Does anyone have Asterisk working with Iconnect here for incoming and/or
outgoing calls? If you would be so kind as to share with me the
configuration you have used, as I cannot seem to get my SIP service to
work although it does seem to be registered with the other end:
hm6*CLI sip show registry
: Sunday, August 10, 2003 3:07 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Iconnecthere
Hi,
On Sun, 2003-08-10 at 09:02, Andrew Joakimsen wrote:
Does anyone have Asterisk working with Iconnect here for incoming
and/or
outgoing calls?
have a look at:
http://www.loligo.com/asterisk/example
Instead of using a PCI card is it possible to use an outside
SIP service for CO lines?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armand A.
Verstappen
Sent: Sunday, August 10, 2003 5:29 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Iconnecthere
Hi Andrew,
On Sun, 2003-08-10 at 19:39, Andrew Joakimsen wrote:
-Original
The past week or so I have been experimenting with Asterisk
and overall find it to be a nice software suite, although I have encountered
some problems, and have found almost no documentation (For example in sip.conf
I needed the commands fromuser= and fromdomain= and only figured out this
$75 for the single ethernet port version and $85 for the dual ethernet
port version.
You can get two for $129 at www.sipphone.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, August 17, 2003 8:23 AM
To: [EMAIL
The Snom VoIP phones support PoE and Nortel makes switches:
http://www.nortelnetworks.com/products/02/bstk/switches/baystack_460/
I am not certain that they are compatible, as I have not used the Snom
phones and have only used the Nortel switches with PoE adapters at the
other end to power
What did you change the DTMF mode to? Where can I find documentation
with all the possible options in the config files?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernest W.
Lessenger
Sent: Sunday, August 17, 2003 4:06 PM
To: [EMAIL PROTECTED]
I have removed all the dmtfmode= statements from my sip.conf to begin
with. Earlier today I downloaded and compiled the latest CVS and from my
testing right now it seems to work a lot better.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernest W.
I have a DID with VoicePulse Connect, but the sound quality is horrible, it is
often choppy and the caller's voice cuts out for 2-3 seconds at least once a
minute, I have contacted VoicePulse many times, and they do not do anything
about it! Does anyone have any similar problems? It isnt my
http://www.marko.net/asterisk/archives/0207/0097.html
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk - linux - JVB
Sent: Tuesday, August 19, 2003
6:12 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
MusicOnHold
Does anybody know
Maybe he figured something out
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
Sent: Tuesday, August 19, 2003 3:28 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Vonage locked ATA-186 question
Why ?
Martin
On Tue, 19 Aug 2003,
Is there a way to determine what codec the remote server
wants to use in a SIP session for an incoming call by looking at something,
possiby sip debug?
I already tried that, it says unknown.
I suspect it is requiring the G723 codec.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut .
Sent: Wednesday, August 20, 2003 3:28 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP using which
Actually I got it working right before I gave up (I had the wrong line
in my config commented out)
But now I get these messages when I try to playback a recording:
NOTICE[16401]: File channel.c, Line 1406 (ast_set_write_format): Unable
to find a path from GSM to G723
WARNING[16401]: File
that Asterisk transcodes (It also has to transcode to for
PSTN calls and voicemail and playing any sound files). Asterisk can't
transcode to or from G723. Nope. Doesn't work. May very well never
work. Use a different codec.
On Wed, 2003-08-20 at 03:47, Andrew Joakimsen wrote:
Actually I got it working
Linuxdevices says $400
http://www.linuxdevices.com/articles/AT9406437906.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Saturday, August 23, 2003 1:33 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Intresting.. hrm
The
I am having similar issues, except that I get the phones extension when
it its called, I tried to set the caller id number, and asterisk
recognizes the callers number, as well as defines it, it just does not
end up on the phones display.
-- Executing SetCallerID(SIP/-08114498, 3057400221) in
Has anyone been successful in using the DTA310 as provided by Packet8
to work with asterisk? I have gotten it to register with Asterisk but whenever
I try to dial a call all I get is silence, when I dial an invalid extension I get
a fast busy signal. When looking at the SIP debug it seems
My sip.conf entry for the cisco looks like this:
[cisco]
type=friend
username=cisco
secret=1234
host=dynamic
defaultip=[The IP of the 7960]
mailbox=
context=sip
callerid=Ben 1
Try to remove the defaultip= string. Do you get
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ben Wern
Sent: Saturday, August 30, 2003 3:02 AM
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960
Andrew,
Thanks for your help!
