Hi guys,
I have one serius problem, some time our customers IP
Phones are not able to register, when i start to
geting the following logs.
WARNING[30665] channel.c: Avoided initial deadlock for
'0x9106ef8', 10 retries!
I am usuing realtime for sip registration the ttl of
phone is 10 or 20.
Hi
all,Do you monitor call quality ?If positive, how do you proceed
?
How do you
estimate user's experience from rough lattency, MOS, throughput and so on
?
Which
issues (echo ? call interruption ?) do you prevent with such monitoring
and which conter-measures do you engage when a problem
Don't forget the CAPI-based cards. I'm very happy with my
Eicon DIVA Server V-BRI and chan_capi from sourceforge.
Haven't had any problems or hiccups from day one after
creating the initial setup with my German ISDN line.
Eicon DIVA cards rocks but a quad-BRI costs 1500€
Other models cost
Mimmus wrote:
Eicon DIVA cards rocks but a quad-BRI costs 1500€
Other models cost this price/100.
Or not?
A Junghanns quadbri is approx 640€.
And 2x HFC-pci ISDN card is 2x30€ or so.. haven't tried this in
production yet :)
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Hi,
Is there something wrong with REALTIME (ARA) when used with
rtcachefriends parameter?
In my sip.conf (Asterisk 1.2.0):
rtcachefriends=yes
rtupdate=yes
rtautoclear=yes
Desired configuration is realtime configuration (via odbc) for SIP
phones + MWI. Realtime means the following: when I make
On Wed, 18 Jan 2006, Mimmus wrote:
Don't forget the CAPI-based cards. I'm very happy with my
Eicon DIVA Server V-BRI and chan_capi from sourceforge.
Haven't had any problems or hiccups from day one after
creating the initial setup with my German ISDN line.
Eicon DIVA cards rocks but
Hi, all. I've got a fax extension in my extensions.conf, but spandsp
never sends my faxes there. Both applications -- txfax and rxfax -- are
registered by Asterisk, so they compiled and installed correctly. I've
got a Sangoma A104 card, and (as some people had suggested) have loaded
ztdummy.
Hello,
I'm using asterisk-1.0.8 with BRI and spandsp-0.0.2_pre20.
Modules app_txfax.so and app_rxfax.so are compiled and loaded sucessfully.
It seems that the channel cant't detect the call as a fax-call
7612022801 is the calling faxmaschine
1209259 is my recieving fax extension
logs are:
Greetings to all,
I am getting the following error on my PRI-connected Asterisk box, and am
just wondering if anyone else has seen this, and if so how they solved it.
Jan 18 10:10:23 NOTICE[6070]: chan_zap.c:7428 pri_dchannel: PRI got event:
HDLC Bad FCS (8) on Primary D-channel of span 1
Jan 18
Hi
all.
I tried to build a
pattenrmatching for a numberrange but the asterisk won't hear on
it:
_49892351207[6-7][0-9]
if i make
a:
_4989235120760
all is
fine
Somebody has a hint
fo rme?
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Greetings to All,
I hope that someone can give me some guidelines with regards to CPU
utilization, and at what level CPU utilization begins to effect call
quality.
Thank you in advance,
Joe
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On 16 Jan 2006, at 11:05, Xavier Gil wrote:We run a asterisk 1.2.1 on a HP Proliant ML310, PIV 3Ghz 2 Mb L2 cache, 1 Gb Ram. We have a TE210Pdigium card configured for E1. This pbx has been running for almost a moth before giving this problems, we have called our telcoand seens that in their side
Hi All,
I have installed Asterisk and able to create Users and get them connected to
Asterisk after authentication. My question is how can I make calls to different
DIAX clients through my Asterisk server. I also have vonage softphone account,
using that I tried calling 18882255322
--
In article [EMAIL PROTECTED],
Dan Austin [EMAIL PROTECTED] wrote:
One of the features that I thought would be popular with the Web-MeetMe
suite is the ability to start all non-admin callers in a muted state and
selectively unmute them. For example any large conference that is
of an
Thanks much, questions about rj11, 2.5mm and quick connector wiring
abound in google search results but to date I hadn't found an answer.
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Quoting Jean-Michel Hiver [EMAIL PROTECTED]:
I don't think there is any way around this problem.
This is more a question of the terms of the agreement between both parties as to
what happens if a particular number was matched by a prefix not listed in the
providers A-Z.
