How normally SIP user is informed by having a new incoming voicemail
and then, how are they read their mails then
i have known that, asterisk will send a mail for the users. then, how to
configure the mail smtp and pop3 for asterisk to send mail then.
thanks
___
On Nov 29, 2005, at 12:25 PM, Michaël Gaudette wrote:
Hi,
I`m a beginning Asterisk and Sendmail user. I am trying to setup my
voicemail to send emails to a certain email address. It doesn't work,
and I
think I've figured out what it is. There is probably a spam-feature
at my
provider (that
I have setup a vanilla macro and dial plan example to use the new Page()
application.
I use the page context so that I can configure each extension uniquely,
to deal with the various different devices, if not I have a default
'CatchAll'. This configuration will support any and all channel types
t
btw, i've patched this part of code and now its working fine for me.
i'm going to upload it.
Paradise Dove
On 11/30/05, Kevin Hanson <[EMAIL PROTECTED]> wrote:
> Paradise Dove wrote:
>
> >>Yes with version 1.2. I have tried already with call-limit and the same.
> >>
> >>
> >i agree with you, it s
I haven't tried any other one since FOP does what I need.
- Waldo
On Nov 30, 2005, at 12:37 AM, Hiu Yen Onn wrote:
other than asternic.org???
do u have any others alternatives
Waldo Rubinstein wrote:
Look at http://www.asternic.org/
- Waldo
On Nov 29, 2005, at 10:40 PM, Hiu Yen Onn w
Richard Malcolm-Smith wrote:
I am so keen on getting the Kirk telecomms 600 system to hook to
asterisk, anyone know where to get one from that will ship to New Zealand?
You could try asking in the asterisk irc channel, there regularly are
some people from New Zealand in there, they might have
Paradise Dove wrote:
Yes with version 1.2. I have tried already with call-limit and the same.
i agree with you, it seems to be a bug which i've submited before (bug
#5281) but it's now closed by bug marshals!
It's not closed. It's suspended waiting input from you:
"Closing until
WHAT IS MISSING? I AM USING LINUX SUSE PRO 9
linux:/usr/src/libpri-1.2.0 # make
CC=gcc ./mkdep -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g
`ls *.c`
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -c -o
copy_string.o copy_string.c
gcc -Wall -Werror -Wstrict-
Hello friends,
I am using asterisk 1.2 with ooh323. I am using sip and h323 phones. My
question is I am using a Welltech FXO box and ip phones by Welltech. Do I
still need to configure zapata.conf and zaptel.conf which I read in the
documentation from asterisk pdf file downoladed from asteris
[EMAIL PROTECTED] 2.1 has a script that installs bristuff
for you. it's called install-bristuff
2.1 should be release soon.
--- Alejandro Vargas <[EMAIL PROTECTED]> wrote:
> I'm testing asteriskathome with an ISDN card
>
> 00:0a.0 Network controller: Cologne Chip Designs
> GmbH ISDN network
>
Matt Riddell wrote:
Colin Anderson wrote:
Steve Underwood also informed me about chan_fax
(http://www.sofaswitch.org/chan_fax/), I'll have a look.
This looks awesome please report back to the list on this if you get it
working correctly.
What's the idea of this? How does it
other than asternic.org???
do u have any others alternatives
Waldo Rubinstein wrote:
Look at http://www.asternic.org/
- Waldo
On Nov 29, 2005, at 10:40 PM, Hiu Yen Onn wrote:
Hi all,
I am one of the client of the SIP/Asterisk, connected via Xlite
client. How should i know the rest of
OK, fine, thanks. Now I have 1.0.9 still BRIstuffed-0.2.0-RC8h "pre-built"
don't know how with hisax harmless (eventhough blacklisted), but all runs.
Halas, I have my initial problem. When I dial zap, I just get a continous dial
tone
example: exten => _9.,1,Dial(ZAP/1,${EXTEN:1})
and
CLI> Execut
Look at http://www.asternic.org/
- Waldo
On Nov 29, 2005, at 10:40 PM, Hiu Yen Onn wrote:
Hi all,
I am one of the client of the SIP/Asterisk, connected via Xlite
client. How should i know the rest of the active SIP users? Are
there any graphical tools giving a list of the active sip peer?