No
: Saturday, August 30, 2003 12:30 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Packet8 DTA310
Post the sip debug .. maybe someone will help you.
Martin
On Sat, 30 Aug 2003, Andrew Joakimsen wrote:
Has anyone been successful in using the DTA310 as provided by
Packet8 to
work
I am aware of a least a few people (including me) who were using the
Packet8 service along with Asterisk for outgoing calls. Last night Packet8 did
a software upgrade and both last night and this morning I have been unable to
make any outgoing calls. Has anyone else noticed this behavior
this debug desn't show the bad call setup. And furthermore all
commands are accepted by the asterisk/UA.
Martin
On Mon, 1 Sep 2003, Andrew Joakimsen wrote:
There might be some other stuff mixed in there as well,
64.36.104.205 is
asterisk and 64.36.104.206 is the DTA
11 headers, 2
No. You can use packet8 if you slightly
modify the asterisk source code (outgoing calls only) or you can use the
service provided by nufone.net
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Friday, September 05, 2003
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Friday, September 05, 2003 11:54 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] VONAGE or IP Dialtone
No. You can use packet8 if you slightly modify the
Prefix your search with site:lists.digium.com
So if you wanted to search the list archives for SIP you would enter
site:lists.digium.com sip
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Saturday, September 06,
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Saturday, September 06, 2003 8:57 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] MP3 streams for MOH: idea
[thread change, different topic]
is there a clean
http://store.yahoo.com/grandstream-networks-inc/products.html
They finally removed the password from their shopping cart!
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Thompson
Sent: Sunday, September 07, 2003 10:36 PM
To:
Does anyone have a source where these can be purchased?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Serge Mankovski
Sent: Monday, September 08, 2003 10:02 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] DLink DG-104S
Hi
in mind tho these things are about $400 in Australia.
J
On Mon, 8 Sep 2003 22:12:52 -0400
Andrew Joakimsen [EMAIL PROTECTED] wrote:
*This message was transferred with a trial version of
CommuniGate(tm) Pro*
Does anyone have a source where these can be purchased?
-Original Message
You should only need licenses for the users connected via IP **AND**
using G729.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chee Foong
Sent: Monday, September 08, 2003 11:48 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
Start off by reading the main page: http://www.asteriskpbx.com
Yes, for a cheap solution Digium sells 1-4 port FXS cards to which you
connect regular phones to. I you wanted a larger setup you could go
for a channel bank.
-Original Message-
From: [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Thursday, September 11, 2003 4:33 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Legal Interception - tapping
At 3:06 PM -0500 9/11/03, Steven Critchfield
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Thursday, September 11, 2003 10:20 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Legal Interception - tapping
pamAssassin 2.55 (1.174.2.19-2003-05-19-exp)
It works fine for me, I created a 2nd music on hold, tossed a bunch of
mp3 files into a directory and I can listen to music on the
speakerphone:
;radio @
exten = ,1,Answer
exten = ,2,MusicOnHold(default)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Sunday, September 14, 2003 2:18 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Does * machine need a sound board for
MOH?
(add to FAQ)
(snip)
Music on
Regards,
Andrew Joakimsen
Envision Studio
http://envisionstudio.net
888-210-8063
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Sunday, September 14, 2003 3:39 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk
Would any code other than dialing the extension to program the phone be
required? I have been in touch with Aastra's support and they seem to be
pretty helpful. Can the phones be reset and then loaded with the code
from asterisk?
Lastly, will these work with a SIP gateway or must they be directly
Try
host=sipauth.deltathree.com
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Wednesday, September 17, 2003 6:46 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Iconnecthere Problem
I can't seem
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael A.
Miller
Sent: Monday, September 22, 2003 10:40 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Recommended OS
Is there a recommended OS that Asterisk should be used with? I have been
trying to
And we all certainly know that Windows is so secure. I am by no means a
Linux or Windows fanatic, they each have their strong spots.
And I find this thread a little off topic, totally not related to
Asterisk or VoIP/phone systems.
-Original Message-
From: [EMAIL PROTECTED]
Try
Nat = yes
Or
Nat = no
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mike Diehl (Encrypted email
prefer
red)
Sent: Monday, September 22, 2003 11:38 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Can't get simple
Show us your sip.conf file, you can (should) block out the passwords.
