Hi,
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Sunday, September 05, 2004 7:04 PM
Subject: RE: [Asterisk-Users] ChanSpy by anthm and more...
Hello,
As I understand * don't supports T.38 in Zap channels (please correct me
if I'm wrong, BTW is there plans for such support?)
I believe it's should support T.38 in pass-thru mode. I mean setup
like this:
Hardware gate with T.38 -- Asterisk -- Hardware gate with T.38
But I had
On Mon, Sep 06, 2004 at 10:40:44AM +0300, Elman Efendiyev spake thusly:
I believe it's should support T.38 in pass-thru mode. I mean setup
like this:
I have seriously been considering giving this a try lately also. I will be
very interested to see how it works out. What sort of problems exactly
Hi all,
I used to have an OpenLine4 card, but decided against using it due to
some problems with hangup detect. Does anyone on the list actively use
Voicetronix's OpenSwitch12? What are your opinions on the card?
Cheers,
Flynn
___
Asterisk-Users
Hi list!
I've got a strange phenomen running asterisk for a while. After about
two or three days without restarts, asterisk is not able to create
SIP-Channels anymore, but gives me messages like
Sep 4 00:12:06 WARNING[7175]: Unable to allocate channel structure
Sep 4 00:12:06 NOTICE[7175]:
Another reason can be a missing or incomplete addhandler line in
/etc/httpd/conf/httpd.conf. The line should look like:
Addhandler cgi-script .cgi .pl
My Redhat installation was missing the .pl and when trying to run perl
based cgi scripts also generated the same 500 error. Correcting the
I have tried to get this working, but can not get it to authorise:
I created my Communicator logon from a Yahoo account (not a btinternet
account). Assuming my Yahoo username is username, BT Communicator
software logs on to the SIP proxy as username[EMAIL PROTECTED]
according to the trace which
Hi,
I have found some cheap single port FXO Cards on ebay. They are
apparently X100P compatible like the cards sold by Digium
My question is 1) Are they any good 2) will 4 such cards work in a
single * box without too much trouble?
Thanks
Andy
___
Thanks for the quick response, William. Yes, I was able to force it by removing ULAW
from the sip.conf but I needed that in there. If it is there, then the Sipura as well
as X-PRO use ulaw. I didn't really want to adjust the clients because I am planning on
having random clients accessing the
box100 wrote:
My iax.conf file includes the following under the general section
A SIPURA is a SIP device, configure the codecs under sip.conf not IAX.conf.
disallow=all
bandwidth=low
allow=g729
allow=ulaw
Thanks,
Roger Easlick
On 02/09/2004 at 10:08 Andrew Newton wrote:
Hi,
I have been looking for info on * and the BT Easicom 1000 without much
luck when i found a post to this list from Andy Powell saying that he
had the phone working quite well. Before i go buy a shedload of these
things I would like to know what
Hi,
I have the following setup:
E100P
SER * - PBX
This works just fine, except when there are users on both boxes (ie. SER
and asterisk), whose usernames are the same, although the realm is
different.
An example:
user '[EMAIL PROTECTED]' wants to call some extension in
Today morning cvs server checkout problem:
cvs server: Updating asterisk-addons/format_mp3
cvs server: failed to create lock directory for
`/usr/cvsroot/asterisk-addons/format_mp3'
(/usr/cvsroot/asterisk-addons/format_mp3/#cvs.lock): Permission denied
cvs server: failed to obtain dir lock in
Jusc couldn'n transmit faxes trouth asterisk. It just hangs up when
starting a fax transfer
If U will do some experiments with it I would be happy to hear any
reslts/info
Thanks
--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
MMm
Got it authenticated at last. Generating the right MD5 hash was a help! But
it is only working on one copy of asterisk (this morning's CVS) so I will
rebuild the other.
Below are my sip.conf settings if it helps anyone. You may need to trace the
login from BT Communicator to see if you need
We are still having problems getting a Wildcard to work with a German E1
(PMX) interface.
When starting asterisk it shows all B-channels starting up successfully
(although our carrier told us only the first B-channel starts, if any at
all).
Incoming calls are not being signaled at all. (They
Hi all.
I've being reading posts from the list since yesterday and I feel this
question was answered a lot time ago, but the list archives are a mess
(yet). I hope some one is willing to help me out.
