Hi Gurus
I am trying to make asterisk load as a linux servics at boot time. I tried
both methods;
(a) /etc/init.d/asterisk
(b) /etc/rc.d/rc.local
But * failed to start.
What is interesting is the message log (attached below), in either case
problem is with chan_h323.so. Which is failing to
Hi,
I am using asterisk with PRI TE410P card. Everything work fine, except
that every time I receive a call, I get File chan_zap.c, Line 3546
(zt_read): Fax detected although they are just normal calls. How can i set
the threshold of fax detection. What might be wrong that tone_detect
I have about a dozen Budgetone 101s and I'm pretty much satisfied with them.
Sorry, Brian; they've got some rough edges, but they're $65, for God's sake.
B.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Wed, 2003-12-24 at 08:18, Brian Capouch wrote:
I have about a dozen Budgetone 101s and I'm pretty much satisfied with them.
Sorry, Brian; they've got some rough edges, but they're $65, for God's sake.
They are $65 yes, but you can get the best analog phones on the market
for that price and
So, you can get a really good analog phone for $65, then you mention
and use an ata... what does this ATA cost?
$65 for the complete set is what I pay for. At that price, I expect an
issue here and there. It is still getting the bugs worked out.
I don't have the money to buy $300 Cisco
hi guys
just got a question, im using grandstream phones with canreinvite=no or woteva, all
nat etc is working perfectly. but i believe because of the canreinvite, when a call
has taken place the voice will be proxied via the sip server to the 2 parties
involved. ( which means the sip server
Miguel Cavazos wrote:
On Wed, 2003-12-24 at 08:18, Brian Capouch wrote:
I have about a dozen Budgetone 101s and I'm pretty much satisfied with them.
Sorry, Brian; they've got some rough edges, but they're $65, for God's sake.
They are $65 yes, but you can get the best analog phones on the
It is unfortunate that Cisco is so damned expensive. $300 is only the tip
of the iceberg if you go the official route... You still haven't paid for
their ongoing maintenance. They should really consider selling their
phones at a better price.
cameron.
On Wed, 24 Dec 2003, Robert Hajime
vocalvoip wrote:
hi guys
just got a question, im using grandstream phones with canreinvite=no or woteva, all nat etc is working perfectly. but i believe because of the canreinvite, when a call has taken place the voice will be proxied via the sip server to the 2 parties involved. ( which means
Karl Putland wrote:
On Sat, 2003-12-20 at 03:22, Senad Jordanovic wrote:
Hi,
I am trying to figure out if * can register as a client on a remote
MGCP service. Just like SIP and other protocols Do. Anyone tried
this?
No I don't believe it can. The MGCP implementation in Asterisk is a
I know this is OT for this list, but I havnt seen it mentioned here and in
the spirit of 'open source' I thought this would be interesting for readers
here:
- Original Message -
From: Jeff Pulver [EMAIL PROTECTED]
Sent: Tuesday, December 23, 2003 11:28 PM
Subject: [FWD] FWD Holiday
hello everyone,
theres a bi-monthly computer fair in amsterdam on saturday and it looks
like a few asterisk users will be attending, and hopefully some more might
be able to turn up. admittedly this probably is a bad idea to advertise
because the more asterisk people the less likely i am to find
i tried with other softphones. the only phone thats
working with GS is Xtern. MSN and SJ doesnt work. Is this a known
issue.
Thanks,
-B
- Original Message -
From:
Balaji NJL
To: [EMAIL PROTECTED]
Sent: Tuesday, December 23, 2003 7:05
PM
Subject: Re:
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Grandstream Quality Survey :P
Reply-To: [EMAIL PROTECTED]
...
I have 2 of these phones and they work fine for my application. Granted
its not the most intensive use and definatly not the most critical users
On Wed, 17 Dec 2003, Victor Medrano wrote:
i did with cisco callmanager with smdi integration .
and h323 .
works very well .
