Hi,
I've tried upgrade my asterisk to 1.0.9...
It's now seemed that asterisk is more stable but it's still dead by
itself occasionally..
Output from gdb yield this:
...
Reading symbols from /lib/libgcc_s.so.1...done.
Loaded symbols for /lib/libgcc_s.so.1
#0 0x00a597a2 in _dl_sysinfo_int80
Am Mittwoch, 14. September 2005 19:12 schrieb Sander:
In queues.conf
; How long do we let the phone ring before we consider this a timeout...
;
timeout = 15
This is set in queues.conf
But this is just the function how long the phones will ring you should not
set this option to long or
Hi!
On 09/14/05 16:51, Johann Steinwendtner wrote:
Hi !
Asterisk sends a RELASE COMPLETE with cause code 34. It seems that
Nortel expects a RELEASE message in this state. The conversion
is done in the protocol engine of the MSDL.
Why would you want the cause code 34 to be sent ? Do you need a
Hi,
if your LEDs are off seems like you haven't loaded some module (zaptel
or wctdm)...or forgot to make a ztcfg.
Type lsmod to check if all modules are correctly loaded.
Giorgio
Innocent Evil wrote:
All of sudden my FXS module is not working.
I have a TDM card with one FXS and one FXO,
Yeah sorry about that. But I didnt see my message in the list, so I thought it
didn't ame through.
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Dave Cotton
Sendt: 14. september 2005 19:45
Til: Asterisk Users Mailing List - Non-Commercial
I have the exact same problem. TXFAX is fine. It's someone in rxfax
that's the problem as my system going into receive mode then hangs up.
Odd thing is I had this working before but now it doesn't.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi,
I'm currently trying to originate a call with 2 variables set. I tried
doing it via manager API and call File and both failed, because the vars
were not separated. I'm using Asterisk 1.2_beta1 on this machine
Can anyone here verify wether this is a bug or just a stupid error on my
part?
Hi Sean,
This is what I've got in my zaptel zonedata.c file for a small * box in
Dublin:
{ 18, ie, Ireland, { 400, 200, 400, 2000 },
{
/* Dialtone = 400//425//450 */
{ ZT_TONE_DIALTONE, 425 },
{ ZT_TONE_BUSY, 425/500,0/500 },
/* Ringtone =
Michael George wrote:
Hello, all!
I'm looking at the wiki page and info on the mailing list and I'm getting
conflicting info...
I am using the manager API from the telnet CLI and I am testing creating calls
with it. I login with events: on and I can originate calls just fine.
However, when
Morten Isaksen wrote:
On 9/11/05, Josip Gracin [EMAIL PROTECTED] wrote:
Is it legal to use RedirectAction to redirect a call that is waiting in
a queue?
It works for me.
Thanks! It does for me too now. It didn't work initially because I've
not set priority in RedirectAction.
Hi.
I'm dialling two numbers - one that's unobtainable, one that's busy.
${DIALSTATUS} is coming back ANSWER each time right before the channels hang up.
Am using the following dialplan macro to dial out.
[macro-advdial]
exten = s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds
hi,
afaik, the action-id provided with the OriginateAction should only show up in
the OriginateSuccess or OriginateFailure event. Intermediate events that are
generated when the channels are create will NOT carry the action-id of the
originate.
The async flag tells asterisk to process
In article [EMAIL PROTECTED], Joerg Lauer [EMAIL PROTECTED] wrote:
Hi,
I'm currently trying to originate a call with 2 variables set. I tried
doing it via manager API and call File and both failed, because the vars
were not separated. I'm using Asterisk 1.2_beta1 on this machine
Can
In article [EMAIL PROTECTED],
Mark Edwards [EMAIL PROTECTED] wrote:
I'm dialling two numbers - one that's unobtainable, one that's busy.
${DIALSTATUS} is coming back ANSWER each time right before the channels hang
up.
[...]
