The most economical way is just multiple asterisk boxes,
even though it may use more space.
On Thu, 3 Jun 2004 20:13:03 -0600
brian k. west [EMAIL PROTECTED] wrote:
Go spec some hardware dsp chips and boards that can do
100 channels... I
think you will fall out of your chair.
bkw
-
I want to setup a phone system that can allow the caller to type in a faxback number,
and automatically be sent a fax file. Is this possible with Asterisk? Or using
Asterisk a Linux fax program? I've seen stuff about incoming faxes, but nothing
about outgoing. This is over a Zap channel,
Am 03.06.2004 um 23:51 schrieb Senad Jordanovic:
Many thanks for the info.
You're welcome!
I need this info to enter it for each country for our upcoming asterisk
web interface solution. If you are interested in asterisk web
interface,
I can notify you when it is ready!
Yes please, this would be
I have done a quick search and there are some nice looking
dsp-pci cards out there. (Dunno abt prices). It may take
some coding to get them working with Asterisk , and one
would not require a super-power quad xeon processor if it
had a huge dsp card.
May be an interesting way to scale asterisk
On Fri, 4 Jun 2004, Doug R wrote:
I want to setup a phone system that can allow the caller to type in a
faxback number, and automatically be sent a fax file. Is this possible
with Asterisk? Or using Asterisk a Linux fax program? I've seen
stuff about incoming faxes, but nothing about
I will try spandsp then.
On another note, what SIP server works best with Asterisk? I want to use Asterisk for
a PSTN gateway, and currently I am using as a SIP server since it seems like
everything else has so many firewall issues.
If I have all my SIP clients using the same codec, does it
It cannot be done in * so in your pbx set the first number to divert to the
second on busy and the other way and not ring both numbers. that will
resolve your issue
At 09:14 01/06/2004 -0300, you wrote:
Hi all,
Anyone know how put my X101P cards to answer at different ring times ?
Like
Hi,
I spent some hours working my way through the WIKI and a number of other
documentations, but after all, three questions are still left:
1. I'm using a Fritz!Card with the i4l driver - no problem at all, my
Grandstream BT 100 rings when I diall a regular phone number.
Is there any need for
Hi all,
for some strange reason, our still-under-test Asterisk deployment wants
to contact the outside world and that raised some eyebrows here...
Just a sample of our firewall log:
--
...a=DROPIN=eth0 OUT=eth2 SRC=192.168.36.199 DST=195.77.113.194 LEN=476
TOS=0x10 PREC=0x00 TTL=62 ID=39572 DF
Clive
You need to make sure there is a link from /usr/src/linux to your RH kernel source
directory. Then it will work. This is not there by default on RH9. Been there, done
that!
I have 2 zaphfc cards running well in my RH9 box. Stick with it - it is worth the
pain.
Rgds
Tim
On Fri, 4 Jun 2004, Stefan-Michael. [iso-8859-15] Günther (in-put GbR) wrote:
1. I'm using a Fritz!Card with the i4l driver - no problem at all, my
Grandstream BT 100 rings when I diall a regular phone number.
Is there any need for me to configure the zapata.conf for the ISDN card to get
Hi list,
I have an asterisk server with a Zap 4 port FXO card connected to the PSTN,
all of the clients are SIP softphones (I have tried LIPZ4 and kphone). I
have successfully configured asterisk to route incoming calls from the PSTN
to an extention that is a SIP softphone and the user can answer
hi all,
at time i am working on the app_prepaid.so module (see:
http://www.voip-info.org/tiki-index.php?page=Modified-Prepaid-Application ). In the
source file a have many ast_log(LOG_DEBUG statements for debugging - but even if i
start asterisk with asterisk -dvvvgc i wont get the
Hi !
it was designed for our receptionist
Please post a picture of that recepcionist .. maybe she can be the
asterisk girl of 2004!
