Thanks. I have a question for the mailing list in general. Where
should the card get marked as in use? Should it be as soon as you enter
the number or should it be when it dials? I don't know for sure.
Darren Wiebe
[EMAIL PROTECTED]
Michael K. Rodriguez wrote:
This is my debug with the
Does the incomming call context in extensions.conf always have to be
[default]?
Can't i define different context for incomming like i can for Outgoing
in the sip.conf? My default conf is getting very large.
Regards,
Arne morten
___
--Bandwidth and
Hi all,
im going to install asterisk with a 4 BRI (HFC
chipset) on a Celeron at 2.6 GHz
I dont known Celeron performance, but i listen
that is not very good.
Could I have some performance isuue with this kind of
processor ?
Thanks for all
Giordano
Arne Morten Johansen wrote:
Does the incomming call context in extensions.conf always have to be
[default]?
Can't i define different context for incomming like i can for Outgoing
in the sip.conf? My default conf is getting very large.
In sip.conf, you can set context in a few places:
In
Hi all,
A small question relating sjphone
Here it is
I am connecting from pc to Asterisk using Sjphone.
I can make outgoing calls according to dial plan setup, but I am not
able to enter options asked during the call like enetering passwords
for voicemail.
SJ phone initiates just another
...or test the PickUpChan command coming with the bristuff-patch from
zapata
Damian Funnell wrote:
Hi,
Does anyone have remote call pick-up working on * (either via SIP or
otherwise)? If so then can you post your features.conf, sip.conf
and/or
zapata.conf?
We can't seem to get
http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+reverse+hold
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi list,
I set up two asterisk servers , 1001
is the first asterisk servers sip user, and 2001 is the second asterisk
servers sip user. Each of them work well, but I don't konw how to
connect them. I want to let the user in 1th Asterisk can call the user in 2nd
Asterisk.
First
What I can configuration my card Junghanns QuadBri?
Where I can download drivers?
Thanks?
___
Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB
http://mail.yahoo.it
___
Hi,
I try to set up planet VIP-050 with asterisk. Everything works fine
instead of the call transfer. When I press # console says something
like this:
Oct 5 11:11:20 DEBUG[25104]: chan_sip.c: sip_rtp_read: Oooh,
format changed to 1024
Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144
hi,
I advice u to use IAX to connect 2 asterisk more better and less bandwitdh. with following config u can do. just re do for sip.
(serverA)iax.conf[general]register = username:password@serverB hostname or IP[serverB]type=frienduser=usernamesecret=password
host=serverB hostname or
Anyone run app_txfax and app_rxfax successfully on asterisk 1.2 beta1?
I get an error when trying to run asterisk:
[app_txfax.so]Oct 5 12:05:24 WARNING[14665]: loader.c:314
__load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol:
fax_set_header_info
Oct 5 12:05:24
Hi,
Junghanns drivers can be found on Junghanns site, they are named
something like BRIStuff.
Giorgio
Fabio Montemaggiore wrote:
What I can configuration my card Junghanns QuadBri?
Where I can download drivers?
Thanks?
this was from Ian at digium:
I have talked to some hardware guys around here, and they said that the
rev H cards might still show up as rev E/F.
The bottom line is that you are having problems with it though, so we
need to take a look at it to see if there are any software-related
problems
trixter http://www.0xdecafbad.com wrote:
Anyone thinking about doing a VoIP business may want to get more info
before proceeding since they may not have the millinos vonage has to
fight this.
Unless of course they don't live in the United Sue'ers of America.
:D
--
Cheers,
Matt Riddell
Yes,
we have an applications that needs to detect the actual answer of the call not
when it is ringing.
CCF
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Angus
ComberSent: Friday, September 30, 2005 19:18To: Asterisk
Users Mailing List
Hi,
Were trying to use TDM04B with a
few analog switches and weve noticed that it works with the tones from USA
only. As its documented saying that those tones are hard-coded in the
source for analog cards.
Wed like to know if theres
anyone who could tell us under which file these
OK, here goes my next problem.