You got SMDI working with CCM?
How?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
For the price, the Grandstream is unbeatable value for money.
Get firmware version 1.04.26 and you should be fine. This firmware fixes
issues our customers had with phone lockups, nat problems, one-way
audio, stun problems.
Best Wishes
Tan
www.telappliant.com
www.voiptalk.org
-Original
I've been having issues getting FWD to work. I posted this same Q to the
FWD forum (no responses yet), but I was hoping someone here had some insight:
http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi?board=news;action=display;num=1072263468;start=0#0
I just signed up for an FWD account (I know I
Hi Tan,
Can you supply us with 1.0.4.26 firmware?
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: 24 December 2003 12:53
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P
For the
On Wed, 2003-12-24 at 14:35, David J Carter wrote:
Hi Tan,
Can you supply us with 1.0.4.26 firmware?
http://www.grandstream.com/TEMP/FIRMWARE/
--
Dave Cotton [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Try it on one of the phones first. We've tested it and it seems to work
fine. Let me know offline how you get on.
http://www.telappliant.com/grandstream/1.04.26.zip
Thanks
Tan
www.telappliant.com
www.voiptalk.org
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
http://www.grandstream.com/TEMP/FIRMWARE/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J Carter
Sent: Wednesday, December 24, 2003 7:36 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P
Hi Tan,
Can you
On Wed, 24 Dec 2003, denon wrote:
I've been having issues getting FWD to work. I posted this same Q to the
FWD forum (no responses yet), but I was hoping someone here had some insight:
My setup is like this:
sip.conf:
register = 21542:[EMAIL PROTECTED]/6002 ; Free World Dialup
merry christmas to all.
sorry (but come the 1st it's a new year and therefore can
create a new
list to atone for) G
Happy Holidays Everyone!
May your uptimes be plentiful, and your core dumps be rare in this season of
hardware failure. And may those who still use the old school 'G'
I just loaded the b13p4.30.zip firmware and now I am not able to log
into the GS admin interface.. anyone else having this problem?
Going to try the next older version..
Later..
mikeu wrote:
http://www.grandstream.com/TEMP/FIRMWARE/
-Original Message-
From: [EMAIL PROTECTED]
I have 6 broken Grandstreams- out of an order of 8. After having tested
over a dozen IP phone products, I found that Grandstream was the worst
choice of the group.
I would never recommend that anyone buys this product unless they are using
it for non-essential use. To put it simply: Grandstream
Thanks!
And may your CDR's be longer and more profitable!!! :)
Merry Christmas everyone!
Panny Malialis
Hotlinks Internet Services
http://www.hotlinks.co.uk
- Original Message -
From: Tony Kava [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 24, 2003 2:30 PM
Subject:
On Wed, 2003-12-24 at 15:50, WipeOut wrote:
I just loaded the b13p4.30.zip firmware and now I am not able to log
into the GS admin interface.. anyone else having this problem?
Yep been there. Panicked, rebooted the phone and it responded as
normal. I just tried it again because of your
SW wrote:
(ast_load_resource): libpt_linux_x86_r.so.1.5.2: cannot open shared object
file: No such file or directory
Dec 23 23:33:52 WARNING[1074494176]: File loader.c, Line 407 (load_modules):
Loading module chan_h323.so failed!
RTFM
cat /path/to/asterisk/channels/h323/README
Jeremy
Well you certainly could. And you'd then have to add the cost of the
ATA to your cost per seat, at least doubling the $65 figure--tripling
it if you meant a Cisco ATA.
NOT, Cisco ATA's can be had fro 99-120 if you are lucky. Then you can
also get a cisco 7905 which can be had on ebay for
Cisco 7905's are damn fine phones for 99 bucks and they blow the
grandstream away...
bkw
On Wed, 24 Dec 2003, Cameron Palmer wrote:
It is unfortunate that Cisco is so damned expensive. $300 is only the tip
of the iceberg if you go the official route... You still haven't paid for
their
Unbeatable maybe... but also very unreliable.