Claudio
- Original Message -
From: Kyle Hagan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 03, 2004 6:21 PM
Subject: Re:
Hi!! I have a problem with DynExtenDB.
This is the message:
ERROR[245776]: app_dynextendb.c:76 dynamic_extension: No DNID in channel
found - not possible to query extension. Skipping.
Can you help me? Thanks...
Fabio Donaggio
___
Asterisk-Users
Thanks for your replies. The hangup is still failing with the latest CVS
head. It seems to be a firmware issue. I am running WJ.00.0a / B.00.13 /
Apr 12 2004 - Is there any newer release of the firmware floating around?
cheers
Dominique
PS:
Another interesting effect(IMHO bug): I cannot access
Hi Martin,
This looks like a SIP reply.
I suspect that a misconfigured SIP phone or proxy is inserting
a Via: header that contains the 195.77 address, or a name that
resolves to it. Capture the packet text with your firewall,
or by running Ethereal on your * machine, or with * itself,
and the
Has anyone ever thought configuring asterisk on a pair of pc's to act as
remote broadcast terminals for the broadcast radio industry? Seems like
a stripped down asterisk on a laptop with a PCMCIA ISDN modem connecting
to another asterisk instance on a PC at a radio station would work
nicely.
Hi,
I would like to ask you for advice how to solve the following case:
I have a client (who happened to be my friend) and I have convinced
him that the IP PBX solution is much better than the conventional
telephone centrals (PBX). At the beginning he wanted to buy PBX
Panasonic, but at
Why? Just use shoutcast/icecast for that.
Bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nik Martin
Sent: Friday, June 04, 2004 7:49 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] (possibly) new use for asterisk
Has
If I turn allow=ulaw on only, asterisk tries to use it
a=rtpmap:0 PCMU/8000
but the ATA says it doesn't have it:
Answering/Requesting with root capability 4
Answering with non-codec capability 0x1(G723)
If I turn allow=alaw on only or with allow=ulaw, asterisk sends it, the ATA
says it has it
If there is already an existing phone system in place, you could easily migrate to an
asterisk based solution if your internal phones are analog. The big question for you
is not number of phone lines, but peak utilization. Here's what I have.
141 Analog Phone Lines
15 SIP IP Phones (Mix Cisco
think I figured out the binary bit thing, so I am posting to list to
hopefully help someone else out
bits 15-8 are all 0 and are reserved
bit 7:value 0:numeric 8 reserved
bit 6:value 0:numeric 4 reserved
bit 5:value 0:numeric 2 dtmfmethod
bit 4:value
Since connecting a PRI to a Digium T100P, I have been seeing the
following messages in syslog every few minutes:
Jun 4 06:51:54 pbx asterisk[13435]: WARNING[1209214400]: chan_zap.c:6176 in
zt_pri_error: PRI: Read on 56 failed: Unknown error 500
Jun 4 06:51:54 pbx asterisk[13435]:
There are commercial providers online that build ready-to-go asterisk
servers and hardware:
http://voipstore.atacomm.com/shops/Browse.aspx/27934028032-27934066432.htm
They should be able to build a turnkey solution for you. There are also
consultants on this board that will probably assist you
Hi all,
I am a new user of asterisk. I was actually searching the net for Call
Managers for Cisco 12 SP+ phones and found a statement on
http://www.wlug.org.nz/BlairHarrison that asterisk works with these old
phones too. Being the users of Asterisksomebody please verify that it
actually does
Does shoutcast run across isdn?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of brian
Sent: Friday, June 04, 2004 8:53 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] (possibly) new use for asterisk
Why? Just use shoutcast/icecast for
On Friday 04 June 2004 05:00, Wolfgang Pichler wrote:
hi all,
at time i am working on the app_prepaid.so module (see:
http://www.voip-info.org/tiki-index.php?page=Modified-Prepaid-Appli
cation ). In the source file a have many ast_log(LOG_DEBUG
statements for debugging - but even if i start
On Wednesday 28 April 2004 17:10, Martin Christian Koch wrote:
Rxfax answers, makes handshake, and crashes once the page starts to send.