I want to set up this:
caller - PSTN (SOMETHING1) -- VoIP - (SOMETHING2)
Hi all,
I'm trying to configure a swissvoice IP10S but after a minutes
this phones appears like UKNOWN in sip show peers and it is unaccesible.
This phone can make call but it can't receive calls.
Any idea?
Regards,
srsergio
___
Asterisk-Users
Hi,
-Original Message-
I'm trying to configure a swissvoice IP10S but after a
minutes this phones appears like UKNOWN in sip show peers and
it is unaccesible.
This phone can make call but it can't receive calls.
What firmware are you running with ? Bog-standard IP10's come
Hi
Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719)
the best/only way to get callerid working in the UK with a tdm400p? I thought Id
seen a patch thatd gone into cvs, but maybe I was just imagining things
;)
Should this patch work against current cvs? Of the 3
SIP version
IP10 SP v0.0.1 (Build 5)
Regards,
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Florian
Overkamp
Enviado el: lunes, 06 de septiembre de 2004 13:42
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: RE: [Asterisk-Users]
Edward Eastman wrote:
Hi
Is this patch
(http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the
best/only way to get callerid working in the UK with a tdm400p? I
thought I'd seen a patch that'd gone into cvs, but maybe I was just
imagining things ;)
Check the bug tracker for
Hi, I have one Ambient MD3200 based modem, and it works. There are
problems, but I saw reporting of such problems also with Digium's
official hardware. I know that everyone should buy official Digium
hardware, because of their marvelous job with Asterisk, but US$ 100
was just too expensive for me,
Hi,
I've read through the Asterisk handbook and I'd just like clarification
from someone that's implemented the above before. Lets imagine I want to
use the CallingCard application and don't want to tell a client to buy a
channelbank (the analog extensions are too many to connect to FXS cards),
Kurt Bauer wrote:
Hi,
I have the following setup:
E100P
SER * - PBX
This works just fine, except when there are users on both boxes (ie. SER
and asterisk), whose usernames are the same, although the realm is
different.
At this point, Asterisk doesn't care about the
Brilliant - thanks, took me half an hour but it's working now.
Just for the record, settings as follows:
The patch on http://bugs.digium.com/bug_view_page.php?bug_id=009
(ast-UK-and-DTMF-pol-CID.diff) doesn't seem to work for current cvs, but I
backed up to cvs as of 31/08/04 and that worked
From: Rodolfo Grave [EMAIL PROTECTED]
I've being reading posts from the list since yesterday and I feel this
question was answered a lot time ago, but the list archives are a mess
(yet). I hope some one is willing to help me out.
I want to set up this:
caller - PSTN
[ wiki on xten/x-lite gets you to a 5mb pdf which tells you how to
do a windows install. deep :-( ]
anyone know how to make x-lite be # key transparent, i.e. send the
key when it is poked?
randy
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Asterisk-Users mailing list
[EMAIL PROTECTED]
This post is simply documenting a spouse-friendly way of using the
spa-3000 as both a fxs and fxo port for basic soho environments in
the US, allowing asterisk to participate as needed/wanted.
All home phones are connected _only_ to the spa-3000 fxs port.
The incoming home pstn line is
1) they will work. Can depend on the phone system in your area and how far
you are from the telco office(in cable feet).
2) 4 would be hard to make work. Each card needs a different and unique IRQ
and hates to share IRQ's with anything.
Lyle
- Original Message -
From: Andrew Newton
Ilia,
Think of a channel bank as a concentrator. In a single T1 channel bank,
you concentrate 24 analog two wire phone connections into a single 4 wire
digital interface.
So in your case, the RJ45 connector is for the T1 interface to Asterisk,
the local CLEC, or whatever you intend to connect
An analog phone need an FXS channel unit to drive them. An analog phone
line needs to connect to an FXO channel unit.
An analog phone goes off hook by putting a short across the tip ring. The
FXS or a pots line from the Telco puts battery on the Tip and
Ground(actually it's a signal return and
If you are in the US, go to your Graybar store and ask for a short
Amp50 cable and a harmonica. The harmonica has an Amp50 connector
on one side and RJ14 jacks on the other. For the cable, they'll ask you
the gender of the connectors at each end--you'll have to determine the
gender of the
Hi! and thanks a million for your answer. You hace cleared many of the
doubts I had, including the differences between the cards. At the same
time, new questions has arised:
Is there a possible configuration in case I dont have a broadband
connection in the called-end, for example, a modem
Rich,
I have an SPA-3000 with a similar setup to yours. Unlike yours, my cordless
phone system is on a SPA-2000 that connect through the SPA-3000 for a PSTN
connection. For LD I use 91xxx for (MCI) or 71xxx for (VoicePulse). In
addition to trunk access codes 7 9, I use FWD by dialing
My response is inline...