A provider must list all
Hi,
Came across this gem tonight - quite funny.
I remember a recent thread about not enough Asterisk work out there so I thought I'd bring this to the list's attention.
Regards,
-- Nick
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Hi
terry ,
I am
an indian. Can u entertain my request , if so , i can send you my
resume
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Chris Earle (CBL) wrote:
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
The AVM
Ok, thanks, it works for me.Regards,YrvingDovid Bender [EMAIL PROTECTED] escribió: If you are new I would reccomend using [EMAIL PROTECTED]http://asteriskathome.soundforge.net . It is a greatresource for beginers. Also get the book (again I donthave the URL if some one does please post
On 1/17/06, steve [EMAIL PROTECTED] wrote:
Message: 20
Date: Mon, 16 Jan 2006 00:18:18 +
From: Mike Hemstock [EMAIL PROTECTED]
Subject: [Asterisk-Users] Choosing an FXO card
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
Thanks. I know that line quality is a factor, and I know I could get a 50$
fax with a PSTN line (that is what I have now). But I have my reasons to
want to setup a fax over IP, and I want to keep going. Where do I find info
on this debug mode? Is there a detaild log in Asterisk that show
Thanks for the patch, I'll give him a try.
Did you use the patch for yourself?
How does your system fetching the peers from db on startup?
Did you use the stupid dummy entries in sip.conf ;-)?
Thx,
reto
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im
Hi I need to authenticate all the asterisk users from the LDAP server instead of from sip.conf.If anybody already have done this then please guide.I tried to integrate authenticate asterisk users from LDAP using the open source project
astirectory1.2.0.After using the astirectory1.2.0 , now when
On Wed, 2006-01-18 at 11:05 +, Tony Mountifield wrote:
[snip]
I reworked the muting logic, and changed MeetMe so that an 'l' flag meant
listen-only (like the current 'm'), and an 'm' flag meant initially-muted.
I also put in Manager Events to inform when a user was muted or unmuted.
I
On Wed, 2006-01-18 at 07:12 +, Paul Redstone wrote:
Hi
Running bristuffed 0.3.0-PRE-1f which is 1.2.1.
Using *2 in features.conf for attended transfer. Works well if someone
answers.
But the following sequence causes issue:
1. Receptionist takes call.
2. *2 then 123 to
On Tue, Jan 17, 2006 at 06:07:27PM +0100, Karsten Wemheuer wrote:
Did I forget something in my conversion command?
Are You using bristuff 0.3.0-PRE-1f?
Yes.
I've had the same issue. Dan Austin
wrote a notice in a mail on this list, which solved the problem.
Configure the following
hi there,
I've been using sms a few months ago with * v1.0.9, but now I need it,
so I'm testing it out again. But for some reason the SMS receiving
doesn't work like it should.
It receives the call from the telecom operator, and it starts the SMS
application, but then i get the following
Hi to all.
I have a fritz card isdn C4, i have troubles to setup in asterisk server,
anybody have the detailed information about install this card???
Thanks to all
Vladimir
- Original Message -
From: John Daragon [EMAIL PROTECTED]
To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk
about this!
how many fxo clone card or x100p original card work in a 1 only box?
Thanks in advance!
Vladimir
- Original Message -
From: Mike Dent [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January
(continuing topposting for readability)
Frankly having to know RFCs intimately sounds a bit over the top to me.
I took the RHCE exam which is seen as one of the toughest exams in the
industry and never had to study RFCs. I don't know any other exams that
require you to do so (based on experience
I have a problem with SIP on my * box.
The * server with a private IP address is behind a NAT
modem-router with a public one.
I try to connect to a SIP provider which has a *
server with public IP but it doesnt works.
When I try making a call, the provider answers to the
SIP INVITE with
On Tue, 2006-01-17 at 18:21 -0500, Andy Green wrote:
Say there!
This is a new Asterisk install on Fedora; I downloaded and installed
Asterisk 1.2.1 and it runs fine. I downloaded the 20051217 version of
chan_sccp, compiled and it installs ok.
When I start Asterisk, it dies on startup
On 1/18/06, Vladimir Montealegre [EMAIL PROTECTED] wrote:
about this!
how many fxo clone card or x100p original card work in a 1 only box?
Thanks in advance!