It receives a .tif file of 8 bytes.
I am *just* starting to play with this. I found out I had libtiff-3.5.7
hanging around on the system, but I had
Darren
Thanks for the info , i have made the changes and my telco says he is seeing
BUFFER CONTENSIONS on one of the two circuit's ( cct ) . cct 1 is performing
correctly cct 2 is not.
My telco is providing me with plain Q.931 not Q.931e ( euro ) he has
mentioned that my sync settings may not be
I'm pretty sure the Error 500 can be ignored. If memory serves, I think it
is related to a transmitter underrun error on the T100P framer chip. This
would occur if the processor doesn't quite keep up with the transmit data
stream on the T1 - ie: is load related.
These occur on all of my
Dear Scott,
The idea is to be used new SIP phones instead legacy phones. With
this the network cables (UTP/FTP-5) will be used instead one cable
network for the Computers and other cable network for the Phones.
Best Regards,
Miroslav Nachev
If there is already an existing phone
Hi Simon-
Just to speed things along, have you see the earlier post regarding Marconi
by Darren Storer on this topic?
Here it is:
Are you sure that NTL have provided you with a true Q.931 EuroISDN PRI
circuit? If the circuit was supplied some time ago for use with existing
equipment it may not
On Fri, 2004-06-04 at 09:07, Bruce Komito wrote:
Since connecting a PRI to a Digium T100P, I have been seeing the
following messages in syslog every few minutes:
Jun 4 06:51:54 pbx asterisk[13435]: WARNING[1209214400]: chan_zap.c:6176 in
zt_pri_error: PRI: Read on 56 failed: Unknown error
I should state that most ISDN lines that radio stations use are not
connected to ISP's but are point to point ISDN connections for high quality
voice traffic. I don't know much of the technical aspects of ISDN, but do
know that they are usually set up for temporary use for sports broadcasting,
Dear Julian,
Quoting Julian Pawlowski [EMAIL PROTECTED]:
Hello folks,
i upgraded to the actual CVS head from yesterday (27.5.) but can not get
incoming SIP calls from my provider (sipgate). If someone calls my
number, my asterisk responds with the following error:
May 27 21:30:21
I am looking to have some DIDs in 602 and 520 (Arizona). Anyone know a
provider that can terminate and forward to *?
-jeff
Jeff Coleman
Resource Strategies, Inc.
The Intelligent Use of Technology
Tollfree: 877-718-7628 x401
Fax: 520-797-0394
I'm running asterisk CVS HEAD from 20040601 with spandsp 0.0.1k and libtiff
3.6.0 (no other copies are installed).
I've put the audio files up at
http://www.mixdown.ca/~andrew/dump/akohlsmith-faxsegfault.tgz -- the machine
I am faxing from is a Canon IR3300 printer/copier/fax, but I get
On Friday 04 June 2004 12:16, Andrew Kohlsmith wrote:
Now one thing I see at the bottom of the output of rxfax below is that
compression is there -- I did *not* install the lzw stuff on the libtiff
webpage. I'm going to try that next.
And after enabling lzw in libtiff it doesn't crash
Hello all.
I am a little (allot) lost on my next hurdle in getting an asterisk system built.
I would like to get my asterisk servers configured exclusively from database. I have read through the wiki on this subject but once again I find that there is a certain level of knowledge that is
On Friday 04 June 2004 12:28, Andrew Kohlsmith wrote:
And after enabling lzw in libtiff it doesn't crash anymore, although it
doesn't work yet, either. :-)
I just had my first successful fax reception... The sending machine thinks
I got it, but I sure don't think so based on the resultant
Hello,
We use SER as SIP proxy/registrar and Asterisk as the media Server for
handling IVRs and Voicemails. The system works fine and the benefit is it is
highly scalable.
