Hi everyone,
I've setup a test extension that should, if my understanding is correct,
play the music on hold mp3. But all I get when I call that extenion is
silence and the following message on the console:
-- Executing SetMusicOnHold(SIP/softphone-ba80, Default) in new stack
-- Executing
Title: RE : [Asterisk-Users] Voicemail + SIP Message header
Hi Deepak,
I had a similar setup, However I was able to configure the softswitch to send a prefix followed by the number dialled in the to field. This way I could route the call to the right mailbox and be able to play the right
Hi Mark,
I have implemented a procedure for automatically calls
from the client-side (IaxClient - E100P)
What I want to do is to detect the call status from
the client-side.
Meaning, if the line is busy/unavailable/fax log the
status and proceed to next call.
Is this possible with Manager API ?
I had the same problem. It turned out that X was hogging all the
resources available. Running without it, the problem cleared up.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Bridges
Sent: Monday, March 29, 2004 9:09 AM
To: [EMAIL PROTECTED]
Does anybody know any alternatives to the TxPort 1558 Auto Protection
switch? This box allows switching from one PRI to another in case one of
them goes down. Conversely, it allows switching a single PRI going to one
PBX (or * box) to another in case the PBX goes down.
Does AdTran make such a
Hello everyone,
Thanks for your feedback regarding testing of our worldwide termination
service. It was very informative and we are happy to say most of the
feedback was very positive considering the distances some of you were
calling in from.
We have set up a dedicated mailing list for
Hi all !
Everything works fine but
when I try to make connference,
I receive unable to open channel
I don't have this problem when
I run asterisk as root.
Probably something wiht file permission. Which ?
Thanks
Andrzej
___
Asterisk-Users mailing list
Hi,
I've search and though I've found a few references I have not
been able to find any concrete examples of * routing a call based upon
the caller ID. The scenario is that I want all calls originating from
number x to be routed to a particular extension, those from yy
to another
Well, I've made progress. As suggested I've just tried it without X running
and the errors have gone. No when I call the extension no errors, but
nothing is played Very odd..
Any ideas?
Matt
-Original Message-
From: jc [mailto:[EMAIL PROTECTED]
Sent: 29 March 2004 09:38
To:
On Mon, Mar 29, 2004 at 12:03:52PM +0200, radan wrote:
Hi all !
Everything works fine but
when I try to make connference,
I receive unable to open channel
I don't have this problem when
I run asterisk as root.
Probably something wiht file permission. Which ?
/dev/zap something, don't
Hello
My setup look's like this:
[* with oh323 and g.729 codec's] - [dlink dg-104SH, h323 with g729]
Dlink configured to send a call to * directly, so when I call 006 it comes
to * extension 006. On it I place echo-test. I hear voice greeting and after
few seconds asterisk hangup channel with
There are specific examples of call routing based on CallerID in the
handbook. You can read the handbook at this URL:
http://www.digium.com/handbook-draft.pdf.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John F.
Baird
Sent: Monday, March 29, 2004
John,
This is referenced as the anti ex-girlfriend feature...
example:
exten = s/12345678,1,congestion
exten = s/24681012,1,Dial(SIP/phone2)
exten = s,1,Dial(SIP/phone1,30)
also check page 31 of the handbook...
hth
Andy
*** REPLY SEPARATOR ***
On 29/03/2004 at 20:34
Good day
Does Asterisk work with the Voicetronix Openline4 cards?
Thanks
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Altus Snyman wrote:
Good day
Does Asterisk work with the Voicetronix Openline4 cards?
Yes, see: http://www.voicetronix.com.au/vpb4_v4pci.htm
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To
You may have the same problem I have which is that the MOH is
very very quiet you need to be in a noise free environemt to hear it!
I can't find an option to increase the volume ?
Chris
- Original Message -
From: Matt Bridges [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March
In article [EMAIL PROTECTED],
Matt Bridges [EMAIL PROTECTED] wrote:
Well, I've made progress. As suggested I've just tried it without X running
and the errors have gone. No when I call the extension no errors, but
nothing is played Very odd..
Any ideas?
Have you got a zaptel card
The thing is,Im not the sharpest tool in the shed, and I really need
help setting it up.I've installed Asterisk but thats how far I'm
getting,would you please Help me,Please
On Mon, 2004-03-29 at 14:33, michiel betel wrote:
Altus Snyman wrote:
Good day
Does Asterisk work with the Voicetronix
Hi,
I don't have a zaptel card installed at the moment. The initial problem was
that mpg321 wasn't working properly, I've got over that problem, but it
sounds really odd Slow and quiet. I'm assuming that' the problem you
are referring to?