Vladimir
I dont think you should be using more than 2 of these? I'd be pleased
to hear from others using more than 2?
Check out the
On Wed, 2006-01-18 at 15:07 +0100, amaury BOSSE wrote:
I have a problem with SIP on my * box.
The * server with a private IP address is behind a NAT modem-router
with a public one.
I try to connect to a SIP provider which has a * server with public IP
but it doesn’t works.
When I try
What do you use to administer the PBX accounts? (Please don't tell me LDIF
text files). Any luck with Luma? (http://luma.sourceforge.net)
__
Mensagem enviada usando o Webmail da ViaLink ver. 2.7.8
Previously, when I wanted to forward to incoming callerid when I
forwarded a call to another number I had to set the callerid on the
outgoing call to be that of the incoming number. So today I do this:
exten = s,n,Set(CALLERID(name)=${CALLERIDNAME})
because I want the outgoing callerid that
On Tue, 2006-01-17 at 22:07 +0200, Zoa wrote:
It took a little longer then expected, but here it finally is, a field
test for the idefisk for linux iax2 softphone.
Great news. I'm going to test it on my Debian Etch box.
Freely downloadable from http://www.asteriskguru.com/tools/
You
Hello everyone,
I have tried finding on forums, archives, etc. for this problem and have
found nothing related to the issue. I have an asterisk server running
Fedora Core 4. It has a public IP address and our phones will be behind
a NAT firewall. Right now, I have two soft phones, one on my
Mark Hulber wrote:
exten = s,n,Set(CALLERID(name)=${CALLERIDNAME})
This could never have accomplished anything, since those two references
affect the exact same variable internally.
because I want the outgoing callerid that I forward to not be the normal
callerid of the local extension
No, actually we dont use asterisk realtime, our system saves in file
the extensions and executes sip reload and other funny stuff
everytime a new extensions is added or modified. The patch was a quick
dirty idea that just came to my mind in the moment i saw your post
because i think in the future
Is there a good Web Conferencing add-on or a compatible package for
Asterisk? I know there are web based controls for the audio, but I am
looking for PowerPoint or desktop sharing functionality similar to WebEx.
Anyone using a package they like?
___
Peter Svensson wrote:
On Mon, 24 Jan 2005, Andrew Kohlsmith wrote:
As far as integrating with a website or database -- that is a piece of cake.
Your backend logic just determines when a call is needed and gerates the
approprate .call file. Just remember to create it in /tmp
Hi Wes, no there isn't.
I tried to fund a bounty for something similar about a year ago with no
takers.
I ended up having a Macromedia Flash Media server system developed
instead. We've made a decision to market this development.
It will retail for $2,000 including licences (Macromedia $450 for
Your patch didn't work, because method invocation 'realtime_user(cat)' is
not equal to the method signature ;-)
You are right, the use of a php script is not very smart because we plan to
setup a system for many thousand users.
I also take a look at chan_sip.c to get it fixed but my c programming
On 16 Jan 2006, at 11:05, Xavier Gil wrote:
We run a asterisk 1.2.1 on a HP Proliant ML310, PIV 3Ghz 2 Mb L2
cache, 1 Gb Ram. We have a TE210P
digium card configured for E1.
This pbx has been running for almost a moth before giving this
problems, we have called our telco
and seens
I want to set up a dial rule like this
9304752#w9#w+NX
The point of this is. It will dial into a pbx with the account number
9304752, wait a second, dial 9 to get an outside line, wait a second for
the outside line, and then dial the number to be called. When ever I
save this in amp
Does Asterisk has any way to detect when the destine grabs up the phone,
via SIP protocol. I need this in order to start a radius event.
If this can't be done, I would like to know is there is any extension,
or software that can accomplish what I need.
Thanks, in advance..
Wile
On Wed, 18 Jan 2006, Wile wrote:
Does Asterisk has any way to detect when the destine grabs up the phone,
via SIP protocol. I need this in order to start a radius event.
If this can't be done, I would like to know is there is any extension,
or software that can accomplish what I need.