Is there a [sipauth.deltathree.com] section in it?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of listas iPfone
Sent: Tuesday, September 23, 2003 3:22 PM
To:
This is what I have in my mgcp.conf
[dlink]
threewaycalling=yes
transfer=yes
callwaiting=yes
callwaitingcallerid=yes
host=dynamic
context=international
nat=yes
;dtmf=inband
disallow=all
allow=g711
allow=ulaw
callerid = Andrew Joakimsen 321
line = aaln/1
callerid = Andrew Joakimsen 322
line = aaln
Looks great. One suggestion would be to add a total at the end with
total/billable durations and total number of calls.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jamie Carl
Sent: Thursday, September 25, 2003 10:20 PM
To: [EMAIL
] CDR Web Search Frontend
Good suggestion! Duely noted. Check after the weekend
and it'll be there.
(i would do it tonite, but it's friday nite and there is
alcohol to be consumed)
J
On Thu, 25 Sep 2003 22:36:31 -0400
Andrew Joakimsen [EMAIL PROTECTED] wrote:
*This message
be unbearable.
Matt Hardeman
PaperSoft
- Original Message -
From: Andrew
Joakimsen
To: [EMAIL PROTECTED]
Sent: Thursday,
September 25, 2003 9:37 PM
Subject: [Asterisk-Users]
VoIP Support for Symbian OS Devices
Does anyone have any insignt
No, because asterisk cannot deal with the G723 codec, it can only act as
a middle man of sorts between devices that support it.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Bill Leckey
Sent: Sunday, September 28, 2003 7:03 PM
To:
]
Subject: Re: [Asterisk-Users] Outgoing call spool
Andrew Joakimsen wrote:
No, because asterisk cannot deal with the G723 codec, it can only
act as
a middle man of sorts between devices that support it.
Ok, that makes sense. Could I get the ringing somehow if I changed to
(say) the G711
DTA310 does not count because I cannot get it to function properly (as
well as another member on this list)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of TC
Sent: Sunday, September 28, 2003 11:06 PM
To: [EMAIL PROTECTED]
Subject:
Please post your extensions.conf and sip.conf sections relevant to
ich/deltathree.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of listas iPfone
Sent: Tuesday, September 30, 2003 3:33 PM
To: [EMAIL PROTECTED]
Subject:
I was told ADSI would not work on a dlink gateway, after setting
adsi=yea in mgcp.conf I now get:
Executing ADSIProg(MGCP/aaln/[EMAIL PROTECTED], ) in new stack
-- ADSI Available on CPE. Attempting Upload.
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'D'
I don't have an adsi phone to
=sipauth.deltathree.com
dtmfmode=inband
context=from-sip
miklos
- Original Message -
From: Andrew Joakimsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 30, 2003 5:27 PM
Subject: RE: [Asterisk-Users] I have a strange problem with ICH calls
Please
How did you get it to work? I cannot figure out how to get mysql cdrs
working, all I get is:
ERROR[16401]: File cdr_mysql.c, Line 130 (mysql_log): Failed to insert
into database.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Leif
What is the
Exten = .Dial(
Line from your extensions.conf?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Kevin
Sent: Sunday, October 05, 2003
7:23 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No
Ringback on Iconnect
When
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No
Ringback on Iconnect
I have tried both of these:
exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED]
exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED]||r
-Original Message-
From: Andrew Joakimsen
[mailto:[EMAIL PROTECTED]
Sent
PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No
Ringback on Iconnect
Changed my conf file to:
exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED],90,r
still no ringback
-Original Message-
From: Andrew Joakimsen
[mailto:[EMAIL PROTECTED]
Sent: Sunday, October
05, 2003 9
Fax with G711 works fine. Modem will be slow, but if you really need to
use it slown them down to 28.8 or 33.6
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eduardo Goncalves
Sent: Monday, October 06, 2003 2:56 PM
To: [EMAIL
How are you transfering to 700? You dial # while in a call and then it
says transfer and you then dial 700, or are you using a different
method?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Juan J. Sierralta P.
Sent: Tuesday,
, 2003 6:46 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Call park on SIP phones
On Tue, 2003-10-07 at 18:23, Andrew Joakimsen wrote:
How are you transfering to 700? You dial # while in a call and then
it
says transfer and you then dial 700, or are you using a different
method
phones
Yes but you can't do native sip tranfers to parking. Thats what I
want.