ITSP is terminating outbound calls to me via IAX2.
Aha,
Hi,My asterisk setup was working when using a Wildcard T100P. I recieved incoming calls and was able to make outgoing calls. I recently decided to change my card to a TDM01B one (one FXO). However when I dial a working extension, I can see on the console that the call goes through sucessfully, but
Sip to sip calls are fine, both local on Asterisk and over a SIP
gateway, however some people who call on the PSTN line say we are very
queit and vice versa, can the volume be turned up on the PSTN line?
The volume buttons on the VoIP phones only turns up the others voice,
so this is a fix for
Hi,
I have a *IAX* phone connected to a LAN and I want to connect to it to
make calls using an Asterisk server located inside another LAN behind a
router (* hasn't a public IP)
I bought a IAX phone because SIP had problems with nat and so on. ::))
How should I configure the IAX phone and
Tirpák Miklós schrieb:
Yes. 34 is required by the Nortel to send the call to an alternative
destination.
Cause 38 or 42 triggers the rerouting also for both options.
Hans
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Hi Paul
There are two settings in zapata.conf called txgain and rxgain. You can set
these to adjust the volume on your PSTN lines. They can be set in db or as a
percentage.
Garth
--- Paul Goodyear [EMAIL PROTECTED] wrote:
Sip to sip calls are fine, both local on Asterisk and over a SIP
Hi,
one of the sip-extensions we created always returns busy when someone
tries to call the phone. The extension itself can place calls.
We're using snom360 phones with the latest firmware. On every one of
those phones when we register with the sip-extension, we've experienced
the same problem.
In a remote location (solar powered) I have an old Gateway 333mhz
notebook running Asterisk and using a PCMCIA AVM Fritz ISDN card. With
the AVM CAPI it works perfectly with chan_capi.
Derek
Julien Goodwin wrote:
Does anyone know of any USB ISDN adapters that work with Asterisk. My
gateway
I thought the txgain, rxgain was purely for echo settings.
Is there a rough guide to this process, or is it a simple case of
changing values and testing them?
Thanks.
On 9/15/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi Paul
There are two settings in zapata.conf called txgain and
I am also finding that I am not getting caller id from the 3000 - I do
get a prefix if I enter it so I am getting the data but the 3000 is not
picking up the incoming callerid from the pstn line.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax:
Hi guys, This is my first post. I'm configuring asterisk together with SER. I'm testing the voicemail feature of asterisk. When I dial the extension it supposed to play first a voice prompt before recording starts. But I can't hear any sounds from Asterisk. Here are my configs: sip.conf
As subject, (i've updated spandsp to latest version) and this is the log:
Sep 15 13:06:50 VERBOSE[14085]: -- Attempting call on Zap/g0/2479 for
application txfax(/var/tmp/ast_fax-1126782409.10240.1804289383.0|caller)
(Retry 1)
Sep 15 13:06:50 DEBUG[14085]: Using channel 1
Sep 15 13:06:50
It appears that 3-way calling is only possible if you
connect an IP phone with this feature to Asterisk. I
can't find any (free) IAX soft phones that do 3-way
calling. Why is 3-way calling not a standard feature
of Asterisk and why don't any IAX soft phones offer
it? Is it such a complicated
gincantalupo ha scritto:
I have a *IAX* phone connected to a LAN and I want to connect to it to
make calls using an Asterisk server located inside another LAN behind a
router (* hasn't a public IP)
I bought a IAX phone because SIP had problems with nat and so on. ::))
How should I configure
I am looking for a China DID so my family in China can call me in UK.
I am looking for an option where the providor can forward me the calls
directly to my * box by SIP or IAX2 (fixed IP).
Any help would be appreciated.
Steven Ducat.
___
--Bandwidth
Title: TE110P - [EMAIL PROTECTED] Install Problems
I
figured it out. The old system (Televantage 3 and 4 I think) has limited
specifications on the T1. After setting up the system, I was able to send
and recieve calls. I still have some work to do like figuring out faxing
and a floating
Hi
On my FC3 box with asterisk 1.0.9MusicOnHold is not working.