I have puchased a DID which I can connect to via SIP
I have been given the following details:
Username: uka1xx
Password: 1000xx
Server: br.net:5160
My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT)
The other end is a Cisco AS5300 (NO NAT)
Hello
I am starting to learn AGI. I have setup an extension to play the
agi-test.agi perl script and the output I get is this on console:
On Polycom 300:
-- Executing Answer(SIP/200-72d2, ) in new stack
-- Executing AGI(SIP/200-72d2, agi-test.agi) in new stack
-- Launched AGI Script
I have the same problem, after about a month the card doesn't report anyincoming calls anymore to the console. Don't know the rev of my card yet,
unloading asterisk and unloading the modules and then restartingeverything does seem to help though, no need to
Could you not just ignore the first answer and
watch out for the answer when the remote end picks up?
Angus
- Original Message -
From:
Chee
Foong
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, October 05, 2005 11:35
AM
Subject:
Where I am situated, our telephone lines are actually extensions (analogue,
2-wire) from a group of Mitel SX-2000 LITE PBXs
I have 2 extensions, 1 of which I have connected to a TDM11B for incoming
and outgoing calls using Asterisk.
Is it possible to configure an SX-2000 LITE to provide
I have the following pair of macros defined to handle outbound calls from *.
Rather than specifying full dialstrings in the main body of extensions.conf,
outbound dial commands are made using a macro call as follows:
Macro
On Tuesday 04 October 2005 18:04, Anthony Rodgers wrote:
I found the best reference to be the SoundPoint IP / SoundStation IP
Admin Guide - SIP 1.5 from the Polycom web site -
http://www.polycom.com/common/pw_item_show_doc/1,1276,4349,00.pdf.
You're right - that admin guide is much more
Hi.
I've started working on a PHP-project that generates the configuration
files i need based on what's in my MYSQL database. I can add, delete and
edit users from the web. I can set up exactly the dialplan i need by
arranging the users in a firms and groups if needed. I've also set up a
java
I am getting ready to purchase my first Digium card to start experimenting with
Asterisk. Before I make my purchase, I wanted to make sure I'm not going to
have issues with these cards (need to see what the specs are on my box, 5V or
3.3V PCI ). I will be using Asterisk @ Home, so will be
Stephen Bosch wrote:
Since Colin Anderson -- in a previous thread -- asked the question about
whether ADSI was dead, I thought it was worth discussing.
Does anybody else have anything to add?
-Stephen-
I hope ADSI is not dead! We have 100 Aastra 390 ADSI phones with 20 of
them in service.
snip
written by Crystal Stream, Incorporated
Here is my extensions.conf file. Things have been left
out or changed to protect the innocent.
Why isn't it working when I call from the outside that
when I press 124 it repeats the menu and doesn't
initiate DISA correctly to dial out?
[general]
Try it out and let us know! J
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis
Sent: Wednesday, October 05, 2005
3:39 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Intel
Pentium Celeron
Hi all,
im
Hey Arne!
My project 'PhoneCALL' http://www.vecsector.com/phonecall does pretty
much the same thing as you are describing - stores the configs in mysql
then submits the changes to flat files reloads asterisk on
completion. For me my clients - there hasn't been any noticeable
difference, as
Hi,
Anyone using inter Asterisk trunking IAX /IAX2 ?
Thanks,
Geo
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur
Brent,
what version of asterisk are you using?
Best Regards
Greg Cirino
Spam and Virus Free Email
included with every email account
Cirelle Enterprises Inc.
25 Indian Rock Rd #421
Windham NH, 03087
603-425-2221
Brent Franks wrote:
I have the same problem, after about a month the card
Doug wrote:
Hi,
Have looked around for info about this:
http://www.google.com/search?q=Transfer+directly+to+voicemail+site:lists.digium.com
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail
If we are using 5 digit extensions (10102: 10 for the company,
102 for the
trixter http://www.0xdecafbad.com [EMAIL PROTECTED] wrote:
Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing
on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP
technologies. Does anyone know which ones this article is talking
about, and if so does
I have configured my asterisk to connect to an H323 gateway in order to
place calls to the PSTN. The calls go through with no problem, but what I
experience is a loss of received sound after about 5 mins in the call (the
sound comes in very intermittent), while the other party continues to
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Enviado el: Martes, 04 de Octubre de 2005 01:31 p.m.
Para: Asterisk-Users@lists.digium.com
Asunto: [Asterisk-Users] DPH-140S SIP Phone oddities
Hi, list!
I'm playing on an [EMAIL PROTECTED]
Giordano Grandis [EMAIL PROTECTED] wrote:
(Article auto-converted from unnecessary HTML to nice plain text.)
i'm going to install asterisk with a 4 BRI (HFC chipset) on a Celeron at
2.6 GHz I dont known Celeron performance, but i listen that is not very
good.
