Hello
I have a problem with bad CDR's after transfer of
call.
This is an example:
I've called from 616222820 to 616222821. Next I've
called from 616668020 to 060034 and then I've transfered the call.
I think, that I should received two CDR's
where:
in first CDR source=616222820 a
Hi
Thinking about starting a asterisk box, but I'm not sure what hardware to run.
I have a Dialogic D/300 board laying around, is it a OK choice?
Anyone using it? experiences? Hints?
Regards
Fredrik
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Robert Goodyear wrote:
Well if you say it's registered, then packets are getting to asterisk
and asterisk is accepting them, and you've allowed that SIP client.
So... if you say there's absolutely NOTHING happening when the phone
dials, then it sure seems like the phone is bad -- again, assu
Hi Guys
Just a stooped question, if I may. I have a bash script that
I would like to run from the acd. Would I use the System command, the API
command or is there a better way of doing this.
I want to create a smb connection to a microsnot machine and
pull a .wav file off to playb
Hi,
> -Original Message-
> I personally don't think it's a good idea to implement it in chan_sip.
> One reason for this is that user1 wants msn, user2 wants jabber, user3
> wants icq, user4 wants gadugadu etc etc. Are you gonna
> implement all this ?
>
> That is, if you mean Instant Mes
On Tuesday 05 July 2005 05:57, Tulika Pradhan wrote:
> call transfer works for me fine without any additions in features.conf
> by simply using Dial(SIP/${EXTEN},20,tT)
> and pressing #
> this works both from caller as well as callee.
>
> tulika
Could you provide me with some more information so I
On Mon, 2005-07-04 at 10:38 -0700, Mike Wissa wrote:
> When you try to start asterisk. the following errors
> appear
>
> Jul 4 10:37:59 NOTICE[4015]: res_odbc.c:518
> load_module: res_odbc loaded.
> .Jul 4 10:37:59 ERROR[4015]: chan_zap.c:6584
> mkintf: Signalling requested on channel 4
Hiii ; actually you are not allowing any codecs in the sip.conf neither alaw nor ulaw
so try this to all phones in sip.conf or put it in the general context (allow=all)
[2011]
type=friend
username=2011
secret=1945
nat=yes
host=dynamic
dtmfmode=rfc2833
canreinvite=no
qualify=200
allo
On Mon, 2005-07-04 at 23:17 -0700, Robert Goodyear wrote:
> On Jul 4, 2005, at 2:43 PM, Jimmy Smith wrote:
>
> > you guys are so friggin funny..
>
> We try. Meanwhile, you are SO illiterate; are you trying?
Very trying.
--
Dave Cotton <[EMAIL PROTECTED]>
_
Hello, friends:
I connected asterisk and Areskicc. but the sip channel can not be
established. the channel status is NO ANSWER.
I know somewhere is wrong in asterisk or Areksicc. any one knows how to
solve this problem, please tell me.
Thanks lot!
zhu
_
On Jul 4, 2005, at 2:43 PM, Jimmy Smith wrote:
you guys are so friggin funny..
We try. Meanwhile, you are SO illiterate; are you trying?
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Hi Rich,
Thank you for this response.
Strange... I have read something about this, but probably misunderstood :
http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html
"3. Start 'ztmonitor' on the target trunk in 'quantitative' mode.
4. Dial the CO Milliwatt test line from ins
Hello people
This is the site: http://www.wifi-cell.com/
Make a call and find out, I am interest it into hear opinions or Beta Testers
See ya
FachtopiaOn 7/1/05, Cory Andrews <[EMAIL PROTECTED]> wrote:
Robert - I suspect what they are doing is just trying to build "buzz"and simply not mentionin
Hi
I found the site, I will call tomorrow to find out prices
http://www.wifi-cell.com/
Good luck! to everybody!
fachtopiaOn 7/1/05, Richard Malcolm-Smith <[EMAIL PROTECTED]> wrote:
If it does materialize, im up for 3 or 4 of them at that price.Huddleston, Robert wrote:> Well poo - if I can use
http://www.wifi-cell.com/On 7/1/05, Huddleston, Robert <[EMAIL PROTECTED]
> wrote:Do you know where to get one of these?> -Original Message-
> From: [EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED]
] On Behalf Of> chawki hammoud> Sent: Thursday, June 30, 2005 4:35 PM> To: Asterisk-Users@lists.d
Well that sounds like a good solution to me if you do not use that codec.