Regards, Girish
From: usedcanon [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk SER (www.IPTel.org)
Date: Thu, 3
I am having a serious problem with my Asterisk system. Every few days, my
PRI line resets and drops all calls. I get these errors in the messages
log:
Jun 3 02:41:11 WARNING[11276]: PRI: Read on 39 failed: Unknown error 500
Jun 3 02:41:11 NOTICE[11276]: PRI got event: 6 on span 1
Jun 3
Where do I identify the listen port on my asterisk box?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Yo have to stop and restart asterisk to get the new seting to work, not
reload
Erick
- Original Message -
From: Matthew Simpson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 04, 2004 8:52 AM
Subject: Re: [Asterisk-Users] miserable time with Cisco ATA186
If I turn
Jeremy,
Thank you. That is what I mean. And I'm sitting here, looking at the
debug window and scratching my head as to HOW he might be using
extension.??
My phone is defined as ACME1000 in sip.conf
In extensions.conf I have a:
Exten = 1000,1,Dial(SIP/ACME1000) (well, basically)
So when I'm at
Hi Jay,
Thanks for the reply.
Yes, we do have a context for incoming calls -- it's used for not only
BroadVoice (which isn't working) and VoiceGlo and iConnectHere
(which are working.) And, yes, we do have a pattern match on our
number in our [general] context in extensions.conf. As of this
also you have to set the sip.conf on the ATA sittings to disallow=all,
allow=g729 OR set the txcodec and rxcodec to 3 on the ATA
- Original Message -
From: Matthew Simpson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 04, 2004 8:52 AM
Subject: Re: [Asterisk-Users] miserable
Patrick Lidstone (Personal e-mail) wrote:
Please excuse me if this is a niaive question...
I have Cisco 7940 (but same applies to Snom's too), and it would be
convenient to have multiple extensions on the same phone registered
against the same asterisk instance. (E.g. one extension which is
netstat -an --inet | grep LISTEN
Ivan
-- Mensaje original --
From: Christopher Wall [EMAIL PROTECTED]
To:[EMAIL PROTECTED]
Subject: [Asterisk-Users] listen port
Reply-To: [EMAIL PROTECTED]
Date: Fri, 04 Jun 2004 10:54:26 -0600
Where do I identify the listen port on my asterisk box?
On Friday 04 June 2004 13:09, Gary Franczyk wrote:
I am having a serious problem with my Asterisk system. Every few days, my
PRI line resets and drops all calls. I get these errors in the messages
log:
Jun 3 02:41:11 WARNING[11276]: PRI: Read on 39 failed: Unknown error 500
This has been
If you dont know to build the database from command line try to get an
MySQL client like PHPMyAdmin or MySQL module of web min can do the same
job. if you have download add-on`s from the digium CVS server then the job
is done.
Hello all.
I am a little (allot) lost on my next hurdle in
On Fri, 2004-06-04 at 12:09, Gary Franczyk wrote:
I am having a serious problem with my Asterisk system. Every few days, my
PRI line resets and drops all calls. I get these errors in the messages
log:
Jun 3 02:41:11 WARNING[11276]: PRI: Read on 39 failed: Unknown error 500
Jun 3
Senad Jordanovic wrote:
In United Kingdom, we have time based dialling pricing from most of
Telco's
based on time the call is placed! It is called PEAK (08.00- 18.00
Mon-Fri), OFF PEAK(18.00-08.00 Mon-Fri) and WEEKEND (all other times!
Could someone from any of other countries let me know if time
Tony Hoyle wrote:
Eric Wieling wrote:
Why are you even looking at VoIP? Analog ports and phones are pretty
cheap. They are not pretty, but they are cheap and all the smarts are
in the PBX.
Free calls to the US, basically, since the leased line is dirt cheap to
run. ie. the purpose of the
Oh, forgot to officially ask...