It starts and stops immediately...
An unknow option mono comes...from where it is originating.??
As there is nothing written in .conf file.
Console output is below:
I am using mpg123 version 0.59r.
Although I am able to
It's because mpg123 is being passed an option --mono. Looks to me (a cursory
guess) that your asterisk installation is trying to force mono sound, and
mpg123 doesn't like it.
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Gurminder Arora
-Sent:
I was wondering if there was a guide to tweaking the linux kernel so
that Asterisk runs best. I know in 2.6.13 they added adjustments to
change the timer frequency from 100, 250 and 1000Hz. But then there are
the different preempt options and the high resolution timers patches.
Has anyone
On Thu, September 15, 2005 6:56 am, Chris Mason (Lists) wrote:
I am also finding that I am not getting caller id from the 3000 - I do
get a prefix if I enter it so I am getting the data but the 3000 is not
picking up the incoming callerid from the pstn line.
I'm getting CID some of the time.
I've been using 2 SPA3000's for several months. Both are running
3.1.3(GWa) software. I do not have any issues with echo. One box is
used to bring in a SBC POTS line and the other is connected to my
Cisco ATA186 from Vonage. The 3000 connected to SBC line relays CID
info, I have never been
Good day all
Is it possable to set asterisk up as a cdr server for other voip units
We got a quintum dx here and its got a option to log to a cdr server on
port 9002
Thanks
Altus
___
--Bandwidth and Colocation sponsored by Easynews.com --
this is not a solution, more a workaround, you can try using svscan service, so when down will automagically briged up.On 9/15/05, stevanus
[EMAIL PROTECTED] wrote:Hi,I've tried upgrade my asterisk to
1.0.9...It's now seemed that asterisk is more stable but it's still dead byitself
Altus Snyman ha scritto:
Good day all
Is it possable to set asterisk up as a cdr server for other voip units
We got a quintum dx here and its got a option to log to a cdr server on
port 9002
That's not a job for Asterisk! The Tenor can connect to a server and
send CSV records over TCP, so
The CID with the Cisco isn't a Cisco issue. It's actually an issue based on the way Vonage passes CID through the Cisco. It doesn't follow the same standard that LECs and others use.
I tried to get this going with an SPA3000 at first as well and never really could get it to go right without
I send CDR to a MySQL database on another machine. Might that work for you?
--On Thursday, September 15, 2005 3:24 PM +0200 Altus Snyman
[EMAIL PROTECTED] wrote:
Good day all
Is it possable to set asterisk up as a cdr server for other voip units
We got a quintum dx here and its got a option
I just did it by ear. Got it right in less than 5 minutes.
--- Paul Goodyear [EMAIL PROTECTED] wrote:
I thought the txgain, rxgain was purely for echo settings.
Is there a rough guide to this process, or is it a simple case of
changing values and testing them?
Thanks.
On 9/15/05,
I tried switching out the 3000 with a X100P card, but the card would
never recognize when the caller hung up. So, I have to keep the
3000. The X100P worked fine with the POTS line.
At 08:52 AM 9/15/2005, you wrote:
The CID with the Cisco
isn't a Cisco issue. It's actually an issue based on the
I am trying to figure out how to try different VOIP providers if they
aren't able to terminate the call because they don't offer service to
that dialing area.
The error that gets logged to the console is:
Sep 13 00:01:43 WARNING[22093]: chan_iax2.c:6835 socket_read: Call
rejected by x.x.x.x: No
Asterisk don't running, because show this message
WARNING[6949]: chan_sip.c:8865 reload_config: Section 'authentication' lacks
type
WARNING[6949]: chan_iax2.c:7491 load_module: Unable to open IAX timing
interface: No such file or directory
WARNING[6949]: chan_skinny.c:2587 reload_config:
Hello all
I have a question about Grandstream HandyTone 386
can Grandstream HandyTone 386 make 2 sim. calls with g729 codec in same time
Iyi Calismalar.