Storm D. J. Petersen wrote:
Hi,
Today I decided to upgrade my * PBX and compiled the latest Development Head
and installed it. I keep getting this message:
WARNING[18268]: loader.c:313 __load_resource: libspeex.so.1: c
Hi, I would like to have a VoIP-card share a physical phone line with
some other analog device. When a call comes in it is supposed to be
answered by that analog device, only occasionally I'd like the VoIP-card
to "join" that conversation. Using the VoIP-card, I'd like to establish
a three-way
I have 5 Lines with BV and I can't route the incoming calls because of
this problem. So yes I think its broken.
David
On Mon, 2005-07-04 at 19:30 +0200, Christian Peter wrote:
> >
> > Yes, I know all of that, the problem is asterisk is NOT trying to
> > match anything except the IP from Broadvoi
I would be glad to have your servers in my Data Center in Tulsa, OK.
OCOSA Communications, LLC http://www.ocosa.com
Ariel Batista wrote:
Sahil Gupta wrote:
Hi,
Is there anybody on the list that recommends anyone for
colocation/telehousing in the US?
I'm after 2 Servers to be hosted in the U
call transfer works for me fine without any additions in features.conf
by simply using Dial(SIP/${EXTEN},20,tT)
and pressing #
this works both from caller as well as callee.
tulika
From: Frank Schoep <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
To: As
Sahil Gupta wrote:
Hi,
Is there anybody on the list that recommends anyone for
colocation/telehousing in the US?
I'm after 2 Servers to be hosted in the US, preferably on the west
coast.
I would suggest www.race.com
Regards,
Sahil Gupta
VoiceValley
Sahil Gupta wrote:
Hi,
Is there anybody on the list that recommends anyone for
colocation/telehousing in the US?
I'm after 2 Servers to be hosted in the US, preferably on the west coast.
Talk to [EMAIL PROTECTED], he can give you a server with a real TI
terminated to PSTN, and at excellent
I needed to learn Linux for a project about 18 months ago, so I went
down to a retail computer store and bought RedHat Linux. Installed with
no problems, and I was up and playing around with it in about an hour.
As I got more into it (and started breaking things) I needed some help.
The help I go
We do colocation, but we are on the east coast. What are your specific
needs? -Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sahil
Gupta
Sent: Monday, July 04, 2005 10:10 PM
To: asterisk-users@lists.digium.com
Cc: asterisk-biz@lists.digium.com
Subjec
Hi,
Is there anybody on the list that recommends anyone for
colocation/telehousing in the US?
I'm after 2 Servers to be hosted in the US, preferably on the west coast.
Regards,
Sahil Gupta
VoiceValley
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Hi All,
There are two option in QoS settings of the SIPURA ATA. ( I can't just
remember them). please tell me what is better and which one should I choose
for my DSL line (128kbps) with a small LAN.
thank you
kumara
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Asteri
Hi All,
There are two option in QoS settings of the SIPURA ATA. ( I can't just
remember them). please tell me what is better and which one should I choose
for my DSL line (128kbps) with a small LAN.
thank you
kumara
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Asteri
That stores the recordings in a MySQL database which in turn has a web
front end to look up the recordings etc?
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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Thanks Carlos this is the link I was looking for.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos
Alperin
Sent: Monday, July 04, 2005 5:28 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk, RH9,
6 beyond-the-network.LosAngeles.savvis.net (208.173.57.30) 33.966 ms
34.143 ms 33.841 ms
7 * * *
hangs there...
savvis invoice paid ?
beyond-the-network a black hole ?
On 7/4/05, Gary Reuter <[EMAIL PROTECTED]> wrote:
> Hmmm
> Can't place calls...
> Can't access website...
> Neither
Hm. I'm not quite sure how I should go about that - maybe I'm looking
at the wrong part of the documentation here. Would anyone be willing
to help?
~Andreas
On 7/4/05, Carlos Alperin <[EMAIL PROTECTED]> wrote:
> What about to define groups by usernames and assign the trunk to the group.
>
> Carl
> > Yes. More than 1 port as source and port forward doesn't work.
>
> Why isn't port-forwarding good enough? Maybe run a separate SIP proxy on
> another port?
I could be wrong here, but it sounds like the OP may not understand
the difference between source port and destination port, and he's
wan
Hi,
Today I decided to upgrade my * PBX and compiled the latest Development Head
and installed it. I keep getting this message:
WARNING[18268]: loader.c:313 __load_resource: libspeex.so.1: cannot open
shared object file: No such file or directory
Jul 5 01:26:32 WARNING[18268]: loader.c:523 load
Yes. That's what I'm trying to sort out with SER.