Could anyone help me out getting asterisk talking to MYSQL? [EMAIL PROTECTED] 6/4/2004 11:36:24 AM
Hello all.
I am a little (allot) lost on my next hurdle in getting an asterisk system built.
I would like to get my asterisk servers configured exclusively
Has anyone have any insight on configuring a Cisco VG-200 (IOS gateway with
4 FXOs) with Asterisk? It seems that the asterisk mgcp channel uses 1.0 and
the VG200 use 0.1. I see protocol errors on both debugs. I upgraded the
ISO, but still no luck?
Otherwise for anyone have samples of configs
Why in the world would he want to do that You are assuming he has a
G729 license.
On Fri, 2004-06-04 at 12:33, Erick Weber V. wrote:
also you have to set the sip.conf on the ATA sittings to disallow=all,
allow=g729 OR set the txcodec and rxcodec to 3 on the ATA
- Original Message
I have two contexts and there I have some sip clients and some iax clients,
in the sip clients a have extentions like, 20, 21, 22, 23, etc; in the iax
clients I have some extentions like 2000, 2001, 2002, 2003, etc.
My extention is 2003 when I make a call the manager program show me that the
On Thursday 03 June 2004 07:05 pm, Andy Powell wrote:
chan_btp
Hi Brian,
You might also like to take a look at chan_btp and the btp daemon
which allows the use of bluetooth devices to change routing. Since
any old linux box that can handle a bluetooth dongle can report
back to a server you can
(sticking my nose in here:) Yes, but it sounds like Gary, in addition, is
getting Red Alarms, which probably indicates a more serious problem... Red
Alarm is a loss of sync from the other end, as I recall. I have *not* found
that slow processor interrupt service or similar maladies cause
Here come the flames.watch out.
Terry, I would suggest picking up a book about MySQL at Borders. Not a
thick one, but a starter. There are many sites on the Net that can help
too..
I would suggest: http://dev.mysql.com/tech-resources/articles/
The first link should get you started. I
Thanks for the info, Ill give it a try.
Terry [EMAIL PROTECTED] 6/4/2004 2:40:16 PM
If you dont know to build the database from command line try to get anMySQL client like PHPMyAdmin or MySQL module of web min can do the samejob. if you have download add-on`s from the digium CVS server then
On Friday 04 June 2004 12:28, Andrew Kohlsmith wrote:
And after enabling lzw in libtiff it doesn't crash anymore, although it
doesn't work yet, either. :-)
I'm wondering if our fax machine is sending iffy compression... There are a
few other fax machines it refuses to talk to and while it is
Hi-
Have you checked out the Wiki on this subject:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20mysql
This should help you get started with MySQL, which works great with
asterisk, I've found.
You have to install the MySQL package - use the free version 4 at
www.mysql.com,
Dear
all,
I am looking for
software sip phone and hardware sip phone for our network with great quality.
Need your suggestion. Thank you.
Best regards
IT Department
Director of Information
Technology
Albert Chong
562-695-8823Ext.2201
I searched the archives and nothing seemed to fit the problem. Most of the
posts I have found say what you said... this was just discussed, but I
cannot find any good information about the actual discussion. If you can
send a link to the thread you are talking about, I will check it out.
It is
I don't think the question you answered is the same as the one I asked. The
problem with mine is the dropping of all the lines/calls. It resets all
the lines.
I get those mystery notices on occasion also, but they don't drop all the
lines until I see the Detected Alarm messages for each line.
Dear
all,
Iwant
toconfigure QoSin my Cisco router and Cisco
Switch..Needsome information.
Need your
help.
Best regards
IT Department
Director of Information
Technology
Albert Chong
562-695-8823Ext.2201
I am trying out a new service from www.talkn.com. They use Sipura to
(Bterminate your service like most providers. They are looking at directly
(Bconnecting into asterisk in the future.
(B
(BRight now my configuration is Talkn$B"*(BSipura$B"*(BAsterisk/FXO Card.