Hello Rich,
On Tue, 2005-11-29 at 07:49, Rich Adamson wrote:
> Couple of other items to look at... the 'zap show channels' should look
> something like:
> pseudoinbound-bus-lin en default
> 1inbound-bus-dia en default
> I don't see the '
On Nov 29, 2005, at 9:27 AM, Mojo with Horan & Company, LLC wrote:
What's the 'format' line of the [general] section of your
voicemail.conf?
It's format=wav49|gsm|wav
You should try not to just tack one line on top of a long message to
list... ;~)
Marty
___
Hi,
I'm using Asterisk 1.2.0 with the latest ASTCC. When ever I try
calling out the astcc.agi script will not complete.
I've checked the astcc.agi, it does have 'HUP' Signaling before
load_config.
Has anyone encounter this issue?
Would appreciate any inputs. Thank you
AGI Debug
-
Chuck Bunn wrote:
> Hi,
>
> Yes that does work but in some cases it just seems it would be clearer
> (also less code) to be able to have them on one line...
But they are two seperate files. The only way you can play them as one is to
create one sound file that contains both entries.
--
Cheers,
Quoting Anthony Rodgers <[EMAIL PROTECTED]>:
> exten => s,4,BackGround(to-compose-a-message)
> exten => s,5,BackGround(press-1)
For the OP -- try this:
exten => s,4,Background(to-compose-a-message&press-1)
For Anthony:
Your version creates a split second gap in DTMF monitoring as the PBX trans
Hi,
Yes that does work but in some cases it just seems it would be
clearer (also less code) to be able to have them on one line...
Thanks
Anthony Rodgers wrote:
exten => s,4,BackGround(to-compose-a-message)
exten => s,5,BackGround(press-1)
doesn't work?
On Nov 29, 2005, at 3:41 PM, <[EMAI
At 9:43 AM +0100 11/29/05, Olle E. Johansson wrote:
John Todd wrote:
[snip]
> 2) The intervals between the INIVTEs after the 407 sequence are:
34ms, 30ms, 49ms, 91ms.
> This is _way_ too fast for response timers to be expiring for
reliable re-transmissions of INVITEs... isn't it? According
Denny Schierz wrote:
> hi,
>
> very often, when the caller hangs up the phone, the isdn phone rings
> without stopping. It seems, that asterisk does noch check, that the
> caller has hang up.
Strange ISDN should provide call progress information. Does misdn not parse it?
--
Cheers,
Matt Ridde
I don't think * was looking for alert-1 at the remote server. My best
guess would be that IAX server at the other ends actually *answer*
your call *before* Dial the destination channel. As soon as that
happens, * will jump to (alert-1,s,1) and playback the message before
the other end has a chance
Hi,
Yes that does work but in some cases it just seems it would be clearer
(also less code) to be able to have them on one line...
Thanks
Anthony Rodgers wrote:
exten => s,4,BackGround(to-compose-a-message)
exten => s,5,BackGround(press-1)
doesn't work?
On Nov 29, 2005, at 3:41 PM, <[EMAI
Mr. James W. Laferriere wrote:
Hello All , no zapata diredtory , tho zaptel README says many
of the testing programs require its libraries .
Please enlighten me . Tia , JimL
The zapata directory was not imported into SVN. If anything actually
does need it, you can get it from C
Hi all,
I am one of the client of the SIP/Asterisk, connected via Xlite client.
How should i know the rest of the active SIP users? Are there any
graphical tools giving a list of the active sip peer? thanks.
___
--Bandwidth and Colocation provide
Larry Alkoff wrote:
I've just heard about DECT which is used for about 50 million phones in
Europe and is just starting to appear in the US.
DECT stands for Digitally Enhanced Cordless Telephone
and supposedly has much greater range than other cordless telephony.
Additionally, you can purchase
Colin Anderson wrote:
>>Steve Underwood also informed me about chan_fax
>>(http://www.sofaswitch.org/chan_fax/), I'll have a look.
>
>
> This looks awesome please report back to the list on this if you get it
> working correctly.