I just need to forward the packets. Anybody with a sample ser.cfg to do
that?
Isamar
On Tue, 5 Jul 2005, Tzafrir Cohen wrote:
> On Tue, Jul 05, 2005 at 08:13:58AM +0900, Isamar Maia wrote:
> >
> > Yes. More than 1 port as source and port forwar
That has nothing to do with the fact that you 're going to check what is
going on on that line.
You're going to check BER (Bit Error Rate) to see if you have line problems.
When I asked you for location, I was asking if your Asterisk box was in your
Computer Room, away from your TELCO provider. I
On Tue, Jul 05, 2005 at 08:13:58AM +0900, Isamar Maia wrote:
>
> Yes. More than 1 port as source and port forward doesn't work.
Why isn't port-forwarding good enough? Maybe run a separate SIP proxy on
another port?
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |
In astcc.agi there is are lines similar to this:
$dialstr = "IAX2/$res->{path}/$phone|30|HL(" . ($maxtime * 60 *
1000) . ":6:3)";
change the 30 to however many seconds you want.
Darren
wassim darwish wrote:
i dont know how to edit the time "3ms" for ringing
in astcc when i
or use an iax client..
many are out there
http://voip-info.org/tiki-index.php?page=VOIP+Phones
On 7/4/05, Joseph <[EMAIL PROTECTED]> wrote:
> On Mon, 2005-07-04 at 17:48 -0400, Jimmy Smith wrote:
> > another example of what i was saying..
> >
> > connect directly if still does this its not you
I think I need some help here. I didn't understand much of what you just said
there.
I though HDLC was a Layer 2 protocol? How can it have a "location"? FCS is a frame check sequence.. right? So I'm getting data out of sequence (or
frames are missing from the sequence?)
My gear is in a data c
Well, that looks correct. Might check zapata.conf and ensure there aren't
any other types, etc, in that. Also, might change both /etc/zaptel.conf and
zapata.conf to use fxs_ks just to be sure there isn't a bug lurking with
the loopstart stuff. Is there a specific reason you're trying to use
loopsta
I believe that you need to analyze the packets at your provider site. They
should be able to do that. Is your HDLC located on your location or on your
provider. This test should be done where Asterisk is running, because is
where the problem is reported.
Start to look for Line Analyzers for HDLC,
Someone please check the NASDAQ section on stocks to see if this people get
broke...
Something has to work, or the one that tried to use it has a bad Ethernet
port.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary Reuter
Sent: Monday, July 04, 2005 7:
> "ztmonitor 3 -v" start ztmonitor in graphical mode on Zaptel device #3.
> What is the correct syntax for dBm or voltage ?
There is no parameter to output voltage values.
Syntax:
[phoenix zaptel]# ./ztmonitor
Usage: ztmonitor [-v[v]] [-f FILE]
_
On Mon, Jul 04, 2005 at 05:51:25PM -0400, Jimmy Smith wrote:
> how about /etc/rc.local
>
> #a line that would work
> path/to/screen -d -m path/to/asterisk -vvgfc
>
The problem with adding stuff to rc.local is restarting. 'init 1; init
3' won't work as planned, for instance.
Also: people ten
> > Hi, I'm using asterisk with a digium TDM400P: I can't send/recive faxes
> > with a Philips fax machine.
> > It seems that the fax machine doesn't recognize the carrier.
>
> I'm afraid that your problem is not spandsp, but the TDM400P. Take a
> look at this (long) thread:
>
> http://lists.di
I never see that, the only solution that I can think about is that if you
want to run two applications at the same time, based on the info transmited
using port 5060 is to have a second machine as a Sniffer with the interface
in promiscuous mode listening port 5060.
Also, the sip device should be
On Monday 04 July 2005 18:34, Rusty Shackleford wrote:
> Wow. Where were you when LiveVOIP needed some good customer service
> people? You'd have fit right in with that outfit.
Dammit someone stole the joke already. There goes my reputation for
quick-wittedness. :-)
-A.
___
On Monday 04 July 2005 17:43, Jimmy Smith wrote:
> you guys are so friggin funny..
>
> all i see bout problems on most providers here are users who never
> read a line of the handbook
>
> i could prolly solve all these eyes closed with the asterisk handbook
> on my side as a friend.
[ ... ]
> my
Hmmm
Can't place calls...
Can't access website...
Neither of the 3 nameservers answer anything...
Anyone heard/know something to explain all this?