And thats what I was talking about. You can't say use a Cisco 7960
and
hit transfer then dial 700 then transfer. WONT WORK.
bkw
On Tue, 7 Oct 2003, Andrew Joakimsen wrote:
You need to enable transfer
http://www.megaglobal.net:8080/docs/asterisk/html/problems.html#MPG321
app_mp3.c requires mpg123 to stream mp3's over Asterisk channels.
If you're running RedHat Linux 7.X you may not have noticed that RedHat
sneakily replaced the 'official' mpg123 with their clone 'mpg321'.
This cloned version
What is the proper method to install/configure an X100P FXO
card?
I setup two machines to talk to each other with IAX and it does not seem
to work. When a call comes into one machine and transfers it to the
other, the machine that is transferring to the other one shows:
-- Accepting AUTHENTICATED call from 65.127.126.42, requested format
= 4, actual format
] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jeremy McNamara
Sent: Friday, October 10, 2003 8:25 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX Not working between machines
Andrew Joakimsen wrote:
I setup two machines to talk to each other with IAX and it does not
seem
to work
] On Behalf Of Jared Davies
Sent: Friday, October 10, 2003 11:09 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX Not working between machines
try adding:
host=dynamic
to the definition of the user in iax.conf
Andrew Joakimsen wrote:
I double checked the contexts
Includes are recursive
Make a context with just all the internal extensions, and then make
contexts for all the outbound calls and another group of contexts just
as you are doing (admin, sales, etc)
Then
[admin]
include = international
include = extensions
[sales]
include = longdistance
What devices do you plan to use? PSTN line in USA and IP phones in
Nepal? Would this be for one user or a large office?
Regards,
Andrew Joakimsen
Envision Studio
http://envisionstudio.net
888-210-8063
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Are you using NAT? Is nat=yes in your
sip.conf? canreinvite=no, reinvite=no ?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Chris Hariga
Sent: Sunday, October
12, 2003 10:42 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No sound
with
In what language is it written in? It
would be interesting to at least look at it and maybe convert it to use MySQL
instead
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Bartosz Jozwiak
Sent: Monday, October 13, 2003
3:49 PM
To: ASTERISK
Is there an underlying reason you want to do this? Because if a call is
already parked on 701 and you transfer another call to 701 to park it,
both callers would be connected.
I am sure there is a better way to implement what you want.
-Original Message-
From: [EMAIL PROTECTED]
Has anyone gotten 3 way calling to work? There seems to be
no way to swap to the other call and sometimes the unit will generate the call
waiting tone ever second. It also seems that if you try to flash the call and
then hang up you have to pick up the phone, flash back to the first call
No. I also run machines with pure VoIP and there is not a single problem
with music on hold.
I don't think an X100P card will help. Anything you gain from the
ztdummy driver will be the same as what you can gain from an X100P, FWIW
the card is just a $10 winmodem.
-Original Message-
Has anyone tested using SIP endpoints (Possibly the ATA-186)
with a connection that has at least 200ms, if not more, of latency? We are
trying to get some stuff setup in Australia and wanted
to know if this would be feasable, are there any added delays? Echos?
Is that why there is an X100P and an X101P? What design is the X101P
based on?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jon Pounder
Sent: Tuesday, October 14, 2003 7:07 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Is it possible to purchase one of these? Where?
Which model? Are you using them directly with Asterisk? Analog phones
should only be 1 line, IMO all the call processing should be handled by
Asterisk.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andy Hester
Sent: Friday, October
Of Andrew
Joakimsen
Sent: Friday, October 17, 2003 4:12 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Paging/Intercom (was: OT - SIP
Auto-Answer
for Cisco 7940/7960!!)
Which model? Are you using them directly with Asterisk? Analog
phones
should only be 1 line, IMO all the call
Why is mine different?
localhost*CLI show translation
Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)
G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX
ILBC
G723 - 45 41 41 41
733 MHz,
128MB.
I assume this is a way to know which codecs was loaded, because if I
unload
G.729 codec disappears from 'show translation' printout. But, who
knows...
Regards,
Gus
- Original Message -
From: Andrew Joakimsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent
I have a Nortel phone on my desk right now. IF the handset is picked up
and you press the speaker button, it does not hang up but switches back
to the handset instead.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Robert Hajime
I have modems that are IDENTICAL to the X101P card, same modem/part
numbers and FCC ID, yet they do not work.