What's the idea of this? How does it relate to spandsp, t38 patc
We had a similar problem a while back and found that it was being caused by
Hyperthreading. If you are using analogue cards then unfortunately you
need to
disable H/T if you haven't already done so.
You also need to confirm that your fxo/fxs card isn't sharing IRQ's with
anything. Don't trust
Any Luck with this? I'm getting frustrated. We need caller ID to be
able to do business properly.
Cheers
[EMAIL PROTECTED] wrote:
Actually, exactly now I am trying to do that also...
Isamar
On Fri, 25 Nov 2005, Aaron Anderson wrote:
Are there any kind of patches or experimental libra
- Original Message -
From: "Jeff Busch" <[EMAIL PROTECTED]>
Hello,
I am running the following configuration:
2.8ghz P4 with 1GB of RAM
Audiocodes MP-108 connected to 5 POTS lines
Polycom IP-500 phones
[EMAIL PROTECTED] 1.3 (this is Asterisk 1.0.9)
End users are complaining of an ec
We have the same problem lately we thought maybe our upgrade and
testing of the .13 firmware.. but we are running 20 phones on a p4 2.6
w/512 4 PSTN lines on TDM400P and have 3 fat client pc's and 7
pcexpanions ( http://ncomputing.com/ ) running to a fat client. We did
not have the issue on fi
exten => s,4,BackGround(to-compose-a-message)
exten => s,5,BackGround(press-1)
doesn't work?
On Nov 29, 2005, at 3:41 PM, <[EMAIL PROTECTED]> wrote:
Hi,
Is it possible to paste phrases together and if so how do I separate
each
phrase.
exten => s,4,BackGround(to-compose-a-message,press-1)
> I've noticed that if I enable the jitterbuffer in iax.conf with Asterisk
> 1.2.0 that Asterisk stops responding to incoming DTMF frames for calls
> between Teliax and my server. I've used "iax2 debug" and Ethereal to
> confirm that Teliax is, in fact, sending the frames.
>
> I only have two
> On Mon, 2005-11-28 at 12:24 -0800, Innocent Evil wrote:
>> What is the purpose of cdr_manager.conf?
> cdr_manager.conf allows you to configure asterisk to send call detail
> records (cdr) via the Manager API.
>
>> How I can configure it?
> to enable CDR via Manager API a cdr_manager.conf looks
Hello All , no zapata diredtory , tho zaptel README says many
of the testing programs require its libraries .
Please enlighten me . Tia , JimL
$ svn checkout http://svn.digium.com/svn/zapata/trunk zapata
svn: PROPFIND request failed on '/svn/zapata/trunk'
svn: Could no
This definitely could be the issue. I am running 15 total devices (7
IP500 phones and 7 PC's along with a networked fax/scanner, and the
Asterisk Server) through a single 16 port switch.
We run one MS Access app on almost all the desktops that is a
client/server app that creates a lot of traffi
I've noticed that if I enable the jitterbuffer in iax.conf with Asterisk
1.2.0 that Asterisk stops responding to incoming DTMF frames for calls
between Teliax and my server. I've used "iax2 debug" and Ethereal to
confirm that Teliax is, in fact, sending the frames.
I only have two IAX2 connec
Jeff Busch wrote:
You are correct. Inside means the IP Network and Outside means the PSTN
accessible via the Audiocodes MP-108 gateway.
No QoS.
We've seen echo on congested LANs within our Enterprise. I'm not sure if
this fits
what your seeing or not. We've placed phones in their own vl
You are correct. Inside means the IP Network and Outside means the PSTN
accessible via the Audiocodes MP-108 gateway.
No QoS.
Thanks - Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Blair
Sent: Tuesday, November 29, 2005 4:20 PM
To: Asterisk
Jeff Busch wrote:
Hello,
I am running the following configuration:
2.8ghz P4 with 1GB of RAM
Audiocodes MP-108 connected to 5 POTS lines
Polycom IP-500 phones
[EMAIL PROTECTED] 1.3 (this is Asterisk 1.0.9)
End users are complaining of an echo and static on the inside end (the
internal side)
Patrick May wrote:
>On Tue, Nov 29, 2005 at 06:52:21PM +0100, Francesco Peeters wrote:
>
>
>>On Tue, November 29, 2005 18:36, [EMAIL PROTECTED] said:
>>
>>
>>>Hello All,
>>>It seems that voicepulse is not taking any new orders on the standard
>>>service plans (though vp connect seems unaffec
Hi,
I think you must spécify a full qualified domain name.