(B
(BWhen someone calls
:-)
flamed? I hope not.
I have already started reading up on mysql and c and Perl and xml and java and r... So many things I need to get working so little knowledge of coding and so little time. All I can offer anyone right now is good will and future benefits from anything
same here, I 4 extensions from 2 different servers without any problems
(Cisco 7960)
Wojtek
- Original Message -
From: John Fraizer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 04, 2004 1:33 PM
Subject: Re: [Asterisk-Users] IP Phone with multiple accounts on same
instance of
Hey gang,
If anyone is in the Niagara area, or Western New York, the upcoming Bitnet
Niagara meeting/speaker might be of interest.
Presentation is on voip, and the wifi network going into Buffalo. Speaker
is Dave Witczak of Cisco
Details at www.bitnetniagara.com.
Jon Pounder
_/_/_/
Here is
what I use on a customer's router. He has a mix of different IP phones
which make it a little strange, but it seems to work. Be aware that
setting COS on an ethernet had severe bugs up until a service release a month or
so ago. I haven't tested the fix yet.
tim
class-map
FYI to all you Grandstream users out there. I just fetched and
installed the 1.0.5.0 firmware, and it appears they have removed the
option to either do or not do SIP registration. Now it appears that one
is going to register with the server specified in the SIP Server
field, without any
We are working on the revision right now. The problem you are having
with ext 20 and ext 2000, 20001, etc. should be fixed in the next
version. It was coded to instr which it saw ext 20 was in the 2000,2001
and lit that button up. We are changing that, it was just temp. We are
using JUST 3
On Fri, 2004-06-04 at 14:02, Gary Franczyk wrote:
I don't think the question you answered is the same as the one I asked. The
problem with mine is the dropping of all the lines/calls. It resets all
the lines.
I get those mystery notices on occasion also, but they don't drop all the
lines
Sorry about the HTML. This should be better.
I now have the mysql database created with the appropriate tables,
keys. Wasn't to hard :)
Im now trying to work out how to populate the database with my users
information.
Thank you for the info and the advice on HTML messages.
Respectfully
Terry
Title: Message
H,
Google
is your friend:
http://www.google.com/search?q=SIP+phones+asterisksourceid=mozilla-searchstart=0start=0ie=utf-8oe=utf-8
The
second result brings you to a page that's all about your
question.
It
also links to a HUGE resource list:
Hi there,
I tried to install appradius with asterisk. The appradius instalation finishes
succesfully, but the problem is that I can't find the app_radius.so and
cdr_radius.so files that the installation indicates to copy to the
/usr/lib/asterisk/modules
If you are getting a red alarm you have serious line or hardware issues.
You need to get a stand-alone analyzer on the T1/PRI to record.
Timothy R. McKee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
hi!
I'm having the same problem, I'm connecting through a Planet VIP-450
ITG, and when I send a DTMF code I get a:
WARNING: codec_ilbc.c:141 ilbctolin_framein: Huh? An ilbc frame that
isn't a multiple of 50 bytes long from RTP (4)?
I tried using different dtmf settings in sip.conf, but the
On Friday 04 June 2004 15:00, it.albertchong.p8.hq.us wrote:
I want to configure QoS in my Cisco router and Cisco Switch.. Need some
information.
I just posted a sample config to this list this week. I suggest searching the
mailing list archives for my posts.
Regards,
Andrew
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote on 05/28/2004
03:51:02 PM:
Dear users:
I have bought TDM04B card and it works in PCI 2.2 ver. slot.
How can I check if specific mother board support PCI 2.2 ver.
I do not have any documentation for that motherboard.
The
Fabio Donaggio wrote:
Hi!! I have a problem with DynExtenDB.
This is the message:
ERROR[245776]: app_dynextendb.c:76 dynamic_extension: No DNID in channel
found - not possible to query extension. Skipping.