Could I have some performance
Geo [EMAIL PROTECTED] wrote:
Anyone using inter Asterisk trunking IAX /IAX2 ?
No - you're the first to think of that. Congratulations.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h
_/ _/_/ _/ _/ _/
I'll jump in here with a couple of comments...
What you're trying to deal with is detecting answer supervision for the
outbound call, and not all telco's provide that. Since there is a wide
variation (in and out of the US), asterisk does not try to detect when a
call has been answered. It
Depending upon the patents in question, a few companies (Cisco comes to
mind) may have prior art here. I know that a company cisco bought was
doing VoIP in 1998, but no indications of which patents this is, or when
they were filed.
Paul
trixter http://www.0xdecafbad.com wrote:
Sprint
On Wednesday 05 October 2005 12:31, Roman wrote:
Anyone run app_txfax and app_rxfax successfully on asterisk 1.2 beta1?
I get an error when trying to run asterisk:
[app_txfax.so]Oct 5 12:05:24 WARNING[14665]: loader.c:314
__load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined
OK, here goes my next problem.
I have puchased a DID which I can connect to via SIP
I have been given the following details:
Username: uka1xx
Password: 1000xx
Server: br.net:5160
My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT)
The other end is a Cisco
I am trying to compile the astrisk-1.0.9 tarball on a RedHat 9 linux box
with dev environment. I get a lot of the following as a result of a make
/usr/bin/ld /usr/lib/crtn.o: invalid string offset 10 for section
`.shstrtab'
and final show stopper
./gentone busy 480 620
make[1]:***[busy.h]
Do you specifically need hardware echo cancellation?
If not you may want to try the TE405P/TE410P first and if you find that
you need the hardware echo-canceller you can upgrade the card for
$900(for some strange reason actually saves $100 over the list price of
the TE411P)
TE405P = $1495.00
I solved this problem by externpass to run a perl script that updates my
database.
Although I do believe that this should work through the built in ODBC
setup.
Ryan Hulsker
On Tue, 2005-10-04 at 17:16, Ryan Hulsker wrote:
I am using asterisk-1.2.0-beta1 with ODBC connecting to a MySQL
Dear Arne Morten
For me the best solution is to use MySQL. That is the reason we have
developed http://astbill.com. There is no performance issues. AstBill
is Open Source and for each SIP and IAX account you can choose if you
want to use Static confguration from config-files or Asterisk REALTIME
Rich Adamson wrote:
I'll jump in here with a couple of comments...
What you're trying to deal with is detecting answer supervision for the
outbound call, and not all telco's provide that. Since there is a wide variation (in and
out of the US), asterisk does not try to detect when a call
Would you consider a crash a month "very
stable"?
As for a version, 1.2beta 1 is very
stable, we've been using it on two high-volume production servers for over a
month now with only one crash in that time. I would recommend Asterisk
1.2beta1Hope this
helps,MATT---
On 10/5/05, [EMAIL
On 10/5/05, Don Pobanz [EMAIL PROTECTED] wrote:
Stephen Bosch wrote:
Since Colin Anderson -- in a previous thread -- asked the question about
whether ADSI was dead, I thought it was worth discussing.
Does anybody else have anything to add?
-Stephen-
I hope ADSI is not dead! We
Hi Matt,
Thanks for the response. I wasn't aware you could upgrade the card. In
that case, I think I'll make my boss happy by saving money and going with the
regular card first.
I'm basically checking out how well Asterisk works before putting into
production for our office. I figure
Hi all,
I'm using Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o on box A with a TE410P
(EuroISDN cpe)
connected to another similar asterisk box B acting as EuroISDN master.
I'm performing some load tests by contiously feeding up to concurrent 30
call files to /var/spool/asterisk/outgoing/ on box A
List users,
Please review this exchange between myself and Matt Florell. I am
looking for ANY data concerning Asterisk servers running standard Agents
and Queues. Hardware configurations, software configurations, Asterisk
configurations (especially the number of agents and queues), and
On Wed, Oct 05, 2005 at 03:51:38PM -0800, Geo wrote:
Hi,
Anyone using inter Asterisk trunking IAX /IAX2 ?