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That would work if for some reason ztcfg didn't refuse to run. I get the
proper output of ztcfg on the console but at runlevel it seems as if ztcfg
has never been run
On Mon, 4 Jul 2005, Jimmy Smith wrote:
how about /etc/rc.local
#a line that would work
path/to/screen -d -m path/to/asterisk
On Mon, 2005-07-04 at 17:48 -0400, Jimmy Smith wrote:
> another example of what i was saying..
>
> connect directly if still does this its not you my friend..
>
> people should really stop using asterisk as first connect attempts to
> test a service.
>
> use a direct client on provider .,
>
>
I have 2 servers, configured identically. Each has a TE405P and 2 PRIs. One server was experiencing crackly audio on one circuit, accompanied by HDLC
bad FCS messages. The telco recabled and moved me to another port on the DMS-100. The audio is better, but there are still bad FCS problems on the
Stefano Arata wrote:
> Hi, I'm using asterisk with a digium TDM400P: I can't send/recive faxes
> with a Philips fax machine.
> It seems that the fax machine doesn't recognize the carrier.
I'm afraid that your problem is not spandsp, but the TDM400P. Take a
look at this (long) thread:
http://lis
Yes. More than 1 port as source and port forward doesn't work.
Isamar
On Mon, 4 Jul 2005, Carlos Alperin wrote:
> Eric,
>
> This is the Sip.conf section where you define the port. Do you want to use
> more than one port as Source?
>
> ;
> ; SIP Configuration for Asterisk
> ;
> [general]
> port
What about to define groups by usernames and assign the trunk to the group.
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas
Turriff
Sent: Monday, July 04, 2005 6:51 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re
Eric,
This is the Sip.conf section where you define the port. Do you want to use
more than one port as Source?
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 69.39.69.183 ; Address to bind to
canreinvite =no
context = pstn
Hi all,
Couldn't find a place to search the list archives...
I'm having issues in getting any sound using a fresh asterisk install
and a SJPhone to connect to it. I went by the instructions pointed at
the "10 minute guide", located here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+quick
> Talking about this:
>
> I have at least 4 licenses on different Servers, and I need to look for each
> one of the e-mails that Digium sent me for each one ( if I can find them).
>
> I know that I have at least two different kinds of installation for those
> licenses.
>
> The question is: If I
Hello.
I am attempting to run an Asterisk server to centralize all Broadvoice
accounts for my employer. The intent is to have the Asterisk server
register with Broadvoice and route both incoming and outgoing calls to
the appropriate Broadvoice username. How do I restrict a given SIP
trunk to certa
Hi There
I am running SuSE Linux 9.3 and installed Asterisk from the RPM's. I
have a Diva ISDN card, making use of the HiSAX drivers.
Configuration seems to be OK. I can receive a call coming in on the ISDN
and route it to a SIP (softphone) on my PC. As soon as I hangup the
call, then asterisk s
On 7/5/05, Dana Olson <[EMAIL PROTECTED]> wrote:
> I think they were hoping that the client would connect to Asterisk,
> which makes it kinda useless, really.. But connecting Asterisk to the
> Gizmo network is handy.
Given that it's a fairly new program, we have to wait a while before
it's mature
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jimmy Smith
> Sent: Monday, July 04, 2005 2:44 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] VOIP Providers Problems
>
>
> you guys are so friggin funny..
>
> all i see bout
Carlos Alperin wrote:
Sipura boxes uses 5060 & 5061. I don't see why you cannot use something like
that. But anyway, that depends on how you 'll going to register them.
They use that for the SOURCE port of packets sent by the SIPura. It
still uses 5060 as the destination port.
--
Eric Wiel
Talking about this:
I have at least 4 licenses on different Servers, and I need to look for each
one of the e-mails that Digium sent me for each one ( if I can find them).
I know that I have at least two different kinds of installation for those
licenses.
The question is: If I upgrade my servers
Sipura boxes uses 5060 & 5061. I don't see why you cannot use something like
that. But anyway, that depends on how you 'll going to register them.
Carlos Alperin
Senior System Engineer
Seneca Communications, LLC
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL P
Bernie Ott wrote:
> http://www.auerswald.de/int/products/c4410usb.htm ) which I connected
> to my 2nd ZAP interface (s0 <-> Zap) via Crossoverr ISDN cable (which
> I crimped myself, I guess that's not the source of my trouble).
What hardware? What driver (zaphfc?) Have you got any messages in sy
Joseph wrote:
I'm testing Teliax tall free number line and I'm experiencing long delay
about 1sec. during conversation.