Cheers
Matt
-Original Message-
From:
extensions.conf:
exten = s,1,Answer
exten = s,2,SetVar(FAXNAME=${TIMESTAMP}-${CALLERIDNUM})
exten = s,3,RxFax(/var/lib/asterisk/fax/new/${FAXNAME}.tif)
exten = s,4,Hangup
exten = i,1,Hangup
exten = h,1,Hangup
And here is the log file:
Page 1 of /var/lib/asterisk/fax/new/20040329-234801-0755965128
Just wanted to let you know that the attachment you sent was infected with a
Virus.
Norton AntiVirus removed the attachment: your_document.pif.
The attachment was infected with the [EMAIL PROTECTED] virus.
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday,
I want to setup a quick * box, and test out it's
functionality using a softphone, but the only thing I'm thinking is, even if I
get my box up and running properly, how will I call it!? I want to test things
like the voicemail system, and all that, befor eI start buying cards and adding
them
Hi All-
I've just finished setting up my first asterisk box with an X100P and
TDM40B, with basic asterisk config mostly from ManxPower's sample
configs. Thanks for posting those, Manx, and thanks for your help on
IRC.
I'm also in the midst of carefully studying John Todd's config files,
and
--On Monday, March 29, 2004 8:24 am -0500 Kevin [EMAIL PROTECTED] wrote:
Hi All-
As I'm doing this, I'm considering installing an asterisk box at my
office (about 6-10 different phone stations) and would like to get
opinions on the best quality and/or most well-supported SIP hard phones
and
I've been using a soft phone (SJLabs make one) to test
functionality.
Regards
Matt
From: Angel Gabriel
[mailto:[EMAIL PROTECTED] Sent: 29 March 2004
14:14To: [EMAIL PROTECTED]Subject:
[Asterisk-Users] testing functionality (how do I do this?)
I want to setup a quick * box, and test out
PS. What's the best source of documentation for the extensions.conf
file? I don't see any in the second draft of the Asterisk Handbook,
not much that I can really make sense of in the wiki page
(http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf)
which seems to
Good day all
Now
I want to install a complete pbx system on my linux box with windows
clients.
Now I have the a openline4 card and 4 lines,but what software do I
need,Asterisk running on the server and?and what for the clients,I
see in the config there is a sip provider config??
Thanks
Altus
Hi,
As I'm doing this, I'm considering installing an asterisk box at my
office (about 6-10 different phone stations) and would like to get
opinions on the best quality and/or most well-supported SIP hard phones
and SIP soft phone clients.
I had great luck with sipura spa-2000 adapters. They
On Mon, 29 Mar 2004, Iain Stevenson wrote:
- cheap option - Grandstream Budgetone - works well but current firmware is
buggy
Which one? I'm running one the latest image available at
http://www.grandstream.com/BETATEST/ (b14p4.54.zip) and my * and my GS are
working OK.
The 4.53 was buggy, but I
Tony,
Whatever phone or softphone you are using, you need to disable silence
suppression. Why? Dunno exactly. In the newest version of Xten, the
feature is Advanced System Settings - Audio Settings - Silence Settings
- Transmit Silence - Should be Yes.
may be it is this. This tip helped me.
Hmmm. Ok. I've compiled and installed ztdummy but I still get a really odd
MoH experience.
Any ideas what I might have done wrong?
Cheers
matt
-Original Message-
From: Matt Bridges [mailto:[EMAIL PROTECTED]
Sent: 29 March 2004 13:55
To: [EMAIL PROTECTED]
Subject: RE:
--On Monday, March 29, 2004 2:09 pm + Hermann Wecke [EMAIL PROTECTED]
wrote:
Which one? I'm running one the latest image available at
http://www.grandstream.com/BETATEST/ (b14p4.54.zip) and my * and my GS are
working OK.
The 4.53 was buggy, but I can't find a problem (so far) with 4.54
Hi all,
I configured Asterisk as shown in
http://www.voip-info.org/wiki-Asterisk+ISDN4Linux
The box running Asterisk under SuSE 9.0 Pro has a Eicon Diva Server BRI
2M ISDN card attached and it seems to be recognized by the system.
I added the following lines to:
* modem.conf
driver=i4l
...
Iain Stevenson wrote:
I'm running 4.50 because of adverse reports of 4.53 etc.
Is abbreviated dialling (aka Early Dial) working yet - it's been out of
commission for most firmware from 35 - 50 releases.