Ugur GUNCER
smime.p7s
Description: S/MIME cryptographic signature
___
--Bandwidth and
it looks to me like you haven't loaded the zaptel.o or ztdummy.o kernel
modules...
On Thursday 15 September 2005 16:10, [EMAIL PROTECTED] wrote:
Asterisk don't running, because show this message
WARNING[6949]: chan_sip.c:8865 reload_config: Section 'authentication'
lacks type
What is the best way to get my CDR information into Oracle?
Is yada there for the best choice?
How stable is yada for this
Thanks Han
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Asterisk-Users mailing list
Derek,
could you give me some details regarding the solar power supply you're using
for your installation?
Thanks!
Jörg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Derek Conniffe
Sent: Thursday, September 15, 2005 12:28 PM
To: Asterisk
On Wednesday 14 September 2005 12:34, Colin Anderson wrote:
I'm curious as to this obsession with uptime is. All of the posts of this
type are along the lines of After X days, Y thing does not work but if I
reload or reboot, it's OK - so why not cron a reboot? Is it considered bad
form or
Show this message
adduser: Warning: The home dir you specified already exists
adduser: The user 'asterisk' already exist, and is non a system user.
dpkg: errrore processando asterisk-cah-capi (--configure):
il sottoprocesso post-installation script ha restituito un codice di errore 1
We are trying to configure two GR303 trunks from an Asterisk
server with a quad span card to a class 5 softswitch (Taqua OCX/TEKELEC
T7000). We show the T1s up but errors on the TMC EOC channels. Has
anyone configured GR303 before and/or setup this type of configuration.
Thanks!
Paul
That is an issue with Vonage not providing remote disconnect supervision through the ATA.The way I got around that was not to use Comedian mail with my home system, but instead, just use an analog answering machine I already had around. Not ideal, I know, but the answering machine hangs up the FXS
What is the best way to get my CDR information into Oracle?
Is yada there for the best choice?
How stable is yada for this
Thanks Han
___
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Asterisk-Users mailing list
I'm proposing to install an Asterisk PBX at a collocation facility for a
remote customer. Each of the customer locations will have an SPA-3000
with the FXO port connecting a POTS circuit and the FXS port connecting
a fax machine or red phone.
In addition to voice traffic, the customer has a
hi all, anybody have a siemens hipath 3500 with a sm2/pri card? because
i need to connect to my box TE110P (e1) and i dont know how is the mode
in the pbx to change it.
thanks
--
.-
Pablo Allietti
LACNIC
___
--Bandwidth and Colocation sponsored by
Title: Message
The
SPA-3000 can do silence and tone detection for hangup and has a variable timer
and sound threshold setting.
-Original Message-From: BJ Weschke
[mailto:[EMAIL PROTECTED] Sent: Thursday, September 15, 2005 10:01
AMTo: Asterisk Users Mailing List - Non-Commercial
Hi,
Can someone point me
in the direction of getting the voicemail - Email to work on [EMAIL PROTECTED] 1.5
Ive put in the email
addresses of voicemail users eg [EMAIL PROTECTED] But i cant find where
to set the email server up. we have a company email server an idealy i would
like to
Asterisk crashes with no errors when I transfer from a device (my phone)to a queue Asterisk crashes with no errors. Also if I xfer from a sip device to another and dont wait for the other user to pickup before xfering the call gets dropped. Any ideas? Im using the latest cvs of asterisk, amp
Andrew Kohlsmith wrote:
Hmm, I guess I won't be buying any Mitel equipment. The MARS rovers were
designed to be totally shut down as a last measure to ensure everything is
starting up as they'd simulated on Earth and that there was no high-energy
radiation glitches due to space travel.