On Wed, 24 Dec 2003 [EMAIL PROTECTED] wrote:
For the price, the Grandstream is unbeatable value for money.
Get firmware version 1.04.26 and you should be fine. This firmware fixes
issues our customers had with phone lockups, nat problems, one-way
Yep I have heard this many many times. Seems like they have a large batch
of phones that are bad.
bkw
On Wed, 24 Dec 2003, Greg Renouf wrote:
I have 6 broken Grandstreams- out of an order of 8. After having tested
over a dozen IP phone products, I found that Grandstream was the worst
Yes when you upgrade to beta code you may have to reboot 3 times for the
phone to function properly. Then cross your fingers that the phone will
accually register with * once you do that.
bkw
On Wed, 24 Dec 2003, Dave Cotton wrote:
On Wed, 2003-12-24 at 15:50, WipeOut wrote:
I just loaded
--On Wednesday, December 24, 2003 07:12:05 -0600 denon [EMAIL PROTECTED]
wrote:
I've got it running through Asterisk - all working fine from a SIP
standpoint. I can dial FWD numbers like 612/613/etc and everything works.
However, if I dial *18005551212 or *408xxx (say, a USA number), I
Brian West wrote:
Well you certainly could. And you'd then have to add the cost of the
ATA to your cost per seat, at least doubling the $65 figure--tripling
it if you meant a Cisco ATA.
NOT, Cisco ATA's can be had fro 99-120 if you are lucky. Then you can
also get a cisco 7905 which can be
Hey all,
We've upgraded our PRI trunk to support what BellSouth calls enhanced
caller id name delivery. The weird part is, I'm only capable of seeing
the name portion on incoming calls within voicemail2's email delivery.
For example, on an incoming call, asterisk is reporting this:
Context from
Brian,
Can you compare Ford and Mercedes or BMW? Both are cars and drives..
but you have different feeling and price in/for each car ..same here
Grandstream is low-cost solution for end-users/small business , Cisco IP
Phones are couple times more expensive ,but they have more features,
less bugs
I am trying to configure cdr on a system. We are using nufone and I have
set amaflags=billing on both of their sections in iax.conf. Incoming
nufone calls show up in cdr with billing, but outgoing calls still show
documentation. What do I need to change? We have a handful of SIP phones,
1
On Tue, 2003-12-23 at 19:22, Brian West
wrote: I'm just double checking.. I was told it wasn't possible but i'm
going to ask just in case. Can you set outbound
callerid on a channelized T1?
I think there is a way to do something like DID with the 4 digits
ofDTMF passed before the call. It
Message: 11
From: Asterisk online forums [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P
Date: Wed, 24 Dec 2003 11:23:14 -0500
Reply-To: [EMAIL PROTECTED]
Brian,
...
We are looking now to improve GS products and start collecting
hi,
i have been using grandstream budgettone IP phones
and they work fine except that these phones times out after some hours.. i ahve
seen that the phones working ok are next day unregistered and sip show peers do
not show their IP and although these phones can make calls , they cannot be
The phone powers up and I can make calls through my Asterisk gateway to
other endpoints. However the four leds under the keypad are permanently
illuminated and the backlight slowly flashes on and off. When I pick up
the handset there is a repeated tone before I get a dial tone.
I know it's
voicemail notification?
-Original Message-From: bam
[mailto:[EMAIL PROTECTED]Sent: Wednesday, December 24, 2003 12:17
PMTo: [EMAIL PROTECTED]Subject:
[Asterisk-Users] Grandstream 102 flashing displayThe
phone powers up and I can make calls through my Asterisk gateway to
What version of the BudgeTone software are you
running?