No. Tested it last week on '4.54 - still broken - but somewhere along
the line they fixed
In article [EMAIL PROTECTED],
Matt Bridges [EMAIL PROTECTED] wrote:
Hmmm. Ok. I've compiled and installed ztdummy but I still get a really odd
MoH experience.
Any ideas what I might have done wrong?
Firstly, in another post you talked about using mpg321. That doesn't
work with Asterisk.
Tony,
Sorry about that. The 321-123 was a typo. I'm running mpg123.
Grep hci /proc/modules returns usb-uhci
Cheers
Matt
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: 29 March 2004 16:09
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: MOH doesn't play
On Mon, 2004-03-29 at 15:44, Martin Mielke wrote:
Hi all,
I configured Asterisk as shown in
http://www.voip-info.org/wiki-Asterisk+ISDN4Linux
The box running Asterisk under SuSE 9.0 Pro has a Eicon Diva Server BRI
2M ISDN card attached and it seems to be recognized by the system.
I
We are looking for the cheapest way to show management we can use an * box as a VoIP
gateway from our Merlin Legend. There are no T1 avail in the Legend but we have some
analog Trunks available as backup for our T1. Since the X100P does not support DID and
it would be way too complicated to set
In article [EMAIL PROTECTED],
Matt Bridges [EMAIL PROTECTED] wrote:
Tony,
Sorry about that. The 321-123 was a typo. I'm running mpg123.
Grep hci /proc/modules returns usb-uhci
OK, you should be fine as far as software goes then. The next most likely
thing is the other suggestion that was
Hmmm. I recompiled ztdummy and it's working completely fine now... Very
odd
-Original Message-
From: Matt Bridges [mailto:[EMAIL PROTECTED]
Sent: 29 March 2004 16:19
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Re: MOH doesn't play
Tony,
Sorry about that. The 321-123 was
I may have downloaded an old CVS snapshot, but the following line seems
to be missing at channels/chan_iax2.c/load_module
ast_mutex_init(waresl.lock);
PauloHM
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Asterisk-Users mailing list
[EMAIL PROTECTED]
Thanks for your help Tony!
I don't know what made it work, but it works and I'm V Happy!
Cheers
Matt
In article [EMAIL PROTECTED],
Matt Bridges [EMAIL PROTECTED] wrote:
Tony,
Sorry about that. The 321-123 was a typo. I'm running mpg123.
Grep hci /proc/modules returns usb-uhci
OK,
I've had some minor problems with the spa-2000
but once you fixed the issues they are pretty solid unit
to use. so i'd recommend them highly but i can't wait for those new iaxy units from
digium to come out
those will hopefully resolve some of my firewall issues.
--
Best regards,
Frankie
This is an update on the status of an Avaya Definity w/ TN767E to
Asterisk T100P (no crossover)
If anyone has a working Avaya TN767 config print out I would really
appreciate it.
I have followed the config on the web, but cannot get it going and I
really can't confirm the Avaya config (but I
Jeb Campbell wrote:
Anyway, the only stuff off list was trying to debug the connection.
1. With a crossover there is no sync (YELLOW and RED alarms)
2. With standard cable I get a pri error that they think they are the
NET, but we are the NET.
(This is asterisk 1.0 stable and the directions from
On Mon, 2004-03-29 at 09:29, Victor Perez wrote:
We are looking for the cheapest way to show management we can use an *
box as a VoIP gateway from our Merlin Legend. There are no T1 avail in
the Legend but we have some analog Trunks available as backup for our
T1. Since the X100P does not
Hi folks,
I managed to compile the zaptel drivers on my MAndrake 9.0 machine. I can
do a modprobe zaptel and modprobe fcfxs as well as ztcfg -vv and all
appears to be fine.
FYI I'm using a no-name V90/V92 56k Intel chip'd voice modem which somehow
Linux sees as a Tigerjet ISDN card.
I have this
On Mar 29, 2004, at 11:06 AM, Eric Wieling wrote:
Jeb Campbell wrote:
Anyway, the only stuff off list was trying to debug the connection.
1. With a crossover there is no sync (YELLOW and RED alarms)
2. With standard cable I get a pri error that they think they are the
NET, but we are the NET.
Hi everybody =)
new to asterisk, i'm trying to understand if it supports pre-paid cards with
pin number, or if any software to manage pre-paid cards pins (better if
open source) may be linked to asterisk.