Anyone have any clues as how to correct this issue? When doing an
lspci both cards show up as TigerJet Networks 320 128K or something
along those lines
-Original Message-
GSM is fairly low bandwidth and sounds pretty good. G711 (ulaw/alaw) is
by far the best, you can send faxes with it (and if I am not mistaken
Ilbic)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matthew Simpson
Sent: Thursday,
Is it possible to generate indications based on the context? And what
abou SIP devices, they generate their own tones
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Adam Hart
Sent: Thursday, October 23, 2003 2:20 AM
To: [EMAIL
It's already been done. The X101P is a $10 winmodem, tested by me as of
last night.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Thursday, October 23, 2003 4:12 PM
To: [EMAIL PROTECTED]
Subject: Re:
Does anyone know how to read? The X100P **IS** a winmodem:
http://www.kobian.com/products.php?productid=180
Buy one here for $15.50:
http://www.accupc.com/itemDetail.jsp?pid=fmint56vs/w
I am using one of these right now along with a real X100P without any
issues. They are IDENTICAL, FCC ID's and
I don't see what the problem is, Asterisk will see them as two separate
extensions
Exten = _9011x#,1,StripLSD(1) (or _9011.# if there is not
a fixed number of digits, change the other xxx to be .)
Exten = _9011x,2,Dial(IAX2/${exten:[EMAIL PROTECTED] (or exten:4 if you
do
In the general section of IAX.conf add
disallow=g729
If you are using G729 for placing calls, place
allow=g729
In the appropriate context.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak
Sent: Friday, October 24,
Look here for more info:
http://www.dslreports.com/forum/remark,8262032~root=voip~mode=flat
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Tuesday, October 28, 2003 12:13 AM
To: [EMAIL PROTECTED]
Subject:
If you already get dialtone, you just need
to setup the extensions
exten =
321,1,Dial(MGCP/aaln/[EMAIL PROTECTED],90,T)
exten =
322,1,Dial(MGCP/aaln/[EMAIL PROTECTED],90,T)
exten =
323,1,Dial(MGCP/aaln/[EMAIL PROTECTED],90,T)
exten =
324,1,Dial(MGCP/aaln/[EMAIL PROTECTED],90)
I have used G723.1 (although unlicensed) with Asterisk. The info is even
in the Makefile, just drop in a few files in your source directoy,
uncomment something in the Makefile and instant G723.1 support...
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
What interface is the phone connected to?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of PBX
Sent: Monday, November 03, 2003 8:01 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] ADSI - PowerTouch 350
I was wondering if anyone
Did you setup your zaptel.conf?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Sathya Weerasooriya
Sent: Wednesday, November 05, 2003 2:14 AM
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: [Asterisk-Users] X100P - module does not
On 7/19/07, mail-lists [EMAIL PROTECTED] wrote:
Hello,
We're in the process of moving to a PRI circuit for our asterisk switch.
Can anyone point me in the right direction as far as PRI Cards are
concerned?
Thanks!
www.sangoma.com those are the best.
On 7/19/07, satish patel [EMAIL PROTECTED] wrote:
I have snom SI 120 sip phone and there is transfer button but id there any
configuration in asterisk part for call transfer feature ???
Nothing else is required. Since the phone has a transfer button there
is no need to use features.conf. What
You should be running the latest Zaptel LibPRI both of which
recently have been updated. We run a similar configuration and have
not seen this problem with the upgrade. I do get a flood of:
Jul 19 19:36:18 WARNING[7277]: pbx.c:815 pbx_find_extension: Maximum
PBX stack exceeded
Jul 19 19:36:18
On 7/19/07, randulo [EMAIL PROTECTED] wrote:
I just noticed that I asked about this same problem in March and got
a workaround (edit makefile) from Tzafrir. Could someone explain why
this codec_zap line is in Makefile has to be manually commented out?
THere must be a reason why this happens
I have already tried to contact to persons from Digium and I did not
receive a response.
I was wondering if there is any plan to support fully faxing in
Asterisk, I.E.: A T38 Gateway of sorts.
___
--Bandwidth and Colocation Provided by
On 7/12/07, Russ McBride [EMAIL PROTECTED] wrote:
Newbie question(s):
From what I can determine it sounds like the SMS messaging isn't as
robust as it could be (?). I'm wondering if there's active work on
that right now or if it's more of an issue about PSTN carrier that
one would be
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