Thanks,
Not sure about IAX (1), but IAX2 is widely used. Before asking trivial
questions you probably should take the time reading about it in
http://voip-info/wiki-Asterisk and similar
The 3101 is in the same boat as the rest
of the 31xx and 2102B/PE series of 3Com phones. They are all SIP
Capable but currently only when used in conjunction with a 3Com VCX system.
Every time the phone boots up, it must download a runtime image from either a
3Com NBX or VCX system. The
One piece of advice though. Eliminate any echo as best you can before
introducing your PBX into Production. Once users have experienced echo
issues they will often continue to complain even after the issue is
corrected. It really can taint the overall preception of asterisk's
performance. Also
You know what.. I have sporadic echo issues too and I just checked my
dmesg and also see that! What's this all about?
*STOP*
You will receive these messages if you send or receive faxes. I asked for
this particular procedure to be executed because I was curious to see if
zaptel
Maybe you can buy this phone and extract the SIP
firmware. These phones pre-date the VCX system.
http://cgi.ebay.com/3COM-SIP-Phone-IP-VBX-Ethernet-Ports-Like-2102-1102_W0QQitemZ5815517891QQcategoryZ11909QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
- Original Message -
From:
Jared
Hi,
Does anybody know what configuration settings are required to setup
Asterisk for vonage?
Zeeshan A Zakaria
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Do you have a SIP phone account with them?On 10/5/05, Zeeshan [EMAIL PROTECTED] wrote:
Hi,Does anybody know what configuration settings are required to setupAsterisk for vonage?Zeeshan A Zakaria___--Bandwidth and Colocation sponsored by
Easynews.com
Lilantha, the tones are supposed to be switched using the loadzone and
defaultzone lines in /etc/zaptel.conf , and, progzone in
/etc/asterisk/zapata.conf.
The information about countries and frequencies/times are at
zonedata.c located in the sourcecode of zaptel. As you may know,
changing
Ah It was a typo. It should work now! L:)
--- Erik Slooff [EMAIL PROTECTED] wrote:
snip
written by Crystal Stream, Incorporated
Here is my extensions.conf file. Things have been
left
out or changed to protect the innocent.
Why isn't it working when I call from the outside
that
when I
On Wednesday 05 October 2005 12:22, Rich Adamson wrote:
Identifying why a echo cancel tone is occurring on a normal voice call
is reasonable, but why would a _local_ echo canceller be needed on a
four-wire full-duplex digital link?
It's to cancel far-end echo which is occuring on the
I found the
problem! Installing spandsp .3 created a symlink that was not
removed. Installing spandsp .2 did not replace the link. That cause
the wrong library linking in
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users
Hello. Has anyone run into problems accessing voicemail with the Sipura
SPA-2002's?
Our SPA-2000's work fine (registers fine, able to make and receive calls
properly also able to access voicemail). We've configured the 2002's
exactly the same way. However, with the SPA-2002 we're unable
On Tue, Oct 04, 2005 at 09:59:56PM +0200, Olle E. Johansson wrote:
I think this is a bug. Please open a report in the bug tracker,
attaching all the requested information. If a re-invite fails, we should
not cancel the call. I am afraid that is exactly what is happening here
and would like to
Michael,
Doing an All-Network setup is completely doable but there are many factors
to consider.
First of all, I didn't see any mention of how many connections it takes
before Asterisk starts having difficulty with DTMF. You mentioned that the
computer is directly connected to a T1, is it the
Hello. Has anyone run into problems accessing voicemail with the Sipura
SPA-2002's?
Our SPA-2000's work fine (registers fine, able to make and receive calls
properly also able to access voicemail). We've configured the 2002's
exactly the same way. However, with the SPA-2002 we're unable
Has anyone succesfully used TDMoE on Fedora Core 2.6.12+ Kernel versions?
I'm having the issue that is in the Mantis bug database with badness with
the kernel.
My Story:
I can get the dynamic span to come up and show OK in the zttool on both
machines. However i get errors every second (Warning:
Hi Andrew,
I'm using a TE406P too, and I have echocancel=yes in zapata.conf.
Is this redundant? Should I take the line out?
Please advice.
Thanks.
AK
On 10/3/05, Rod Bacon [EMAIL PROTECTED] wrote:
Which version of asterisk and zaptel are you using?Will they work with 1.0.9
Hello. Has anyone run into problems accessing voicemail with the Sipura
SPA-2002's?