On Wed, 18 Jan 2006, Michael Sampson wrote:
I want to set up a dial rule like this
9304752#w9#w+NX
The point of this is. It will dial into a pbx with the account number
9304752, wait a second, dial 9 to get an outside line, wait a second for
the outside line, and then dial
I'm creating the 1.call file in another directory
(/var/spool/asterisk/outgoing/tmp1), then moving it to the
/var/spool/asterisk/outgoing directory. I have another putty session
running on asterisk and logged into the asterisk console, but see no
activity after the file is copied in
On Wed, 18 Jan 2006, Kevin wrote:
I am able to call between all the phones and talk to anyone that answers
it, hardly any problems that can't be corrected by tweaking a few
things. The problem I am running into is sound, for say the demo, voice
mail, etc. If I program say a beep, or
You can't *copy* the file into the outgoing directory. You
must *mv* it.
First, remember also to check that call file is owned by the asterisk user.
Mimmus
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Sorry if Ive
missed the bus, but was enough interest gathered, and has anyone managed to put
together a UK voice file collection?
Cheers,
Adam
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On Wed, 18 Jan 2006, Wile wrote:
Does Asterisk has any way to detect when the destine grabs up the phone,
via SIP protocol. I need this in order to start a radius event.
If this can't be done, I would like to know is there is any extension,
or software that can accomplish what I
Joseph Rothstein wrote:
I am getting the following error on my PRI-connected Asterisk box,
and am
just wondering if anyone else has seen this, and if so how they solved it.
Jan 18 10:10:23 NOTICE[6070]: chan_zap.c:7428 pri_dchannel: PRI got event:
HDLC Bad FCS (8) on Primary D-channel of
On Wed, 18 Jan 2006, Patrick wrote:
(continuing topposting for readability)
Frankly having to know RFCs intimately sounds a bit over the top to me.
I took the RHCE exam which is seen as one of the toughest exams in the
industry and never had to study RFCs. I don't know any other exams
Hi,
look there http://snom.com/wiki/index.php/Xmlobjects
for snom 360...
Hirosh Dabui
[EMAIL PROTECTED] wrote:
What hardphones support xml/html/xhmtl/microbrowser? I need an
inexpensive SIP hardphone that can run simple applications (queue
status, etc).
The phones I know of:
Aastra 480i,
Hi All,
I installed Asterisk and trying to configure Vonage with it. After getting
authenticated when I try to call to a number I get the following errors
First I get
Sip read:
SIP/2.0 407 Proxy Authentication Required
CSeq: 104 INVITE
Proxy-Authenticate: Digest realm=216.115.20.41,
How hard would it
be to turn the Manager into an RSS feed if XML were an option? (stupid
example but you get my idea)
How hard is it now for a particular task, about 20 lines of code, in
less than an hour. Writing any sort of xml proxy to the manager
interface is really trivial for any
On 18:21, Tue 17 Jan 06, Andy Green wrote:
Say there!
This is a new Asterisk install on Fedora; I downloaded and installed
Asterisk 1.2.1 and it runs fine. I downloaded the 20051217 version of
chan_sccp, compiled and it installs ok.
When I start Asterisk, it dies on startup at the
Hi Ron,
Thanks for the reply. I used your config and still no upgrade. Using that file, the phone doesn't ask for the 7960-font.xml, but rather it just loops between the .tlv and the SEPMAC.xml. It never requests
OS79XX.txt. I'm starting to thing that contrary to what I've read, a blank
Viggiani Domenico wrote:
You can't *copy* the file into the outgoing directory. You
must *mv* it.
First, remember also to check that call file is owned by the asterisk user.
Mimmus
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Sorry for bad english
I installed asterisk and misdn by
svn co http://svn.digium.com/svn/asterisk/team/crichter/0.3.0 asterisk-0.3.0
but asterisk don't show any channel
and
/etc/init.d/misdn-init restart
---
Unloading module(s) for your misdn-cards:
Is anybody using 1.2 (or 1.2.1) in a production network using Realtime
(voicemail, sip or extensions) with 100+ SIP phones? If so, what are
your experiences? We've been running 1.0.3 for about a year and it's
been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm
afraid of
I think he is getting at something like a Zap channel that passes on it's
own CID info from zapata.conf, as opposed to the calling channel? Perhaps it
is a zap issue, and is as simple as placing callerid=asreceived in
zapata.conf.
OR
Maybe it is the way Dial() works in 1.2 versus 1.0 - with the
I do not see the '1' in front of the number you are trying to dial
Viggiani Domenico wrote:
You can't *copy* the file into the outgoing directory. You
must *mv* it.