[EMAIL PROTECTED] uses Sendmail. The default configuration of
AAH's sendmail is not to relay the email to an external server, but to try local
delivery. You must modify the sendmail.mc file in /etc/mail with this
line:
define(`SMART_HOST',`mail-out.your.provider')
where
Best scenario does not route faxes over the IP network as a VoIP call.
You can either use spandsp as a fax on the Asterisk box, (has problems,
but the delveloper is behind solving them)
You can route the calls to a fax server located in the same colo via
tdm. (you can use HylaFax on Linix of any
Hello,
we have a small Asterisk Network where Siemens PBX's are connected via PRI
(Zap) to an Asterisk and
the Asterisk's are connected through IAX, so this looks like this:
Phone1 --- Siemens PBX --- Asterisk --- (IAX) --- Asterisk --- Siemens PBX ---
Phone2
Now, when Phone1 calls Phone2
Great comments everyone thanks and thanks for not flaming me.
Rebooting indicates that there is a problem that needs to be addressed. It
has nothing to do with uptime wars but with reliability and stability. I
don't care if the system's down for 3 minutes due to reboot, I am concerned
that
I have a TDM02B to bring in two POTS lines for my incoming calls; I
need to point each line to a different IVR... is there somewhere that
can I can look to get this setup working?
Basically, each line is for a different business. I know that for a
DID the routing is simple but I'm not seeing
7. RE: Stupid tricks: preventable? (Colin Anderson)
Colin Anderson [EMAIL PROTECTED] wrote:
i think you need a restart, then:
[your-local-extension-context]
exten = _,1,Gotoif([${CALLERIDNUM}=${EXTEN}]?2:4)
exten = _,2,Playback(you-are-a-frigging-idiot-stop-that)
exten =
On Thursday 15 September 2005 11:38, Paul wrote:
They designed it to be shut down. I guess that means it doesn't just
roll over like a dead cow.
Actually dead cows aren't back-heavy. They typically just keep whatever
position they were in when they took their last breath, much like the telco
I used the latest version (.3) and also
the previous .2 ver (pre20). The spandsp seems to compile but when I download
the rxfax/txfax .c files and drop them in the apps directory that is where I
get the compile error.
Dave
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
We use SpanDSP to recieve faxes on our PRI, a couple hundred a day with a
failure rate of ~5% which is pretty good I think but enough to tick people
off. Always the same fax numbers fail. What I did is have an exception list
context that is run just before RxFax. If the caller ID matches a bad
I have the same problem with several softphones (Xlite), but there's
one, Firefly I think, that worked. I found it strange, but not a real
problem for me. I have the same asterisk server version, wich came with
the last [EMAIL PROTECTED] distribution.On 9/15/05, Sherwood McGowan [EMAIL PROTECTED]
On 17:04, Thu 15 Sep 05, Han van Hulst wrote:
What is the best way to get my CDR information into Oracle?
Is yada there for the best choice?
I would use ODBC.
--
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key:
I would agree wholeheartedly with everything Colin just said. I've had
extensive experience with SpanDSP and with routing thru to an ATA.
Both are touchy and work OK, but not well enough to make the users
stop calling me :) Currently, we use SpanDSP and it works OK -
although sometimes pages get
Hi,
Last night I could dial *8 and pickup a call that was
ringing to another phone.
This morning, I searched on the Web for a solution to
mixed up digits when dialing on a Polycom Soundpoint 501.
I found that if you go to the SIP page on the phone's
Web interface and change the Digitmap
On Thu, Sep 15, 2005 at 10:04:15AM -0600, Colin Anderson wrote:
Great comments everyone thanks and thanks for not flaming me.