Hi! and thanks a million for your answer. You hace cleared many of the
doubts I had, including the differences between the cards. At the same
time, new questions has arised:
Is there a possible configuration in case I dont have a broadband
connection in the
Hello there,
I have just laid my hands on a pair of used Mediatrix APA III-4FXO/FXS VoIP
gateways. Anyone with the manuals so that I can configure them for use on my
Asterisk SIP server ?
Peter Mwondi
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I have an SPA-3000 with a similar setup to yours. Unlike yours, my cordless
phone system is on a SPA-2000 that connect through the SPA-3000 for a PSTN
connection. For LD I use 91xxx for (MCI) or 71xxx for (VoicePulse). In
addition to trunk access codes 7 9, I use FWD by dialing 8xx.
Can someone tell me
how to get to a mailbox login prompt when accessing the Asterisk VM remotely via
a PSTN line? I am running version CSV 8/25/04.
Thanks,
Larry
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
You could dedicate a PSTN line( phone number)
for that purpose. You could put a menu system(auto-attendant style) and
just dial 8500(demo is set for this exten to be the gateway to VM). Or if
your operator answers, have her transfer your call to 8500.
Lyle
- Original Message -
I am looking at building an IVR product with a few interesting features
and need some more information about how asterisk and VoIP work and what
I can get from them.
As far as I can tell when I use ISDN/GSM telephone networks the DTMF
information travels as data representing 'start tone' and
Thanks again Rich. I hope you dont mind to answer a few more
questions
My response is inline...
Hi! and thanks a million for your answer. You hace cleared many of the
doubts I had, including the differences between the cards. At the same
time, new questions has arised:
Is
Rich,
At first I thought it may be a setting in the SPA (and maybe it is I don't
know), but I also had this problem when using the SPA-2000 to a x100p
interface. I don't think it is a problem with the SPA because when I am in
a conversation utilizing the VoIP trunk (VoicePulse), and a second
On most VM systems you can press the * key or # key to get
a login prompt during your greeting. Is that not possible with this
system?
Thanks,
Larry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle
GieseSent: Monday, September 06, 2004 11:55 AMTo: Asterisk
Users
When a SIP call starts (INVITE / 200 OK), asterisk seems to create a random port number for voice (rtp) packets. Is it possible to force this port value (without using reinvite since i am trying to use SIP against something else than sip)
thanks a lot in advance
On Wed, 2004-09-01 at 22:02, Edward Eastman wrote:
Hi, thanks for the reply, only just got round to having a look at it again
(annoying how real life gets in the way of the important stuff ;)
I've had a go at ramping up the tx/rx gain but it doesn't seem to make any
difference. FWIW it's
Is there a possible configuration in case I dont have a broadband
connection in the called-end, for example, a modem connection?
No, there is no modem support built into asterisk. The problem is that
modems typcially do not support the bandwidth needed
Could someone please recommend a reasonably priced IP phone
that works well with *, has a decent (full duplex, echo canceling)
speakerphone, has at least two line appearances, and can
transfer / conference reliably?
The Wiki lists 35 brands of hardphone, but:
1. Most seem to be toys.
2. For many,
It is but you need to modify your dial plan to make it work.
I do it like such
[inbound] ; context that takes inbound calls and matches em and routes according
exten = 91808,1,Macro(stdexten,101,SIP/101) ; fwd
exten = 55,1,Goto(all-exten,101,1)
; fwd goes start to my stdexten to 101
Larry == Larry Shields [EMAIL PROTECTED] writes:
Larry On most VM systems you can press the * key or # key to get a
Larry login prompt during your greeting. Is that not possible with
Larry this system?
If you hist * during the outgoing message you'll get sent to the a
extension, if that exists
Well, I wont get tired of saying thank you for the quick answers.
Rich Adamson wrote:
Is there a possible configuration in case I dont have a broadband
connection in the called-end, for example, a modem connection?
No, there is no modem support built
Ok I think I found what I am looking for. You
need to added two items to your configs.