Or may be asterisk linked to a mysql/postgre/others database to update
check credit
Don't know about the Avaya, but the Fujitsu 9600 PBX needs:
Fujitsu T100P
7 4
8 5
4 1
5 2
Worth a try...
Jeb Campbell wrote:
This is an update on the status of an Avaya Definity w/ TN767E to
Asterisk T100P (no crossover)
If
Google: Asterisk Calling Card
- Brent
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, March 29, 2004 11:23 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] pre-paid (new to asterisk, pls don't shoot
[EMAIL PROTECTED] wrote:
I am trying implement two-stage dialing.
Scenario is following:
1. * Dials SIP agent
2. SIP agent answer the phone and provide dial tone
3. * Sends DTMF string
4. Bridge channel with calling party
I thought that something like:
exten =
Hello again,
I guess I solved part of my problems...
Now I can call an internal extension which matches a cell-phone using
the ISDN-card... but Asterisk refuses to call:
---
-- Executing Dial(SIP/mmielke-b282, Modem/g1:) in new stack
-- Called g1:
Mar 29 19:20:34
Hello people.
I've been thinking about a really cheap way to connect 2 PBX centrals i
got, using 2 * boxes , 2 x100p cards and my 256 DSL connection to the
Internet, I'm planning on connect the x100p cards on each establish a
iax connection between them, so for example, when i dial 234 (line would
When
asterisk receives a SIP INVITE request, it has a request URI as a field (in my
case: [EMAIL PROTECTED]).
The following message header of the SIP request also contains a To: field. In
my case the To: field is [EMAIL PROTECTED]
I'd like asterisk to "go to" extension 4121891 when it
Title: Chan_phone problems
Hi All,
I'm really new to Asterisk, and I'm having a little trouble with my test setup. Things are pretty simple so far:
Linux 2.4 kernel (Redhat 9)
Linux PhoneJack-Lite interface
What happens is that I get dialtone, but dialing doesn't seem to do
-This is my 'sip.conf' file:
;*
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming
Since there has been at least some discussion on this, I tried to purchase
SmartNet maintenance today (US) via telephone. Here's the results.
Cisco does not sell any maintenance agreements direct to a customer unless
the customer is already large enough to purchase items direct(including
On 05:43 AM 3/29/2004, Iain Stevenson wrote:
Lot's of references to this topic on the list - a quick search will provide
loads of feedback. At a top level:
- cheap option - Grandstream Budgetone - works well but current firmware is
buggy
- best option - Cisco - good * integration. Make sure
On Mon, 2004-03-29 at 11:26, Martin Mielke wrote:
Hello again,
I guess I solved part of my problems...
Now I can call an internal extension which matches a cell-phone using
the ISDN-card... but Asterisk refuses to call:
---
-- Executing Dial(SIP/mmielke-b282, Modem/g1:) in
Hi All,
I have an annoying problem. Out going SIP/sipphone.com
calls work fine. Internal calls work fine. However, incoming SIP calls
DIAL and ring, but send a busy signal when picked up. The same
happens if I take the SNOM200 out of the loop and just try to answer and
playback a
On Mon, 29 Mar 2004, Chris A. Icide wrote:
Actually after having gotten my hands on a Polycom IP600, I have to say
I much prefer it over the Cisco 7960 I've been using for almost 2 years
now.
Just curious, does the Polycom have a backlight on the LCD? That's one of
my wishlist items that I
What does your extensions.conf look like?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of pesb
Sent: 29 March 2004 18:48
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk + GrandStream SIP phones
-This is my 'sip.conf' file:
Steven Critchfield wrote:
[ snip ]
You should have a / instead of a : in the dial.
It doesn't help...
See error message:
---
Mar 29 20:34:06 WARNING[393232]: chan_modem.c:181 modem_call:
Destination g1/y requres a real destination (device:destination)
---
btw, __TRIM__ the
On Monday 29 March 2004 12:25, pesb wrote:
I have 2 SIP GrandStream phones, both phones are correctly
registered to the Asterisk server. But, when I try to make a call
from registered phone '1005' to registered phone '1004', dialing
1004, Asterisk responds with the 'Status: 404 Not Found'
is the log file:
Page 1 of /var/lib/asterisk/fax/new/20040329-234801-0755965128.tif:
1116 rows received
0 total bad rows
0 max consecutive bad rows
Rx page end detected
Changed from phase 5 to 3
Slow carrier up
Slow carrier down
Slow carrier up
EOP: 2f
EOP with final frame tag
In state 5
No backlight.