- Original Message -
From:
Chandra
To: [EMAIL PROTECTED]
Sent: Wednesday, December 24, 2003 12:09
PM
Subject: [Asterisk-Users] Grandstream
budgetTone registration time out
hi,
i have been using
The phone powers up and I can make calls through my Asterisk gateway to
other endpoints. However the four leds under the keypad are permanently
illuminated and the backlight slowly flashes on and off. When I pick up
the handset there is a repeated tone before I get a dial tone.
I know it's
At my home office I have a X100P card in a server that I've been using
for testing. The machine it is in is connected to a HP fax machine and
then to the wall outlet. This morning the SBC installer showed up at my
house for the ADSL install on that line. He said they detected a
short. So
Title: Leterhead
Mine does
that as a message indicator when mail is in the mailbox.
You get a
flashing display and a stuttered dial tone for the first few seconds.
Dave
-Original
Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On
Behalf Of bam
Sent: 24 December
bam wrote:
The phone powers up and I can make calls through my Asterisk gateway
to other endpoints. However the four leds under the keypad are
permanently illuminated and the backlight slowly flashes on and off.
When I pick up the handset there is a repeated tone before I get a
dial tone.
I
--On Wednesday, December 24, 2003 07:12:05 -0600 denon [EMAIL PROTECTED]
wrote:
I've got it running through Asterisk - all working fine from a SIP
standpoint. I can dial FWD numbers like 612/613/etc and
everything works.
However, if I dial *18005551212 or *408xxx (say, a USA
When moving from 1.0.3.x firmware to 1.0.4.x you must reboot
2 times :
first time for loading the new bootloader from tftp
second time for getting the 1.0.4.x firmware.
GS are ok for their price. but of course, you get
what you paid for. with 1.0.4.26 firmware I'm quite
happy, finally there's
Robert,
We are going to deploy GS phones in our free voice network, therefore we
require somekind of web-presence, which will reflect GS support,etc.
Unfortunately not all of our users are subscribed to Asterisk mailing
list.
Acting as GS distributors, we are making separate forum for this,
--- Chandra [EMAIL PROTECTED] wrote:
i have been using grandstream budgettone IP phones and they work fine except that
these phones times out after some hours.. i ahve seen that the phones working ok are
next day unregistered and sip show peers do not show their IP and although these
phones can
Title: Fax capabilities of various services
For the Vonage, Packet8, etc services, are they all able to handle fax machines on their little interconnect boxes?
Ray Burkholder
[EMAIL PROTECTED]
http://www.oneunified.net
704 576 5101
--
Scanned for viruses & dangerous content at
One
Hello. I'm a new Asterisk user, but I'm impressed with the
flexibility and versatility of Asterisk, and am moving quickly to adopt
it's main-line use in our company. Hopefully, you'll be hearing more from
me as the project moves forward.
Right now, though, I have a question about
Skinny phone functionality is 'richer' than SIP phone functionality. First
off, on a skinny phone, in hands free mode, you can start dialling and the
phone will automatically go off hook. Sip requires you to manually hit the
speaker button, hit new call, or pickup the phone before dialling.
Hello ...
I'm working with SCCP only phones (ie, Cisco 7910s) and happened to
notice that the chan_skinny driver seems to be missing some significant
features. Most, if not all, button features (STIMULUS messages) are not
implemented and callwaiting crashes the phone.
Has there been much
At 11:10 AM 12/24/2003, you wrote:
Skinny phone functionality is 'richer' than SIP phone functionality. First
off, on a skinny phone, in hands free mode, you can start dialling and the
phone will automatically go off hook. Sip requires you to manually hit the
speaker button, hit new call, or
On Wednesday 24 December 2003 13:06, Brian Buhrow wrote:
Hello. I'm a new Asterisk user, but I'm impressed with the
flexibility and versatility of Asterisk, and am moving quickly to
adopt it's main-line use in our company. Hopefully, you'll be
hearing more from me as the project moves
Open software seems to work.
Why don't we try it with hardware.
1. pick an embedded processor.
It should have a nice linux gui support (like x jtag debugger).
2. pick a linux based cad system we all have easy access to and place
schematics under cvs.