Rebooting indicates that there is a problem that needs to be addressed. It
has nothing to do with uptime wars but with reliability and stability. I
don't care
On Thu, Sep 15, 2005 at 04:48:04PM +0200, [EMAIL PROTECTED] wrote:
Show this message
adduser: Warning: The home dir you specified already exists
adduser: The user 'asterisk' already exist, and is non a system user.
dpkg: errrore processando asterisk-cah-capi (--configure):
il sottoprocesso
Here is what happens on startup:
Sep 15 13:23:51 DEBUG[28130] res_config_mysql.c: MySQL RealTime:
Static SQL: SELECT category, var_name, var_val, cat_metric FROM
settings WHERE filename='voicemail.conf' and commented=0 ORDER BY
filename, cat_metric desc, var_metric asc, category, var_name,
Sahil Gupta wrote:
Client (MERA) -- H323 -- Asterisk -- IAX -- Asterisk
You don't specify which H.323 channel driver you are using; there are
least four possibilities at this time, so that would be helpful information.
___
--Bandwidth and
Could you point to a specific issue? Any chance it could be backported
to 1.0? This will mean less griff.
http://www.google.ca/search?q=tdm+static+site:lists.digium.comhl=enlr=rls
=GGLD,GGLD:2004-23,GGLD:enstart=10sa=N
When do you reboot? every day?
Yes I do but the problem, for me, manifests
Hi
I'm looking for a company who can provide me an Internet connection
between africa and Europe.
Plesa If someone can give me some contact name or company dont
hesitate to send me a mail at [EMAIL PROTECTED]
Best regards
___
--Bandwidth and
FlagTel offer dedicated circuits between Egypt and Europe, if that
helps...
Regards,
Sahil Gupta
VoiceValley
On Thu, 15 Sep 2005, [ISO-8859-1] Stéphane LAVRI wrote:
Hi
I'm looking for a company who can provide me an Internet connection
between africa and Europe.
Plesa If someone can give
There may be a better way, but you can use the incoming call settings
in asterisk to point the first line at the first IVR. Then, if you
search the archives, you will find a post by me regarding custom
incoming routing and AMP, which describes how I did it. Be sure to
read the whole
Joshua Abbott wrote:
Has anyone every heard of Phonecall? : www.vecsector.com/phonecall/
Feedback?
Is there something as good as it or better ?
Recommendations?
I've heard of it! ;-)
Currently, the biggest trouble with it is the hardware configuration.
I'm working on a new Hardware
On Thu, Sep 15, 2005 at 12:04:41PM -0400, C. Hatton Humphrey wrote:
I have a TDM02B to bring in two POTS lines for my incoming calls; I
need to point each line to a different IVR... is there somewhere that
can I can look to get this setup working?
Basically, each line is for a different
Hi,
I have this scenario.
In my desk I have a phone connected to a FXS module of my * server. On another desk there is a phone but it is a SIP softphone (SJphone).
I hear the SIP softphone is ringing, then I try to take that call with my Zap phone in my desk dialing *8, but I get fast busy tone.
What is a good way to set up in the dialplan for the case where a
call fails (say due to congestion or whatever) and then asterisk
immediately dials again, with a different trunk or perhaps another
destination number?
Thanks
--
Eric Smith
___
Hello,
I have asterisk server with two isdn bri cards (billion) using zaphfc
driver. Also I have from telephone
company routed (for example) 16 pstn numbers. It is technically
possible to dial SIP phone from outside
before answering isdn pstn line.
I have local numbers 201,202,203 and from
did you edit extconfig and put a line similar to
voicemail.conf - realtime,mysql,database
then delete voicemail.conf from your asterisk configs directory and try
again.
matt
Matt wrote:
Here is what happens on startup:
Sep 15 13:23:51 DEBUG[28130] res_config_mysql.c: MySQL RealTime:
Stéphane LAVRI a écrit :
Hi
I'm looking for a company who can provide me an Internet connection
between africa and Europe.
'Africa' and 'Europe' are both rather big, so what you're saying doesn't
make much sense. Pehaps if you outlined your requirements a bit better,
you could get some
Hi all,
I have an EICON DIVA PCI 2.02 with I4L.