; Add the following context to your extensions.conf
(2001 is the main mailbox that allmymessages are left
in)
[vmlogin]
exten =
a,1,VoicemailMain(2001)exten = a,2,Hangup
exten =
i,1,Hangupexten =
thats about the most unsecure thing I've ever seen. there is a reason you
don't run apache as root and therefore having a script that sudo's is just
as bad.
try using the manager interface for better security. * shouldn't be running
as root either if we want to get nitty-gritty about security.
Larry,
I have my extensions.conf set up to wait for a * before ringing the inside phone, like
this:
[incoming]
exten = number,1,Answer
exten = number,2,Ringing
exten = number,3,ResponseTimeout(2)
; Cover * for voicemail access
exten = *,1,VoiceMailMain(1)
; Ring IAXy after ResponseTimeout = 2
check rtp.conf
-Kannaiyan
- Original Message -
From:
[EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, September 06, 2004 6:15
PM
Subject: [Asterisk-Users] SIP rtp port
forcing
When a SIP call starts (INVITE
/ 200 OK), asterisk seems to create a
Jamie Carl wrote:
Bob Knight wrote:
There is a linux package called mbrowse that you can use with your
mediatrix mibs.
I can get and walk everything in my 1204's.
For some reason I have not had any success with writes, but I have not
spent
that much time on it.
I don't even have the MIBs which
Brad,
I like the idea that no inside extension rings when you want to check VM
from an outside line.
Thanks,
Larry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brad Ediger
Sent: Monday, September 06, 2004 1:51 PM
To: [EMAIL PROTECTED]
Subject: Re:
Hey gang,
Sometimes when I call my cell thru * I only hear the ringing tone once or
twice then its completly silent until I pick up my cell phone and answer my
call. Any ideas on why I only hear a few ring tones?
Thanks,
Matthew
___
Asterisk-Users
You can only restrict the range of ports used, in rtp.conf.
I suppose restricting it to 2 ports starting on even number might do it,
but if you're not using SIP on one end, how are you going to start a call?
You need to have at least rudimentary call control for SIP invite and SDP
exchange, and
Good call Daniel I didn't even notice that.
As far as number of license it really depends on how many concurrent
calls you will be doing and if asterisk needs to transcode at all. If
you call from g729 device to g729 you are fine but g729 to vm would be
1 license etc.
On Mon, 06 Sep 2004
I was checking on the harware page of Digium and I found that there are
many TDM cards.
TDM10B - 1-port FXS bundle
Order Online
TDM40B - 4-port FXS bundle
Order
Online
TDM01B - 1-port FXO bundle
Order
Rodolfo Grave wrote:
Can you explain further what a FXS and FXO port represents in a call
process in general?
FXO port - Expects to RECEIVE dialtone and ring voltage
FXS port - Expects to PROVIDE dialtone and ring voltage
___
Asterisk-Users mailing list
hi guys i,m very new to asterisk and i need a little help with it i just
installed and configered the asterisk and i'm trying to dial from outside
and get to my voicemail but this what i'm getting from my asterisk box{Sep
6 13:08:58 NOTICE[-1093239888]: chan_sip.c:3922 sip_reg_timeout}so
In article [EMAIL PROTECTED],
William Suffill [EMAIL PROTECTED] wrote:
Good call Daniel I didn't even notice that.
As far as number of license it really depends on how many concurrent
calls you will be doing and if asterisk needs to transcode at all. If
you call from g729 device to g729 you
In article [EMAIL PROTECTED], Eric Wieling [EMAIL PROTECTED] wrote:
Rodolfo Grave wrote:
Can you explain further what a FXS and FXO port represents in a call
process in general?
FXO port - Expects to RECEIVE dialtone and ring voltage
FXS port - Expects to PROVIDE dialtone and ring
Thanks.
Tony Mountifield wrote:
In article [EMAIL PROTECTED], Eric Wieling [EMAIL PROTECTED] wrote:
Rodolfo Grave wrote:
Can you explain further what a FXS and FXO port represents in a call
process in general?
FXO port - Expects to RECEIVE dialtone and ring voltage
FXS port -
Gonzalo,
I have an APA III-4FXO and I tried using your configurations, I received the
message below:
-- Executing Dial(SIP/2010-edfc, SIP/[EMAIL PROTECTED]) in new stack
Sep 6 16:54:51 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x814bf0c (len 774) to 192.168.199.5 returned
Tony Mountifield wrote:
In article [EMAIL PROTECTED], Eric Wieling [EMAIL PROTECTED] wrote:
Rodolfo Grave wrote:
Can you explain further what a FXS and FXO port represents in a call
process in general?