First, remember also to check that call file is owned by the asterisk
user.
Mimmus
and here's the asterisk console output I get:
-- Attempting call on IAX2/voipjet/9529337367 for
[EMAIL PROTECTED]:1 (Retry 1)
-- Hungup 'IAX2/voipjet-3'
asterisk1*CLI
Your .call file now is OK.
You have some other problem but I cannot know what (registration
Hi,
I have a few of Atcom AT320 phones. I can choose if install SIP or IAX2
firmware.
Any suggestion?
Mimmus
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How hard would it
be to turn the Manager into an RSS feed if XML were an option? (stupid
example but you get my idea)
How hard is it now for a particular task, about 20 lines of code, in
less than an hour. Writing any sort of xml proxy to the manager
interface is really trivial for any
wath number of pstn lines you have? and how many extensions with soft
phones? and how many hardware phones you have?
wath is your hardware machine?
Thanks for the Reply!
Vladimir
- Original Message -
From: Peder @ NetworkOblivion [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
On Wednesday 18 January 2006 18:22, Peder @ NetworkOblivion wrote:
Is anybody using 1.2 (or 1.2.1) in a production network using Realtime
(voicemail, sip or extensions) with 100+ SIP phones? If so, what are
your experiences? We've been running 1.0.3 for about a year and it's
been rock-solid.
Hi all!
This is my VoIP network scheme
H323EndPoint ---- GW
H323/SIP-IN -- -- SIP Phone
||
(Sipquest) ||
|
| ||
|
| | |
H323EndPoint - GK1 GK2-|
|-- SER SIP Phone
||
| |
[EMAIL PROTECTED] wrote:
I'm creating the 1.call file in another directory
(/var/spool/asterisk/outgoing/tmp1), then moving it to the
/var/spool/asterisk/outgoing directory. I have another putty session
running on asterisk and logged into the asterisk console, but see no
activity after
On Wed, 18 Jan 2006, Javier Oviedo wrote:
Jan 18 18:06:17 NOTICE[16386]: chan_sip.c:11213 do_monitor:
Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11
seconds
Jan 18 18:06:17 WARNING[17340]: file.c:583 ast_readaudio_callback:
Failed to write frame
== Spawn
I had the horrible problems with a TE205P installtion
with 1 E1 connected to
the telco and the other E1 connected to a digital
Panasonic PBX.
I had frame slips, dropped calls, bad frames on
D-Channel, and I did 2
things to stop most of these problems:
First thing is suggested before:
you mean it did not compile? no time now to check, but i remember
realtime_user() receives a char pointer and returns a sip_user
pointer. cat is a char pointer and user is a sip_user pointer, it
should work tough. Well any way, i hope you can test my new patch in
the night or tomorrow.
If you
Anyone out there using small-midsized (2-4 TB) SAN solution among
multiple Asterisk systems? I don't have the budget for an EMC-caliber
solution, and can't seem to find much else out there.
Thanks,
Adam
The contents of this email message and any attachments are confidential and are
intended
hum. I tought that as long as you have a real time engine in
extconfig.conf the peers should be loaded at start up. What do you
have in your extconfig.conf?
Regards
On 1/17/06, Reto Kortas [EMAIL PROTECTED] wrote:
Hi all,
we use OpenSER together with Asterisk.
All SIP users registers with
Tony wrote:
I needed the same functionality. There wasn't a way to do it in the
current version of MeetMe. Also, the current muting logic is a bit
of a mess.
I concur.
I reworked the muting logic, and changed MeetMe so that an 'l' flag
meant listen-only (like the current 'm'), and an 'm'
I am posting only as an opinion, my comments are based upon my
observations and NOT on FACTS.
I remember that when the exam or certification was first announced. Many
were upset at how the exam was to be administered, whom was going to
profit, and most inportantly, who was going to write the
On Wed, 18 Jan 2006, Hirosh Dabui wrote:
look there http://snom.com/wiki/index.php/Xmlobjects
for snom 360...
nice... any hope for snom 320?
-Dan
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Hi All,
According to the wiki, we need to have both callwaiting=yes and
callwaitingcallerid=yes , and that's what I have in zapata.conf.
I can hear the call waiting alert tone when a 2nd call comes in during
an established call, and I can switch between the calls without
problems. However,
Hi,
Are you using
exten = fax,1,rxfax(. in extensions.conf
and
faxdetect=both in zapata.conf?