I'm unable to call some numbers, in general numbers with automatic
responders that do not rings.
It's seems asterisk does not understand that the other party has
answered, so after a timeout it reports 'busy', but in real the other
end has
Check out google with: VSAT Africa, lots of companies provide IP links
overthere. If it is good enough for voip... I don't yet know.
Stefan
On Thu, 15 Sep 2005, Jean-Michel Hiver wrote:
Stéphane LAVRI a écrit :
Hi
I'm looking for a company who can provide me an Internet connection
Many of the satellite companies block voip because they have the sevice for
sale them selfes. And dedicated satellite internet is VERY expensive. We
arranged a 512/512 connection today for a callcenter in Nigeria and they
will pay 6000 usd per month.
-Original Message-
From: [EMAIL
Andrew Kohlsmith wrote:
On Thursday 15 September 2005 11:38, Paul wrote:
They designed it to be shut down. I guess that means it doesn't just
roll over like a dead cow.
Actually dead cows aren't back-heavy. They typically just keep whatever
position they were in when they took
Massimo Frisoni ha scritto:
I have an EICON DIVA PCI 2.02 with I4L.
I'm unable to call some numbers, in general numbers with automatic
responders that do not rings.
It's seems asterisk does not understand that the other party has
answered, so after a timeout it reports 'busy', but in real the
Ive reduced my problem down to
this:
[EMAIL PROTECTED]:/usr/src/asterisk/asterisk-1.0.9/apps#
make
cc -D_GNU_SOURCE -o app_rxfax.so
app_rxfax.c -lspandsp -ltiff
app_rxfax.c: In function `rxfax_exec':
app_rxfax.c:263: warning: passing arg 1 of
`fax_init' from incompatible pointer type
Hi There,
Thanks for all your suggestions. I have now compiled asterisk from cvs
running on FD4. I have performed all the suggested configurations:
busydetect=yes ;changed 17.03.04 from no
busycount=7 ; added as above
for me the distro asterisk package didnt hang up properly on busy
Hi There,
I understand that asterisk can recieve faxes, but im wondering if anyone
has got it to work with telecom faxibility in NZ? e.g. get asterisk to
ignore the call if it rings with the faxibility ring?
Thanks
Simon
___
--Bandwidth and
Hi Jorg,
Ha ha - I tend to keep my off grid projects cheap simple. I noticed
that the gateways battery pack was 18V so I opened the battery pack up,
removed the batteries and connected a cable to the power connector
inside and ran it out through a hole I made in the back of the battery
Hello Ronan,
Thursday, September 15, 2005, 10:13:13 AM, you wrote:
Hi Sean,
This is what I've got in my zaptel zonedata.c file for a small * box in
Dublin:
{ 18, ie, Ireland, { 400, 200, 400, 2000 },
{
/* Dialtone = 400//425//450 */
{ ZT_TONE_DIALTONE, 425 },
Hi. I'm using /var/spool/asterisk/outgoing files to place automatic
calls, but I'm having trouble setting the Caller ID for the second half
of the call.
In other words, when we call the first number, we want the Caller ID
set to our number, but then when we connect them to the second number,
we
I am wanting to share the Console/dsp port between asterisk and mplayer.
I have alsa running on the box.
In module.conf I did noload for chan_oss. (actually I tried it both ways)
and if asterisk is running mplayer wont run.
Any thoughts on what I might be missing to get these two programs
to
Just reboot is a bad attitude.
If there is a memory leek, the fact that a reboot will free the leaked
memory is not a good reason to not fix the memory leek.
That kind of attitude is why windows does need regular reboots.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I did this for our website (to be released RSN), it has a contact form that
the customer plugs in their phone number. When they do, Asterisk calls them
and dumps them to an IVR. Pressing 1 in the IVR takes them to a salesperson.
My working config is:
In the .call file:
CallerID: 194 (our
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