John
On Mon, 2004-03-29 at 12:30, Nate Carlson wrote:
On Mon, 29 Mar 2004, Chris A. Icide wrote:
Actually after having gotten my hands on a Polycom IP600, I have to say
I much prefer it over the Cisco 7960 I've been using for almost 2 years
now.
Just curious, does the
Try this small extensions.conf
Don't think I have missed owt.
My config files are here, you just need to add your own extension numbers.
http://www.codepipe.com/id25.htm
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of pesb
Sent: 29 March 2004 19:26
I have two Adtran 750's connecting our analog phones to asterisk. On
occasion, I get a channel that gets stuck off hook. 'show channels'
shows:
Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None)
And will just stay like that until the phone is manually picked up and
hung up again (or
The VON show has started off with a number of interesting announcements.
First among these is a big announcement from Pingtel that they have created
a not-for-profit corporation called SIPFoundry. This new company includes
Pingtel (which has recently open sourced their SIPExchange PBX), Vovida
I've just started having the same problem here today. I did and upgrade over the
weekend to
Zaptel-0.9.0 and the release candidate for Asterisk-1.0 CVS 3/28/04.
I have 6 Adtran 750 FXS_KS for all channels. 1 T-1PRI and one EM_W T-1.
-sb
-Original Message-
From: [EMAIL
Title: [Asterisk-Users] VON Update - Pingtel Creates new SIP Open Source Group
cut-n-paste from site:
(seems to me like a SIP instance of CLASS Asterisk)
Mission
The mission of SIPfoundry is to promote and advance SIP-related Open Source
projects. Through SIPfoundry the users,
how do you get the phone message button to light when there is a message?
eliot
-Original Message-
From: [EMAIL PROTECTED] on behalf of David J Carter
Sent: Mon 3/29/2004 1:58 PM
To: [EMAIL PROTECTED]
Cc:
Subject: RE: [Asterisk-Users]
I just built the latest * from CVS, spandsp 1i, and hacked app_rxfax
for the 'lock' parameters (I used 0 in all cases). The result of my
first fax transmission attempt is the following, which I post in case it
helps the developers any... The only thing I changed is the phone number
(TSI), which
Where and when is the rollout meeting? I'd love to attend.
Thanks!
Paul
Paul Mahler
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven M. Sokol
Sent: Monday, March 29, 2004 11:36 AM
To: [EMAIL PROTECTED]
Subject:
Me too.
Isamar
On Mon, 29 Mar 2004, Paul Mahler wrote:
Where and when is the rollout meeting? I'd love to attend.
Thanks!
Paul
Paul Mahler
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Monday 29 March 2004 13:58, Robinson, Eliot S. wrote:
how do you get the phone message button to light when there is a
message?
mailbox=1234
OR
[EMAIL PROTECTED] ; if you're not using the [default] context
; in voicemail.conf
-Tilghman
On Monday 29 March 2004 13:36, Steven M. Sokol wrote:
Can somebody out there take a look at the SIP Forge site and let us
all know what the crux of the organization is set to be? They are
having an open roll-out meeting tomorrow evening which should spell
out some of the goals of the
How do you configure extensions.conf to let you punch out to
VoicemailMain when an individual voicemail prompt has picked up?
We have a few extensions set up. voice mail is extension 8500 and we
have another extension for SIP on extension 12. SIP dials out fine.
SIP can dial 8500 and get
Looking for ideas on asterisk@ home. 1 snom
200 sip,
1 telephone line zatel 100, and 3
users.
According to Iain Stevenson:
Welcome to the very much less than wonderful world of Cisco software
support. When will those guys simply make the software downloadable
straight away from their website for a modest fee?
According to Chris HARIGA:
If you pay 8 USD for 1 year support
The rollout is tomorrow evening, Tuesday March 30 from 6:30 to 7:30PM at the
Hilton in Santa Clara, just across the street from the Conference Center.
It's being held in a ballroom (I don't have the name with me, I'll try to
post it later).
I hope to see you there.
Steve
-Original
Call your local independent computer
retailer and find either a retired PIII box or a low end Celeron box (I buy
them in single-unit quantities for $300 each). Order the X100P from Digium.
Configure. Call. Repeat if necessary.
What kind of ideas are you looking for?
Thanks,
Steve
Sounds like a codec mismatch to me. I had a similar
problem with ICH.