3. pick some type of gpio or serial
Dear All,
On behalf of XVOIP, LLC/Stealth Telecommunications, WISH YOU A
MERRY CHRISTMAS AND A HAPPY PROSPEROUS NEW YEAR.
Thanks to everyone for such great place as Asterisk community, for all
your answers, suggestions, time, examples, help.
We plan to support Asterisk project and
My phone's booted up and registered OK but a strange thing noticed on the
tftp uploads.
bootloader.bin
bt.bin
voc.bin
html.bin
vp.bin
ht.bin
The first phone uploaded the first four bin files.
The second phone uploaded the first five bin files.
Neither phone uploaded the ht.bin file.
Both phones
I had been thinking of doing this, but lack the
electronics expertise to do such a thing.
I basically need phones that look like trading
turrets, so I can sneak them into this one trading
firm.
Good idea, let's see if there's any traction.
-Original Message-
From: [EMAIL PROTECTED]
In the process of investigating a Cisco ATA 186 that was locked by
Vonage, I found that you can still unlock the device yourself. But
there's a catch.
The device's design has a great plus: a DIP32 *socketed* SST28SF040A
flash chip. I found an 8 digit unlock code at 0x03FA71-0x03FA78. I do
My Grandstream phone seems quite happy to accept a new firmware, but
having tried the latest beta firmware, which I am unhappy with I want to
update with an older version. How do I do this?
Thanks,
Michael
--
Michael T Farnworth
Maxima Systems Ltd (http://www.maximasystems.com)
16 Woodbourne
I asked Vonage about unlocking it and they refused to. They don't offer
an unlock service for $15.
Matt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lion Templin
Sent: Wednesday, December 24, 2003 2:53 PM
To: [EMAIL PROTECTED]
Subject:
--On Wednesday, December 24, 2003 6:35 pm +0100 Arnold Ligtvoet
[EMAIL PROTECTED] wrote:
Read the fwd announcement. Jeff Pulver mentioned the fact that * users
cannot use the free holiday calls, since FWD cannot be sure that * is not
being used by more than 1 user at the same time.
Where in
Hi,
Why do I get registration refused errors with Asterisk and
voip providers? I did everything correctly and every provider I signed up with
gives me that error:
Dec 24 15:30:13 NOTICE[-1254995024]: File chan_iax.c, Line
3955 (socket_read): Registration of 'in-STn46BoD89' rejected:
Still, there seems to be a you get what you pay for theme to many of
today's posts and this clearly applies to support on FWD. Naybe we should
remove the signature from * that enables FWD to identify * systems :-)
That certainly seems the case for today's theme... It is certainly the
right
Skinny phone functionality is 'richer' than SIP phone functionality.
First
off, on a skinny phone, in hands free mode, you can start dialling and the
phone will automatically go off hook. Sip requires you to manually hit
the
speaker button, hit new call, or pickup the phone before dialling.
Ray Burkholder wrote:
Skinny phone functionality is 'richer' than SIP phone functionality. First
Skinny *functionality* seems to be 'richer', but it's implementation in
* is woefully under-functional. Regardless of individual phone feature
sets, SIP is far better implemented in * than skinny.
--On Wednesday, December 24, 2003 10:06 pm +0100 rnc Info Lists
[EMAIL PROTECTED] wrote:
Still, there seems to be a you get what you pay for theme to many of
today's posts and this clearly applies to support on FWD. Naybe we
should remove the signature from * that enables FWD to identify *
Hi Patrick,
Citeren Patrick [EMAIL PROTECTED]:
I am trying to get Capi Dial to use a specific outgoing msn. I can't get
it to work. If I make a test call to 0703241494 (same isdn line, just
one of the other numbers) I don't get CLID at all. Any ideas?