Can you help me? Thanks...
Help yourself by not using dynextendb.
Jeremy McNamara
Hi,
i am using iax client and when i try one of my extension that play MusicOnHold()
it give me this error, who have an idea about this
- Executing MusicOnHold([EMAIL PROTECTED]/1, ) in new stack Jun 4 15:36:37
WARNING[1217602880]: chan_iax2.c:2838 iax2_send: timestamp is 0?
Jun 4 15:36:37
On Friday 04 June 2004 15:02, Gary Franczyk wrote:
I searched the archives and nothing seemed to fit the problem. Most of the
posts I have found say what you said... this was just discussed, but I
cannot find any good information about the actual discussion. If you can
send a link to the
Title: Message
flamed? I hope
not.
I have already started reading up on mysql and c and Perl and xml and
java and r... So many things I need to get working
so little knowledge of coding and so little time. All I can offer anyone
right now is good will
On Fri, 2004-06-04 at 15:02, Timothy R. McKee wrote:
Here is what I use on a customer's router. He has a mix of different
IP phones which make it a little strange, but it seems to work. Be
aware that setting COS on an ethernet had severe bugs up until a
service release a month or so ago. I
Thanks for the input all.
Its working now. Very cool stuff, just wish asterisk took more
advantage of the database. In time I guess.
Regards
Terry
[EMAIL PROTECTED] 6/4/2004 3:04:18 PM
Sorry about the HTML. This should be better.
I now have the mysql database created with the
Brian Capouch wrote:
FYI to all you Grandstream users out there. I just fetched and
installed the 1.0.5.0 firmware, and it appears they have removed the
option to either do or not do SIP registration. Now it appears that
one is going to register with the server specified in the SIP Server
When I emailed support with a pre-sales question about this and they
said they would be providing a generic BYOD where they provide the
authentication information and you configure your own client. He said
they would offer it this week, but that was on Wednesday and he could
have meant within
When I first got this going, I tested with a spare SPA-2000 as well as
Xlite. Once I got it working there, I transferred the settings to *.
Sorry I couldn't be of more help.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Swan
Sent: Friday, June
Assume you have the messages button on your Cisco phone set to dial
3009. Here's an sample dialplan entry that will make the DND and
ToVM and Messages button work as expected. This should work for
both -stable and -head.
exten = 3009,1,GoToIf($[X${RDNIS} != X]3009,4)
exten =
Title: Message
I hope
you will start carrying VOIP equipment too.
-Original Message-From:
it.albertchong.p8.hq.us [mailto:[EMAIL PROTECTED] Sent:
Friday, June 04, 2004 1:54 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users]
Recommendation for sip phone
Dear
all,
Tomas Prybil wrote:
Brian Capouch wrote:
FYI to all you Grandstream users out there. I just fetched and
installed the 1.0.5.0 firmware, and it appears they have removed the
option to either do or not do SIP registration. Now it appears that
one is going to register with the server specified
When reading the feature section of *.ororgt
mentions a/ululawwould that imply G711? Also,
it said that fax is incomplete. Has there been any
more development work on fax? Will * support t.38
anytime soon?
Kurt
__
Do you Yahoo!?
What type of cisco phones? i'm using 7960's and i know they don't have a
to voice mail button. That annoys me.
At 02:59 PM 6/4/2004, you wrote:
Assume you have the messages button on your Cisco phone set to dial
3009. Here's an sample dialplan entry that will make the DND and
ToVM and
Hi Matt,
On the ATA, set TxCodec=2 and RxCodec=2 (G.711u).
Also, set AudioMode=0x00160016 , which will force G.711 .
After saving, reload the /dev page to be sure that these
values are set as expected.
In Asterisk, allow=ulaw only.
If it still doesn't work, use the NPrintf field and
prserv,
What type of cisco phones? i'm using 7960's and i know they don't have a
to voice mail button. That annoys me.
How about the Messages button?
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