The destination mail server try to resolve your Linux Box domain name , and
he can't because of you domain name "localhost.localdomain"
If you haven't a domain name you can create a dyndns.org domain linked with
you linux public IP.
S
Hello,
I am running the following configuration:
2.8ghz P4 with 1GB of RAM
Audiocodes MP-108 connected to 5 POTS lines
Polycom IP-500 phones
[EMAIL PROTECTED] 1.3 (this is Asterisk 1.0.9)
End users are complaining of an echo and static on the inside end (the
internal side), but the outside end o
On Tue, Nov 29, 2005 at 06:52:21PM +0100, Francesco Peeters wrote:
> On Tue, November 29, 2005 18:36, [EMAIL PROTECTED] said:
> > Hello All,
> > It seems that voicepulse is not taking any new orders on the standard
> > service plans (though vp connect seems unaffected) due to the fcc
> > rulings.
>
Hi,
Is it possible to paste phrases together and if so how do I separate each
phrase.
exten => s,4,BackGround(to-compose-a-message,press-1)
and
exten => s,4,BackGround(to-compose-a-message|press-1)
do not work...
Thanks
___
--Bandwidth and Colocati
SIP v7.5 changes the way the time server is addressed. The release notes and
the Asterisk site have the details. You also need to make sure your time
server
supports whichever mode you select. I think Cisco recommends anycast but
we use unicast and it works fine.
Jeremy Koski wrote:
Make s
Hi there:
I have a running asterisk ver 1.2 box. It supports the following SIP methods:
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
If I want to disable OPTIONS method, what can I do on the sip setting
files? Or I need to have an alternative way to reach it?
Thank you for your help.
Wilso
H,
Not sure if this is normal but I thought the coma ',' was replaceable by the
pipe command '|' and vice versa? When I used comas instead of the pipe command
in AgentCallbackLogin certain SIP phones do not here the operator prompts when
calling the agent extension. Is this normal - I thought the
Hi,
Not sure if this is by design or my error... Several agents are logged in. One
of the phones was turned off without the agent logging off first. After the
phone was powered down all calls routed to the powered off agent and no other
phones rang. Is there a way to turn this behavior off. (I wan
You have to have a login to the Cisco site to download the firmware.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Riek
Sent: Tuesday, November 29, 2005 2:02 PM
To: asterisk user list
Subject: [Asterisk-Users] Cisco 7970
I have the same pro
Make sure in your cnf file you have the following line:
sntp_mode: unicast ; unicast, multicast, anycast, or
directedbroadcast (default)
We experienced some problems with NTP and the phone crashing with 7.3 and
7.5, so we are currently running 7.4, FYI.
On Tue, 29 Nov 20
I have the same problem after doing a factory reset.
Does anybody have the website link to download
firmware for the Cisco phones?
Thanks,
John Riek
I ran into this same problem the other day. What you
need to do is
put all firmware files in the tftp root directory. The
trick with the
fil
Thanks Colin. That makes sense, but how do I modify this? I am no Linux
expert, but the passwd file doesnt seem to conatain any SMTP configuration.
When you said "run non-root", you meant Asterisk or Sendmail running as
non-root?
Mike
-
You can also modify the passwd file in /etc to put a friendly name on the
SMTP envelope instead of "root" b/c some spam filters will block even if the
email address is ok but the sender is "root"
Best practice is to run non-root though. I have an "Asterisk" user and an
"Asterisk" group. If you mo
Hi all, I'm trying to change the keys associated with the blind transfer
function. I've been mucking around in features.conf, but nothing I do
seems to make any difference ( and I've tried to intentionally break it
). I have restarted the * server between each modification.
Is this a known t
Hi,
I’m experiencing an issue with the CP-7940G phones
where upon a reset the time and date are displayed on the top row of the LCD
screen, but after approximately 30 minutes the time and display disappear,
while otherwise the phone continues to work as normal. If I unlock the
phone se
Hi:
I have a cellsocket that converts the digital line to
analoge line,but is there a digital cellsocket?
