, 24 de Mayo de 2005 01:35 a.m.
On 24/05/05, Anton Krall [EMAIL PROTECTED] wrote:
I would like yac to have the hability of running any external command line
for example, opening a http:// url or running any command line.
OK - presumably with parameter substitution from the incoming callerid
and/or extension - or would you
On Tue, 24 May 2005, altus wrote:
Good day all
I need some help
What is the device= in capi.conf
It must be the last entry of a section and defines the number
of B-channels of that section (controller).
Normaly you should have one section for each controller ending
with 'devices=2' (if it's a
Bob Goddard a écrit :
On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote:
Hello,
I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found
at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send
a register staement (nothing in thertereal log). With the 1.0.3.81
It is called search the wiki!!!
http://www.voip-info.org/wiki-Asterisk+Zap+channels
But can only be done for ZAP channels..
On a side note.
When are you guys going to fix your QWEST peering out of Richmond??? I
would really like to be able to use my Asterisk box during business
hours. But a
Is it possible to have the server entry on the
Provisioning file a domain (e.g. sip.mydomain.com)
instead of an ip address?
no the IAXy does not do DNS
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Don't know if this had been done before, but I went and HTMLized for my
own use (and anyone else's) the Asterisk Sounds listing(s). Here are
the links to the finished pages:
Asterisk 1.0.7 Sounds (sounds.txt) Transcriptions Listing:
http://www.nathanpralle.com/software/ast_soundlist.html
no the IAXy does not do DNS
Accidental hit of return...
so you have to reprovision if you have a server on a dynamic ip. I do
this by generating a new iaxy.conf file each time the ip changes and
sending it out tot he IAXy. Works every time.
___
Maybe this marks me as a real newb but where do I set the number of
rings that a phone has before it sends it to voicemail?
go to CLI and type show application dial
Also for some odd reason when I ring an extension attached to my
sipura 2100 ATA it takes it about 12 seconds to start ringing
JD wrote:
I'm still relatively a novice with asterisk and am having issues with
echo.
The calling party that calls a PSTN number doesnt hear the echo, but
the answered
side via sip or forwarded to another PSTN number over voip hears
excessive echo that
makes it difficult to communicate.
Anyway that works would be nice :) What I would like to do Is get the
callerid and then pass that parameter in order to open a http url containing
the callerid to pass or open a CRM webpage like SugarCRM.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On
go in to the [EMAIL PROTECTED] set up
http://yourip/maint
inter in your user name and password go
to
config edit
sip.conf
go to the general section
Note: if you are behind a NAT Firewall, you will
probably need to add thefollowing lines to hte [General] section of your
sip.conf file. Adjust
I would say that we would need to be able to scale to the 200+ consecutive
call range in the near future (6 months), and hopefully to the 500+ within
the next two years.
We plan on using at least 1+0 raid with a minimum of 4 disks (minimum of 10K
rpm SCSI). We also are planning on recording all
and man does it kick ass to!!! :)
email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn
messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5
- Original Message -
From:
Ariel
Batista
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Tuesday, May
HelloI've been having the following problem today :I have a quite simple dialplan made to receive a fax:[answer-extension]exten = 1,1,Answerexten = 1,2,Macro(setcallerid)exten = 1,3,Ringingexten = 1,4,Wait(3)exten = 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},${EXTENSION})exten =
Ok, from what I can see _NO_ passive ISDN cards will work with Asterisk on freebsd, is this correct is it likely to change soon?Secondly, if this is likely to be the way for a while, what is the lease expensive card that will work with FreeBSD?Also, can I use DID (Direct Inward Dialling) on
We have setup a Lucent TNT as a SIP gateway
connecting our 3 PRIs. Currently, when our callers make calls out on the
PSTN, the caller-id is being rejected by our service provider because the Lucent
is setting the type of number in the PRI message to international instead of
national. I did
Does anyone have the MySQL add-on as a binary for OS X? Or am I
getting it wrong and MySQL is installed by default?
Thanks.
Michael
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they ever going to fix it?
email:
[EMAIL PROTECTED]
gmail:
[EMAIL PROTECTED]
msn messenger:
[EMAIL PROTECTED]
aim:
hanksmith5
skype:
hanksmith5
- Original Message -
From: Bob Goddard [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Michael Stearne wrote:
Does anyone have the MySQL add-on as a binary for OS X? Or am I
getting it wrong and MySQL is installed by default?
Thanks.
Michael
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Note: this is for chan_capi-0.3.5 and will be changed from 0.4.0
0.4.0? who is working on chan_capi? long time with no updates from.
I'm rewriting the chan_capi logic.
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Michael,
I think this is not really an asterisk related post.
Next time use a mysql list or google, or your own research.
You can install mysql from source using Fink on OS X.
http fink.sf.net
Or, if you wanted, you could go to the Mysql downloads page and there
is a binary there.
Mysql
OK, If you are going to be recording all calls you will need to rethink
things a bit. Recording calls limits you to 50-60 consecutive conversations
per server before audio distortion starts to occur. You will probably want
to think about limiting yourself to 3 T1s per machine. There are many ways
On 5/24/05, Roman Volf [EMAIL PROTECTED] wrote:
Michael Stearne wrote:
Does anyone have the MySQL add-on as a binary for OS X? Or am I
getting it wrong and MySQL is installed by default?
Thanks.
Michael
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Asterisk-Users mailing list
I have the following issue:
1) Call comes in from PSTN to Asterisk (IP A) and
Asterisk forwards call to a SIP Proxy (IP B)
2) SIP Proxy (SER) forwards the call to a registered
user. User does not answer and Call Forwarding is
turned on for the user and the number to forward the
call is a PSTN
Matt,
Are you doing any call recording / monitoring? What percentage?
Ilan
On 5/23/05, mattf [EMAIL PROTECTED] wrote:
For an inbound call center with 4 T1s and 30-50 agents on you would do just
fine with a single, one-processor machine. We have handled more than this on
a single P4 server
Thanks Kyle. I was actually talking about the adding mySQL
functionality for Asterisk to OS X, not the MySQL server for OS X
itself.
Thanks,
Michael
On 5/24/05, Kyle Loree [EMAIL PROTECTED] wrote:
Michael,
I think this is not really an asterisk related post.
Next time use a mysql list or
On Tue, 24 May 2005, Sergio wrote:
Note: this is for chan_capi-0.3.5 and will be changed from 0.4.0
0.4.0? who is working on chan_capi? long time with no updates from.
I'm rewriting the chan_capi logic.
kapejod already did 0.4.0-PRE1, which I use as base for cleanup, fixing and
new
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