FXO port - Expects to RECEIVE dialtone and ring voltage
FXS port - Expects to PROVIDE dialtone
On Mon, 06 Sep 2004 13:22:51 +0300, Vladyslav [EMAIL PROTECTED] wrote:
Today morning cvs server checkout problem:
cvs server: Updating asterisk-addons/format_mp3
cvs server: failed to create lock directory for
`/usr/cvsroot/asterisk-addons/format_mp3'
Hi there,
we have a strange problem with a E100P Digium card at a Deutsche
Telekom S2M port. (Asterisk 1.0rc2)
We are calling from a number in the same local area code and there
seems to be only the 6 most significant numbers of the target adress
arrive in Asterisk.
For example, we are
Hi there,
there is an option in zaptel.conf where you can configure a
prilocaldialplan to national, local, international and more.
What does that do?
I've tried to make sense of the source code, but I can't figure it out.
Regards,
Oliver.
___
Hi there,
I noticed, that in my other PBX, I can (and have to) set my local area
code and the master number (in our case a local 5 digit number that
prefixes the extensions) of our PRI EuroISDN line. I imagine that this
has some sort of use to those softwares. Is there a place where I have
to
hi Flynn,
I have an OpenLine4 on my setup. Everything appears to work finw and
I am not having the hangup detect but I am having problems when
voicemail tries to record via vpb channel. Did you ever have that on
your OpenLine4?
I have not tried out the OpenSwitch12 but I am a bit scared with
Are there any codecs that are particularly good for fax traffic? Any to avoid?
---
Eric Jacksch
[EMAIL PROTECTED]
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
Hello
I´ve just install last cvs version (Mon Sep 6) of Asterisk with asterisk-oh323-0.6.3b and pwlib-v1_6_6-src.tar.gz, openh323-v1_13_5-src.tar.gzand .
this is the error loading asterisk with chan_oh323 module::
[cdr_csv.so] = (Comma Separated Values CDR Backend)[cdr_manager.so] = (Asterisk
On Mon, 2004-09-06 at 17:09, Eric Jacksch wrote:
Are there any codecs that are particularly good for fax traffic? Any to avoid?
Google, google, google google.
http://www.google.com/search?hl=enie=UTF-8q=fax+codec+site%3Alists.digium.com
please exert effort before sending a question to the
Hi there,
the X-Pro SIP phone has up to six lines. Is there a way to have 4 of
them ring at the same time if there are 4 people in a Queue?
The idea is that we don't want an auto-voice system so our agents do
need to see if there are multiple people in the queue and be able to
handle more than
On Mon, 6 Sep 2004, Oliver Breidenbach wrote:
We are calling from a number in the same local area code and there
seems to be only the 6 most significant numbers of the target adress
arrive in Asterisk.
For example, we are calling 9123 and the CLI shows only the 91
and tries to
On Mon, 6 Sep 2004, Oliver Breidenbach wrote:
there is an option in zaptel.conf where you can configure a
prilocaldialplan to national, local, international and more.
What does that do?
That is what the a subscriber number (more or less caller id) is
marked as when it is sent in the setup
Hi,
I just upgraded asterisk from 0.7 to latest CVS-head.
Now the indications (ringing, busy, ...) are a horrible
noise.
What went wrong?
Roger.
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Peter,
thanks for trying to help.
I've enabled overlap dialing with no effect.
pri intense debug span 1 gives this output:
Informational frame:
SAPI: 00 C/R: 1 EA: 0
TEI: 000EA: 1
N(S): 102 0: 0
N(R): 105 P: 0
41 bytes of data
-- ACKing all packets from 104 to (but not
Yes, thanks for the comment, and I did configure the sip.conf for the Sipura -- sorry
for the confusion. The reference to iax.conf is because I am running FWD through IAX
so I would need to configure iax for those connecting through FWD, wouldn't I?
Roger
Update:
editing channels/chan_zap.c and setting #define DEFAULT_CIDRINGS 2
and recompile seems to have fixed the problem although it still shows
only the higher 6 numbers in the CLI console...
Very, very, very esoteric.
Cheers,
Oliver.
On 07.09.2004, at 01:30, Oliver Breidenbach wrote:
Peter,
Eric Jacksch wrote:
Are there any codecs that are particularly good for fax traffic? Any to avoid?