__
Yahoo! DSL Something to write home about.
Just $16.99/mo. or less.
dsl.yahoo.com
___
I tried that, didn`t do anything. My guess is that the serveremail line
changes the name in the from field, but not the MAIL FROM: call in SMTP.
Mike
> It seems that in both the 1.0 line and the 1.2 line, the [general]
> section of voicemail.conf has an option:
>
> ; Who the e-mail notificati
I have a rather curious integration problem. I need to direct a call
connection based on the codec used for the connection.
If my softswitch attaches to the Asterisk server using G729 I toss the
connection into a requested conference - that works fine.
On occasion my softswitch will attach to th
It seems that in both the 1.0 line and the 1.2 line, the [general]
section of voicemail.conf has an option:
; Who the e-mail notification should appear to come from
[EMAIL PROTECTED]
Moj
Michaël Gaudette wrote:
Hi,
I`m a beginning Asterisk and Sendmail user. I am trying to setup my
voicemai
> Yes with version 1.2. I have tried already with call-limit and the same.
i agree with you, it seems to be a bug which i've submited before (bug
#5281) but it's now closed by bug marshals!
by reading chan_sip.c u will find out that in function
update_call_counter() it first tries to update th
Hi,
I`m a beginning Asterisk and Sendmail user. I am trying to setup my
voicemail to send emails to a certain email address. It doesn't work, and I
think I've figured out what it is. There is probably a spam-feature at my
provider (that I am using as smart host in sendmail) to not accept emails
Hello All,
I am using * 1.2, BRIstuff 0.3 PRE1, Dual HFC-PCI, 1x TE, 1x NT
I am using DECT phones on a Siemens ISDN phone/DECT-base.
My dial options are rTtWw, automon=*1, blindxfer=##
Whether I am calling (to my cell) or being called (from my cell), only the
caller can initiate recording or tran
Yes with version 1.2. I have tried already with call-limit and the same.
On 11/28/05, Kevin Hanson <[EMAIL PROTECTED]> wrote:
Alvaro Parres wrote:> Hi list...>> I have been testing the hint extension. And i detect
> that when i have in the sip.fg of the extension the> incominiglimit=X (any n
Thanks Colin, this is a fantastic list! All I need to do now is get my
butt in gear and set up the box(es)!
I am using this dialplan with DID's to great effect, I have 130 guys doing
exactly what was discussed here. After 12 seconds ringing their SIP or IAX
client, the dialplan calls the cell
Hi Johann,
we engineered QueueMetrics out of the queues of * version 0.7, but never
found that origposition argument. And it's not present in our current 1.2.
Where did you find it?
Yours
l.
In data Tue, 29 Nov 2005 19:57:47 +0100, Johann
<[EMAIL PROTECTED]> ha scritto:
Entries in the q
>Steve Underwood also informed me about chan_fax
>(http://www.sofaswitch.org/chan_fax/), I'll have a look.
This looks awesome please report back to the list on this if you get it
working correctly.
___
--Bandwidth and Colocation provided by Easynews.c
I am using this dialplan with DID's to great effect, I have 130 guys doing
exactly what was discussed here. After 12 seconds ringing their SIP or IAX
client, the dialplan calls the cell automatically, during working hours. If
they don't pick up after 18 seconds, voicemail. After hours, both phones
Miguel Soto wrote:
Is the right behavior of the IAXmodem to display
"Registration completed successfully" and "remote hangup" many times?
You'd have to show me an example for me to say for certain, but my guess
is that if it looks wrong to you then it probably is wrong. This output
shoul
Anyone know if this can be made to work?
I've only been able to get SIP-SIP call pickup to work.
Steve
---
>>> as far as I know, no.
Il lun, 2004-07-05 alle 18:56, Adolfo R. Brandes ha scritto:
> I've looked in the obvious places but haven't found a definitive
> answer to the followin
Hi Niklas
Thanks for this information I will be sure to follow it.