When I call myself over FWD the response is normal no delay or cut
messages.
When I call my number over FWD the is a long delay, welcome message
usually cuts off few first word
how about /etc/rc.local
#a line that would work
path/to/screen -d -m path/to/asterisk -vvgfc
-d -m Start screen in "detached" mode. This creates a new session but
doesn't attach to it. This is useful for system startup
scripts.
On 7/4/05, Carl
another example of what i was saying..
connect directly if still does this its not you my friend..
people should really stop using asterisk as first connect attempts to
test a service.
use a direct client on provider .,
Asterisk is complicated with many settings, unix flavors , hardware
and ba
I've not seen a troll like this for at least a week. :-) If you can
get them with your eyes closed then do us all a favour and help the
newbies! If you don't like that then go away
Darren Wiebe
[EMAIL PROTECTED]
Jimmy Smith wrote:
you guys are so friggin funny..
all i see bout proble
you guys are so friggin funny..
all i see bout problems on most providers here are users who never
read a line of the handbook
i could prolly solve all these eyes closed with the asterisk handbook
on my side as a friend.
wake up..
i work for a hosting provider and we get lots of users assuming
Carlos,
Sip is fine, I made it work in the past. Do you have your section sip on the
5300?
That is the key
Comprendistes?
Carlos Alperin
Senior System Engineer
Seneca Communications, LLC
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Did you check the log files looking for load errors?
Carlos Alperin
Senior System Engineer
Seneca Communications, LLC
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende
Sent: Monday, July 04, 2005 3:06 PM
To: Asterisk Users
I'm testing Teliax tall free number line and I'm experiencing long delay
about 1sec. during conversation.
When I call myself over FWD the response is normal no delay or cut
messages.
When I call my number over FWD the is a long delay, welcome message
usually cuts off few first words and during conv
Dan,
I believe that this can help you on your question.
http://www.automated.it/guidetoasterisk.htm#_Toc49248757
Regards,
Carlos Alperin
Senior System Engineer
Seneca Communications, LLC
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
I have a copy of RH9 and would like to build a Asterisk box for my
office but I do not want to load any unnecessary software.
Can someone provide me a list of required items above a "minimal
install". Thanks in advance.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECT
Juraj Bednar wrote:
> Hello,
>
>I was again asked to try to add support for presence
> (SUBSCRIBE/NOTIFY) and IM using SIMPLE. I have few questions:
>
>a.) are there any, at least partial projects, patches, anything,
> that at least partly implements presence and/or IM to chan_sip? I
> do
On 7/4/05, Dana Olson <[EMAIL PROTECTED]> wrote:
> I installed a vanilla 2.4.31 kernel from kernel.org and my system was
> working great.
>
> Then I tried upgrading zaptel to 1.0.9 and now I get unresolved symbols:
>
>
> # modprobe zaptel
> /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/
I installed a vanilla 2.4.31 kernel from kernel.org and my system was
working great.
Then I tried upgrading zaptel to 1.0.9 and now I get unresolved symbols:
# modprobe zaptel
/lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o:
unresolved symbol proc_mkdir_R8712438a
/lib/module
On Mon, Jul 04, 2005 at 09:55:46PM +0200, Roland Zagler wrote:
> i experienced that on some configs the "service asterisk restart" does
> not work correctly, so go to "/etc/rc.d/init.d" and edit the file
> "asterisk"
> and insert a "sleep 5" between stop and start in "restart".
Why is that?
'res
The weird ring back is a symptom of a problem with SIP message
syncronization (the equivilent of PSTN glare).
1) phone sends INVITE
2) * accepts the call, sends appropriate progress messages, then a 200 OK
3) the conversation proceeds
4) the caller and callee hangs up
5) * sends a BYE to the phone
Hi,
after you have done "make", "make install" and maybe "make samples" in
asterisk source-dir just do a "make config" and all will be done for
you.
to check if it worked, simply issue "chkconfig --list asterisk" to see
the
runlevels asterisk is started or not.
to start zaptel drivers do the sam
we may use it to store CDR records there in the
> future.
>
> I have decided to update the installation to 1.0.9. However, during "make",
> I receive:
>
> [...various errors...]
>
> I checked the version of FreeTDS, it is freetds v0.64.dev.20050704. (upon
>
I think they were hoping that the client would connect to Asterisk,
which makes it kinda useless, really.. But connecting Asterisk to the
Gizmo network is handy.