If yes, have you tried assigning an extension number for receiving
fax? (instead of the fax extension)
Andy
On 1/18/06, Ken D'Ambrosio [EMAIL PROTECTED] wrote:
Hi, all. I've got a fax
Christoph Eicke wrote:
On Wednesday 18 January 2006 18:22, Peder @ NetworkOblivion wrote:
Is anybody using 1.2 (or 1.2.1) in a production network using Realtime
(voicemail, sip or extensions) with 100+ SIP phones? If so, what are
your experiences? We've been running 1.0.3 for about a year
With most asterisk installs, there's no difference between an extension
and a phone number. For example, from your internal context, a phone
could dial one number and get an internal destination, or dial another
number and get an external destination. i.e. in my office I could tell
a
Hi
I wonder whether anyone got the Sipura ata 3000 to decode British
Telecoms callerid and pass it to asterisk?
The userguide seems to suggest that this is not possible, is that right?
Conrad
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On 1/18/06, Adam Robins [EMAIL PROTECTED] wrote:
Anyone out there using small-midsized (2-4 TB) SAN solution among
multiple Asterisk systems? I don't have the budget for an EMC-caliber
solution, and can't seem to find much else out there.
We used these guys http://www.raidzone.com/ about
In the past we have used an IPStor backend serving up LUNs over iSCSI to
intel and qlogic iscsi hba's directly on the asterisk boxes. It worked
like a charm.
I believe IPStor is fairly cost effective.
Ben
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I've seen there are spandsp, iaxmodem to simulate a modem for
receiving faxes under asterisk. Hylafax can be tricked to think it
is a real modem.
Can I create a dialup/access server with the same method? Can ppp run
on top of this software modem?
___
Hello,
I know that Mog was trying to get the bug-tracker cleaned up as
the number of bugs has increased substantially over the past few months. I
figured that I would do my part to bring attention to a couple of bugs
that are interesting and have some wide reaching impact. That being
Hello,
I just wanted to comment and ask for advice on how to fix a
DTMF recognition problem that apparently has been affecting asterisk users for
a long time now.
Duplicate digits are detected by Asterisk specifically when
using out of band tone recognition.
I've done extensive
Yep; chan_skinny is no-loaded... any other ideas? I'll try that other
mailing list, as well... but any help would be appreciated.
Andy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent: Wednesday, January 18, 2006 12:00 PM
To:
Hey there,
Just wanted to drop a line and let people know that I'll be
heading to San Francisco for O'Reilly's Etel. If you are interested in
attending, there are some free passes floating around. If anyone is
interested in getting together for a beer, let me know!
Info on the
Just got a Polycom 301 and I'm configuring. Examples given in wiki
recommend using dtmfmode=inband, so that's what I set in sip.conf for
this phone, as I have for various other IP phones on my network. But
the telephone does not seem to send DTMF tones up thru the network
(although I hear
SIP/localext
or ZAP/someothr
or providers SIP/1xxxnnn
On 1/18/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
With most asterisk installs, there's no difference between an extensionand a phone number.For example, from your internal context, a phonecould dial one number and get an
On Wednesday 18 January 2006 23:00, Conrad Wood wrote:
Hi
I wonder whether anyone got the Sipura ata 3000 to decode British
Telecoms callerid and pass it to asterisk?
The userguide seems to suggest that this is not possible, is that right?
Conrad
Check this
On Wed, Jan 18, 2006 at 01:31:33PM -0800, Frank Liu wrote:
I've seen there are spandsp, iaxmodem to simulate a modem for
receiving faxes under asterisk. Hylafax can be tricked to think it
is a real modem.
Can I create a dialup/access server with the same method? Can ppp run
on top of this
Frank Liu wrote:
I've seen there are spandsp, iaxmodem to simulate a modem for
receiving faxes under asterisk. Hylafax can be tricked to think it
is a real modem.
Can I create a dialup/access server with the same method? Can ppp run
on top of this software modem?
IAXmodem and spandsp
Hello,
I have a problem with an LAN-Server behind an NAT-router.
Asterisk Version 1.2.1 or 1.2.2 doesnt matter
10 minutes after starting Asterisk I loose all registrations at external
SIP-proxys.
The reason seemed to be that Asterisk send every second an request to every
sip-proxy Request:
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