---
Eric Jacksch
[EMAIL PROTECTED]
See http://www.opencall.org/faq
Regards,
Steve
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[EMAIL PROTECTED]
Stewart,
Checkout the netweb-301 and 302 available on ebay now and also from Netweb
Group. This Phone is now available for $79.99 on ebay
Use the links below:
http://cgi6.ebay.com/ws/eBayISAPI.dll?ViewSellersOtherItemsuserid=netwebgroupinclude=0since=-1sort=3rows=50
Chris Lee wrote:
I am looking at building an IVR product with a few interesting
features and need some more information about how asterisk and VoIP
work and what I can get from them.
As far as I can tell when I use ISDN/GSM telephone networks the DTMF
information travels as data representing
Ive been searching the archives for the proper Wait
for Dial tone command in the Dial(Zap/g1/18005551212)
dial sting. Does anyone have an
example of its use?
Arick
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[EMAIL PROTECTED]
On Mon, 2004-09-06 at 16:34, Oliver Breidenbach wrote:
We are calling from a number in the same local area code and there
seems to be only the 6 most significant numbers of the target adress
arrive in Asterisk.
Try pridialplan=unknown
--
Eric Wieling * BTEL Consulting *
On Mon, 2004-09-06 at 19:18, Arick Davis wrote:
Ive been searching the archives for the proper Wait for Dial tone
command in the Dial(Zap/g1/18005551212) dial sting. Does anyone
have an example of its use?
There isn't one. However there is a wait .5 second. However it only
works on ANALOG
SeshKanuri wrote:
Checkout the netweb-301 and 302 available on ebay now and also from Netweb
Group. This Phone is now available for $79.99 on ebay
Use the links below:
http://cgi6.ebay.com/ws/eBayISAPI.dll?ViewSellersOtherItemsuserid=netwebgroupinclude=0since=-1sort=3rows=50
Hello,
After poking and prodding at Asterisk and Zaptel for over a couple years
now, I've dedicated some time to actually reading the code and trying to
figure it out.
It's been fascinating. With the driver source on one part of the screen
and a pdf of Linux Device Drivers on another part I've
Last week I was able to do some debugging of the problem I'm having with
IAX2/GSM, residential-grade broadband, and VOIP.
To summarize, I am having a great learning experience with * and Zap cards,
SIP and IAX2. I hit a wall though, when I registered with iaxtel and tried
doing VOIP.
I spend
I use Polycom IP 500's with Asterisk. These phones have 3 line appearances
and excellent full duplex speakerphones. They work very well with Asterisk,
and I was able to use the Web interface to set them up quite easily. The
default Web password given at voip-info.org is wrong, I added a comment on
Bob Knight wrote:
I have MIBs for whatever version I am running that I am more than
happy to share. Anyone know where I can place these for public access.
Sort of like the freedomphones site for Polycom. We could then
put pointers on the wiki.
Thanks for the info tho. If mbrowse is console
Thanks, Tony, you answered a question about g729 licencing and * conferences that I
wanted ask. Very enlightening. I was wondering about that because it seemed to be
using a license for each connections, despite the Sipura natively supporting g729, but
I wasn't sure that that is the way it has
Can anyone tell me how I can implement the features added in
the following link for call transfer? The authors seem to feel they are finished
but it doesn't appear to have been integrated into what everyone can download.
It is referred to as a patch but I don't understand how it could be
Roger,
at no point did I say I was finished with this patch. I did get a
little frustrated early on in the development. Currently this patch is
broken due to recent changes in cvs, and I'm about to tag it with a
post-1.0 tag in the bugtracker since there seems to be lots of interest
in
You could add an extension to your default context that takes you to VM:
exten = 500,1,VoiceMailMain
exten = 500,2,Hangup
Simply include the default context in your incoming context
include = default
Roger
From: [EMAIL PROTECTED] on behalf of Larry Shields
Wow, you guys are fast. My apologies, twisted. I realize there must have been a reason
why it wasn't marked resolved and included in the CVS HEAD, but I was under the
impression that those who wanted to and have the knowhow could download and apply the
patch. Didn't mean to imply you or anyone
Hello !
I try to use on the same server E100P + TDM40B without success.
When i add one one of them, no error on module load.
But when both are active, zaptel see strange configuration.
Here is my configuration :
zaptel.conf :
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
fxols=32-35
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