Many Thanks
Scott Pinhorne
Niklas Larsson wrote:
On Tue, 29 Nov 2005 06:14:54 +, scott wrote:
Is anyone using a vegastream product with asterisk? I have various
numbers coming into the vegastream vega400 and was after so
Adam Goryachev wrote:
>
> Don't assume that we read this list every 5 secs I haven't read the
> mailing list since last week
You're right, thanks for your reply.
>
> In any case, you have two options:
> 1) Do it with meetme like you do now...
Lee Howard, the author of IAXmodem agrees w
Alejandro Vargas schrieb:
2005/11/29, Tomasz Chmielewski <[EMAIL PROTECTED]>:
you have to use bristuff-0.3.x (from www.junghanns.net) with a HFC card,
not HiSax (well, technically, you could use HiSax too, but avoid that if
possible).
I prefered to use hisax because it is already included in
Entries in the queue_log file do not match what the documents say. The
COMPLETECALLER and COMPLETEAGENT events do not have the 3rd agrument of
origposition. I'm using Asterisk 1.0.9 currently(will be upgrading
shortly). I've checked and this should be done by the old stable
version we are ru
Hi everybody:
Is the right behavior of the IAXmodem to display
"Registration completed successfully" and "remote hangup" many times?
Regards
Miguel
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE
Channel: Local/[EMAIL PROTECTED]Callerid: 01612370660MaxRetries: 5RetryTime: 300WaitTime: 45Context: serverdownExtension: sPriority: 1On 29 Nov 2005, at 15:39, Tony Spencer wrote:I'm a bit of newbie to Asterisk so I'm not to sure.I was just given the task to try and make this work.You could be corr
Are there any example configs? Or does anybody have a default config
for this setup:
1 analog digium clone card for an analogue line (my home line)
Several sip phones (a few of them on the outside of my lan (NAT fw
between) and 2 insde my lan)
Or a simple way of configging through a frontend/scr
Jason Marshall wrote:
OK, then this is easy. Instal Asterisk in the central location, along
with a Sipura SPA-3000. Configure that unit to answer the incoming
POTS line and act as a VOIP gateway for Asterisk. Then configure two
additional SPA-3000 units, one at each employee's location. Then,
On Mon, 2005-11-28 at 12:24 -0800, Innocent Evil wrote:
> What is the purpose of cdr_manager.conf?
cdr_manager.conf allows you to configure asterisk to send call detail
records (cdr) via the Manager API.
> How I can configure it?
to enable CDR via Manager API a cdr_manager.conf looks like this:
;
Since upgrading to 1.2 I'm seeing the following iin my
/var/log/asterisk/messages:
Nov 29 11:50:20 NOTICE[13094] app_dial.c: Unable to
create channel of
type 'Zap' (cause 17 - User busy)
Nov 29 12:02:06 WARNING[12977] chan_iax2.c: chan_iax2:
ast_sched_runq
ran 249 scheduled tasks all at once
Th
But, star at least works. I've got *xxT in my digitmap and it caught
*69. In fact, my 1.5 admin guide refers to Section 2.1.5 of RFC 3435,
the MGCP rfc, which does allow the * to be used
Moj
Rich Adamson wrote:
I'm trying to implement some of the "star services" such as *61 for
weather or *7
On Tue, November 29, 2005 18:36, [EMAIL PROTECTED] said:
> Hello All,
> It seems that voicepulse is not taking any new orders on the standard
> service plans (though vp connect seems unaffected) due to the fcc
> rulings.
>
> We'll see what happens, anyone having similar problems with other
> servic
On Tue, November 29, 2005 16:04, Giovanni Miano said:
> zahfc mode loaded ?
> try lsmod to verify
>
> try ztcfg -vvv
> sleep 3
> ztcfg -vvv
>
Also helpful is
cat /proc/zaptel/*
This'll tell you whether zaptel is loaded, whether the channels have been
defined, and what their status is...
On 00:24, Tue 29 Nov 05, bram kortleven wrote:
> Are there any example configs? Or does anybody have a default config
> for this setup:
>
> 1 analog digium clone card for an analogue line (my home line)
> Several sip phones (a few of them on the outside of my lan (NAT fw
> between) and 2 insde my
On 18:26, Mon 28 Nov 05, Sascha Deri wrote:
> I made an error in what I previously wrote. What actually works in v1.2 is:
>
> exten => asterisk,1,VoicemailMain(${CALLERIDNUM})
>
> Which is what Michael originally wrote. My bad!