; use 0703241432 as outgoing msn
We bought 50 of these phones and deployed them at customer sites. But after 4
months of operation we have decided that they are completely unfit for our
use. The have many bugs. The worst one is the one where the phone stops
registering, others include: web page dies, numerous break in the
On Wed, 24 Dec 2003, Iain Stevenson wrote:
I have exactly this problem and posted a bug report to FWD about a week ago
- no response yet. It's bizarre that FWD recognises you to dial another
user but not to call outside their network. Sounds more like a FWD problem
than a * problem to
Michael,
A reply I received from Grandstream.
Depending on your firmware version. Firmware family 1.0.4.x is not
interchangeable with 1.0.3.x and therefore cannot downgrade back. What is
the current firmware version and what version do you want to roll back to?
Regards,
Richard Huang
Hello,
I've successfully installed Asterisk from last CVS and configured it
for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip
server.
All are work fine at G711 codecs, but then I disable all codecs except
g729 some calls failed (Not all calls. Some calls passed at g729
Have you bought G.729a from Digium which cost $10/channel?
At 02:04 25/12/03 +0300, you wrote:
Hello,
I've successfully installed Asterisk from last CVS and configured it
for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip
server.
All are work fine at G711 codecs, but then I
No, I did't bought any license from Digium. But as I say at my previous
post, only _some part_ of my g729 calls are failed !
I think if I need license for G729 at Asterisk then all of my calls must
to fails. Is it right ?
--
Antonio
-Original Message-
From: Peter Brown
I'm going to take a stab at this, so someone correct me if I'm wrong! If
you're calling one g729 device from another, the call is actually handed off
without any decoding done, therefore the licensing isn't needed. If * has
to connect the g729 call to another format, then the licensing comes in
Where can you get a cisco 7905 with a SIP license and power supply for $99?
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 24, 2003 5:30 AM
Subject: Re: [Asterisk-Users] Grandstream Quality
In my case I see only g729 codec request from CPE (see mgcp CRCX) and
only g729 from PGW2200 (see debug of sip messages) and I don't need and
transcoding from one codec format to another codec format.
Could you expain to me why asterisk starts transcoding process from
g729 to alaw ?
--
antonio
Hi,
Does asterisk support any kind of voice encryption?
Matt
Interesting! Surely it would be another greate project.
Happy christmas!
- Original Message -
From: Bob Knight [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 24, 2003 11:30 AM
Subject: [Asterisk-Users] time to build an open phone?
Open software seems to work.
Why
Title: X100p problem
I am having a problem with the x100p cards. It doesn't matter whether the card is in the machine or not, all I get is a busy signal when calling. The Asterisk box doesn't give me any errors and doesn't show that any call is coming through. I removed the cards from the
Hi Sean-
I've had pretty good experience with Digium boards - all of mine have been
shipped quickly, and all have worked upon arrival. Don't have experience
with the X100P, only the quad T1/E1 boards.
You didn't provide much information about what you tried already. Have you
got
Jeremy,
Ok, that worked. Thanks for your help, really appreciate it.
Let me copy this to the list, someone will find it useful.
So, If you want to run * at bootup, and you have chan_h323,
(a) then you should modyfy init.asterisk script with the path variables
(shown below) and copy it to
We wish you a merry IAXmas
We wish you a merry IAXmas
We wish you a merry IAXmas
and a happy new year!
From all of us in PK, Merry Xmas astmasters!
May 2004 bring freedom from SIP/H323/MGCP/SCCP and all other junk
protocols and may you realize the true spirit of IAX!
- wasim
and a special
On Wednesday 24 December 2003 20:14, Michael Welter wrote:
Besides the ata186, which phone is next up the food chain?
We are testing the Sipura SPA2000 and so far so good.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Vonage is running the latest 2.16-2 firmware. No longer applicable.
cameron.
On Wed, 24 Dec 2003, Doug Shubert wrote:
this security hole has been around for some time
http://www.securiteam.com/securitynews/5PP0G0K75U.html
Lion Templin wrote:
In the process of investigating a Cisco ATA
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