Our SPA-2000's work fine (registers fine, able to make and receive calls
properly also able to access voicemail). We've configured the 2002's
exactly the same way. However, with the SPA-2002 we're
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich
Adamson
Sent: Wednesday, October 05, 2005 9:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Echo Canceling
{clip}
Identifying why a echo cancel
I'm also using Broadvoice and was having a lot of problems with DTMF.
I 'm in Ft. Lauderdale, FL and I was (inadvertently) using the dca
proxy. When I changed it to use the Miami proxy, my DTMF tones
started to work reliably
I had done some digging and found various posts on the internet where
Hi Asteriskers!
Ive enable CDR to store
data on a remote machine using MySQL. But I have a problem. Analyzing the log,
I see some ERROR messages as:
--
SIP/21-3787 is ringing
== Spawn extension
(default, 21, 1) exited non-zero on 'SIP/21-ce14'
Oct 5 13:22:54
ERROR[8576]:
On Mon, Oct 03, 2005 at 01:05:55PM +0200, Giordano Grandis wrote:
Hi,
just a question: anyone has never installed g729 codec on VIA
motherboard with C3 processor ?
I'm having problem with IPP libraries, and Intel said that it works only
on Inter processor.
Any suggestion?
Here you are:
- sip.conf
[]
type=friend
username=
secret=password
host=dynamic
context=sip-trusted
[EMAIL PROTECTED]
nat=yes
qualify=yes
canreinvite=no
- voicemail.conf
= ,Randy Vinas,[EMAIL PROTECTED]
- extensions.conf
exten = ,1,Macro(sip-stdext,,Bellcore-r2)
At 09:11 10/5/2005, Don Pobanz, wrote:
Doug wrote:
Hi,
Have looked around for info about this:
http://www.google.com/search?q=Transfer+directly+to+voicemail+site:lists.digium.com
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail
If we are using 5 digit extensions
Hey I just setup service with IPComms and they are
telling me to setup such as this:
iax.conf:
[IPCommsNet]
type=user
host=69.15.xxx.xx
context=voicepulse-in ;(changed by me)
nat=yes
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=gsm
When I'm calling once of my numbers it's giving me
I would be curious about what other members of this list think about
the best practices for giving clients functionality on their desk
phones.
See, I have a dirty little secret. One of the primary justifications that is
used for VoIP PBX is consolidated physical network - I mean, it's supposed
to
http://news.com.com/Sprint+Nextel+sues+Vonage+over+VoIP+patents/2100-7352_3-5888789.html?tag=nefd.top
Does anyone have any clue what the suit is over and if/how this
affects Asterisk's implimentation of VoIP?
___
--Bandwidth and Colocation sponsored by
SORRY! Duplicate... ignore this thread.
On 10/5/05, Matt [EMAIL PROTECTED] wrote:
http://news.com.com/Sprint+Nextel+sues+Vonage+over+VoIP+patents/2100-7352_3-5888789.html?tag=nefd.top
Does anyone have any clue what the suit is over and if/how this
affects Asterisk's implimentation of VoIP?
On Tue, 4 Oct 2005, Leigh Fereday wrote:
I upgraded to CVS, but get the same message in the log.
If you are on CVS-HEAD or 1.2, perhaps busypattern= will help you.
Call into your Asterisk box on one of the incoming analogue lines and dial
through to an extension. Whilst listening to the
Doug wrote:
Another method would be to prefix with a digit instead of suffix with an
*. For us, all of our extensions are three digits and begin with a 5
or a 6 (5xx or 6xx). To transfer to voice mail we stick an eight in
front of the extension (85xx or 86xx). It works well for us.
Don
On Wednesday 05 October 2005 19:52, Rich Adamson wrote:
Hello. Has anyone run into problems accessing voicemail with the Sipura
SPA-2002's?
Our SPA-2000's work fine (registers fine, able to make and receive calls
properly also able to access voicemail). We've configured the 2002's
On Wednesday 05 October 2005 17:46, Ricardo Poppi wrote:
Lilantha, the tones are supposed to be switched using the loadzone and
defaultzone lines in /etc/zaptel.conf , and, progzone in
/etc/asterisk/zapata.conf.