:) To err is human :)
I know for sure it had to work since I copi
On Tue, 29 Nov 2005 06:14:54 +, scott wrote:
> Is anyone using a vegastream product with asterisk? I have various
> numbers coming into the vegastream vega400 and was after some
> exmaple config for use with the asterisk server so it can perhaps
> reister with the vega and recieve these number
On 09:46, Tue 29 Nov 05, Erik wrote:
> Leif Neland wrote:
> >
> >
> >
> >
> >> On 08:48, Tue 29 Nov 05, [EMAIL PROTECTED] wrote:
> >>
> >>> >From memory (at a previous installation) you will need a newer
> >>> version >of
> >>> Asterisk than 1.09 for the lights to work.
> >>
> >>
> >> on 1.0.9
Hello All,
It seems that voicepulse is not taking any new orders on the standard
service plans (though vp connect seems unaffected) due to the fcc
rulings.
We'll see what happens, anyone having similar problems with other
services as of today?
Greg
___
Are your interrupts getting hogged by anything else? I'd recommend
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting if
you haven't already read it. Have you tried booting with noapic kernel
option? You may then have to shuffle cards around to make your sangoma
not share an
Actually, why not:
exten => *67XXX,1, {etc}
-Original Message-
From: Steven [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 29, 2005 9:45 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Caller ID Block (*67)
Could you just use a different start number?
9 t
Actually, Matteo meant zaptel drivers, not motherboard or chipset
drivers from ASUS :)
Mojo
gincantalupo wrote:
Hi Matteo,
thanks for answering, your advise seemed right but no pci or motherboard
driver is avalaible on ASUS site.
I think we'll use another motherboard.
This is another motherb
What's the 'format' line of the [general] section of your voicemail.conf?
Martin Joseph wrote:
On Nov 28, 2005, at 3:55 PM, BJ Weschke wrote:
On 11/28/05, Martin Joseph <[EMAIL PROTECTED]> wrote:
I am only able to get comedian voicemail (ie dialing 1234) to
record or
playback messages if I
OK, then this is easy. Instal Asterisk in the central location, along with a
Sipura SPA-3000. Configure that unit to answer the incoming POTS line and act
as a VOIP gateway for Asterisk. Then configure two additional SPA-3000 units,
one at each employee's location. Then, configure Asterisk (I re
I risolved my problem: I have kernel source in /usr/src/linux-2.4.686 instead of /usr/src/linux, therefore the qozap.c doesn't compiling.ThanksTzafrir Cohen <[EMAIL PROTECTED]> ha scritto: On Tue, Nov 29, 2005 at 03:56:24PM +0100, asterisk183 wrote:> I am trying to install the qozap driver
Hi,
I am trying to execute the following asterisk command from one of my AGI script.
By providing 'C' flag, I exected CDR would reset.
Problem is, CDR was reset but CDR didn't grab destination number (extension)
from the Dial command.
Well my AGI script was executed after answering a call on a c
Is there a way to monitor zaptel errors with something like Nagios?
I have a TE405P and seldomly I see messages like this:
Zaptel: Master changed to TE4/0/1
wct4xxp: Setting yellow alarm on span 4
wct4xxp: Clearing yellow alarm on span 4
which means that somehow the T1 went down and came back u
Rich Adamson wrote:
Thanks for the heads up. More dissappointing is that the E/F card is
the newer card purchased. Where can I go to see when certain revisions
were released? Surprising that the newer card just purchased (to me)
is the older rev :(.
You can probably search
Could you just use a different start number?
9 to dial out. 8 to dial out with blocked callerID.
Then just preface the callerID block code for the Telco.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - -
On Tue, Nov 29, 2005 at 03:56:24PM +0100, asterisk183 wrote:
> I am trying to install the qozap driver, but when I doing:
> make all
> the shell command show error in qozap.o.
>
> What can I doing for compiling qozap.o?
>
> Thanks
Start by giving the telepathy-chalanged among us som
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