On Mon, 29 Mar 2004 19:23:15 +0100, jc wrote:
Hi All,
I have an annoying problem. Out going SIP/sipphone.com
calls work fine. Internal calls work fine. However,
incoming SIP calls
DIAL and ring, but send a busy signal
After doing cvs checkout -r v1-0_stable asterisk and typing the
usual make clean ; make install, I got these messages:
...
make[1]: Entering directory `/usr/src/asterisk/codecs'
make -C gsm lib/libgsm.a
make[2]: Entering directory `/usr/src/asterisk/codecs/gsm'
/var/spool/asterisk -o
Could you have asterisk running and not allowing you to overwrite while
trying to install? Do you have root rights to create files in the
asterisk folders?
John Chambers wrote:
After doing cvs checkout -r v1-0_stable asterisk and typing the
usual make clean ; make install, I got these
Hello,
I find that when 2 extensions are connected, and one of the extensions
hangs up on the call, the other will receive a busy signal (as if to
indicate that the call is over).
Does this sound like a config problem, or is it the default behavior of
*?
Example:
[ext-testing]
exten =
As you see, * generates no busy tone, it hangs up the channel. It's your
client which generates the tone. This is not something to be done from *.
Regards,
Doichin Dokov
Ryan Courtnage wrote:
Hello,
I find that when 2 extensions are connected, and one of the extensions
hangs up on the call,
Before you use old used crap. Look
at dell for a base server 279 after rebate.
http://www1.us.dell.com/content/products/features.aspx/pedge_400sc?c=uscs=04l=ens=bsd
Dave
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven M. Sokol
Sent: Monday, March
Title: RE: [Asterisk-Users] SoftFAX/spandsp - rxfax findings (spandsp-0.0.1i)
Hi,
One more step ahead, with:
- no problems receiving from the regular fax machines in the office,
- 3 attempts receiving from J2: 2 successes, 1 failure (page cutoff),
- 3 attempts receiving from Dialogic
Roderick Montgomery wrote:
* Have you purchased used hardware off eBay and need the latest software?
There's an inspection and relicensing process that can make your gear fully
legit.
Ummm, with a mountain of red tape surrounding the whole deal, and
basically at a cost that ensures most people
A license for a used 7960 is about a $115 hickey. Ouch.
http://www.z-buy.com/product.asp?item=ET-SWCCMUL7960
As they say in Vegas, thanks for playing.
John
On Mon, 2004-03-29 at 15:14, Roderick Montgomery wrote:
According to Iain Stevenson:
Welcome to the very much less than
Hi Shawn,
Simple really. Assuming your house is wired for data
litter your handsets about the house and then build the
* server.
When the analogue line rings have * ring all the SIP
phones at once. Whoever picks up first gets the call.
If no-one picks up after say 30 seconds forward the
call to
Hi,
I just got started with Asterisk. Installation was OK, no errors.
But how do I activate the IAX and SIP channels now? I loaded the
modules, but nothing happened, there's no connection to the
relavant ports.
Anything I forgot to do...?
Thanks in advance!
thomas
---
Thomas Schroeter //
Try to add a qualify= to sip.conf, and try to exec a sip show peers.
In spite of phones appears like register, if you use NAT, your firewall
can cut communication. Try the next:
Just after phone register call to it, and then wait for a minutes and
try to call again. Could you call first time
I have my *box running, I was looking for a little help with the scrpits. My
console dail's, my zap card will answer and I made a IAX call to digiam. But
my snom 200 phone does not work and I have 3 phatum users to answer for on
the zap teleco line.(for joe press 1, for joe2 press 2, and for joe3
I was wondering if there is any information on how to select which codec
is best to use between two * server. The local IP phones use G.711 to connect
to the local server. I usually find that using a low end codec like GSM
between servers will degrade the voice a lot, same with iLBC. Does
maybe something like;
exten = s,1,Dial(SIP/12,10)
exten = s,2,DigitTimout,3
exten = s,3,Background(vm-transfer)
exten = s,4,Voicemail,u12
exten = #,1,VoiceMailMain
Alternatively
exten = s,1,DigitTimout=3
exten = s,2,Background(silence)
exten = s,3,Dial(SIP/12,10)
exten = s,4,Voicemail,u12
exten
Hello, I have instaled a quadBri card in linux
box with asterisk. And i have frequently ip to isdn communications cuts
(interumptions).Any idea are
welcome.Thanks.
I have to add something:
When starting in debug mode, it says:
[chan_sip.so] = (Session Initiation Protocol (SIP))
== Parsing '/etc/asterisk/sip.conf': Found
== SIP Listening on 62.x.x.x:5060
== Using TOS bits 0
== Registered channel type 'SIP' (Session Initiation Protocol (SIP))
==
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