Also look at /etc/asterisk/indications.conf
Paul
--
Paul Hewlett -
Juanjo,
can you provide some more detail about which version you are using both for
asterisk and OpenH323, the hardware dimensioning and the amount of traffic
you manage with this solution: how many lines, codecs you use?
We should manage a full blown PRI (30 channels), the server is SuperMicro
this isn't working
[IPComms-in]
exten = s,1,Noop(${DATETIME} ${CALLERID})
exten = s,2,SetCallerID(${CALLERID})
exten = s,3,Answer
exten = s,4,Goto(main-menu,s,2)
exten = s,5,Hangup
What I have is a block of 20 DIDs and I want to accept
calls from all of them.
It would be way to freaking
it is trying to match the did in your context which it can't do
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hello,
We've released another update to our Asterisk GUI Client suite: 1.1.7
http://astguiclient.sf.net/
The client suite runs on Windows, UNIX and Mac, includes the
astGUIclient client-side web app which extends your phone's
functionality and the VICIDIAL client-side web app auto-dialer. This
Crystal Stream, Incorporated [EMAIL PROTECTED] wrote:
Hey
Moo.
I just setup service with IPComms and they are
telling me to setup such as this:
iax.conf:
[IPCommsNet]
type=user
host=69.15.xxx.xx
context=voicepulse-in ;(changed by me)
nat=yes
dtmfmode=rfc2833
disallow=all
Here you are:
- sip.conf
[]
type=friend
username=
secret=password
host=dynamic
context=sip-trusted
[EMAIL PROTECTED]
nat=yes
qualify=yes
canreinvite=no
- voicemail.conf
= ,Randy Vinas,[EMAIL PROTECTED]
- extensions.conf
exten =
At 2:43 PM -0700 10/4/05, trixter http://www.0xdecafbad.com wrote:
Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing
on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP
technologies. Does anyone know which ones this article is talking
about, and if
Hi!
If you have problems with MusicOnHold or run Meetme, please, gives this
patch a try as it might help you. If enough people test this, it will
potentially included in the upcoming 1.2 release.
Here's the info on Mantis:
http://bugs.digium.com/view.php?id=5374
Please, provide feedback on
Anyone aware if Digium or Sangoma, or possibly a function of Asterisk,
supports answering machine detection on an outbound call?
--
Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
454 Sonwil Drive
Buffalo, NY 14225
+++
tf voice -
I'm seeing the same behavior on a Debian system with 2.6.12.
I have two systems with Digium Quad T1s in each and I trunk them with
TDMoEThis always worked great on 2.4 and up to 2.6.8 but beyond that
it either spits out copious amounts of kernel badness and paralyzes the
system completely or
Hi Asteriskers!
I've enable CDR to store data on a remote machine using MySQL. But I have a
problem. Analyzing the log, I see some ERROR messages as:
-- SIP/21-3787 is ringing
== Spawn extension (default, 21, 1) exited non-zero on 'SIP/21-ce14'
Oct 5 13:22:54 ERROR[8576]:
Crystal Stream, Incorporated [EMAIL PROTECTED] wrote:
this isn't working
[IPComms-in]
exten = s,1,Noop(${DATETIME} ${CALLERID})
exten = s,2,SetCallerID(${CALLERID})
exten = s,3,Answer
exten = s,4,Goto(main-menu,s,2)
exten = s,5,Hangup
What I have is a block of 20 DIDs and I want to
I am using the inter asterisk trunking and the article in voip-info.org will not work.
On 10/5/05, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Oct 05, 2005 at 03:51:38PM -0800, Geo wrote: Hi, Anyone using inter Asterisk trunking IAX /IAX2 ? Thanks,Not sure about IAX (1), but IAX2 is widely
I have setup Asterisk with chan_oh323 to connect to the PSTN over H.323
in
--may i know which version of asterisk and oh323?
On 10/5/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Juanjo,
can you provide some more detail about which version you are using both for
asterisk and
try compiling with 586 and change the makefile to disable mmx codes (if any). I remember tohave this working on a few different processors, but forgot how I did it.
-apu
On 10/3/05, Giordano Grandis [EMAIL PROTECTED] wrote:
Hi,
just a question: anyone has never installed g729 codec on VIA
Is there a way I can have voice mail check calls coming from my internal
users automatically get to the right extension, without having the user
enter their extension?
I'm thinking that I could have the local SPA boxes translate, or have
each user live in a context where the extension in question
1 - 100 of 170 matches
Mail list logo