--
Dana
On 7/4/05, Adrian A <[EMAIL PROTECTED]> wrote:
> I have a Gizmo account working perfectly in my Xten Eyebeam, so there
> sho
hi ferdy,
again, thanks for all your help. I will try this and report back.
as for your questions:
1. my version is from stable, Asterisk CVS-v1-0-01/22/05-12:27:04
2. the line used that gets this database result is:
exten => 154,2,DBPut(DB(CFIM/${CALLERIDNUM})=${CALLERIDNUM})
which is, of cou
You start to not make any sense, you posted a question like this:
i try to do the following:
1) call comes in on extension 999
2) caller should hear music (NOT MoH!!!)
3) a call should be initiated to SIP Phone 100
4) when SIP Phone 100 is answered, a sound file should be played to the
user at SI
Dave Cotton wrote:
Using a vanilla 2.6.13-rc1 kernel zaptel compiles but then when
modprobing dmesg gives:-
zaptel: Unknown symbol class_simple_device_add
zaptel: Unknown symbol class_simple_destroy
zaptel: Unknown symbol class_simple_device_remove
zaptel: Unknown symbol class_simple_create
Th
Hi, I'm using asterisk with a digium TDM400P: I can't send/recive faxes
with a Philips fax machine.
It seems that the fax machine doesn't recognize the carrier.
How can I see the spandsp logs? I've enabled debug on the asterisk CLI,
but I can't see any output while the txfax/rxfax application run
Hi list!
I have several boxes running asterisk as I want, no problems there but the
one thing I haven't sorted out is how to properly start asterisk on boot
time.
This is probably a n00b class question but how do I properly set this up
(I didn't find any docs on this).
The zaptel script al
El lun, 04-07-2005 a las 09:53, Paul Goodyear escribió:
> Is the X100P FXO PCI Card capable of detecting a fax, answering the
> call, and then emailing the fax content to an email address?
For me work fine this card, the spanDSP and the
Follow these steps:
/etc/asterisk/zapata.conf
faxdetect=in
${DIALSTATUS} will tell you, also rtfm that will help you a lot.
The wiki is at: www.voip-info.org
Google is at: www.google.com
Browse this list: lists.digium.com
If you want to search the list with google, then type in
site:lists.digium.com when you enter your search terms on google.
On 7/4/05, D
Hi the list,
"ztmonitor 3 -v" start ztmonitor in graphical mode on Zaptel device #3.
What is the correct syntax for dBm or voltage ?
TIA
Best Regards,
Francois BERGERET,
France.
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when i dial an extension and the time on ring expiry how to know
if called party is bussy or not answer.
thanks-- David Romero##
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I have a Gizmo account working perfectly in my Xten Eyebeam, so there
should be no problem using it for Asterisk. You already have the
username (1747...etc) and your password, the proxy is
proxy01.sipphone.com (or you can sniff packets to see where SIP
messages are being sent to).
On 6/30/05, Rob
Hi all,
I need some help/guidance on writing the specs needed on a project
that will be scaling up to 10,000 users.
I will have some T1's to provide PSTN connectivity, and all the users
will be SIP and/or H323 phones. Services offered will include
conferences, voicemail (20 megs per users), etc
David Wilson wrote:
I have a weird thing happening sometimes with users calling from a
GrandStream phone through Asterisk onto a PSTN.
Sometimes after a user hangs up a call on a GrandStream phone the phone
starts ringing after a couple seconds.
When the call is answered there is no one there.
Hello Everyone,
This is my first post, and this is my problem :-).
I have a [EMAIL PROTECTED], work excellent (only internal users), but i need
outbound calls. One person give me an access to his "Cisco 5300 Media
Gateway", he give me a dial rule and the router ip address.
I've create
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I
try to dial an extension (from X-Lite), next time it is done.
X-Lite does not have a tone, nothing and does also have no time
out. It seems it is not connected to the server. Howeve
The /var/log/messages lists:
kernel: Module 0: Not installed
kernel: Module 1: Not installed
kernel: Module 2: Not installed Jul 3 22:21:10
kernel: Module 3: Installed -- AUTO FXO (FCC mode)
kernel: Found a Wildcard TDM: Wildcard TDM400P REV I
(4 modules)
the ztcfg -vv:
Zaptel Configuration
===
on spamdsp page found thisIt seems possible for the libtiff library to fall over when handling some
bad TIFF files. If spandsp is being used with Asterisk, this might bring the
entire PBX down. So far only one person has reported this. Recent security
update patches for libtiff 3.5.7, 3.6.0, and 3.
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