On Thu, 2003-08-14 at 22:41, John Todd wrote:
> Since nobody else took the hint, I submitted it as a feature request for SIP.
>
> http://bugs.digium.com/bug_view_page.php?bug_id=104
>
> Personally, this is not high on my "I'd love to see this fixed" list.
> However, many others here are less
I must buy channel banks for ~120 lines. After some googling and ebay searching i see
that ADIT 600 has exelant proce//... for me.
Just wandering how it works with asterisk.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/m
> From: John Todd [mailto:[EMAIL PROTECTED]
>
> exten => 4001,1,Dial(SIP/gadams,10,r)
Hmm, the syntax below is what one of the Digium guys put in. Tried your
format, getting a ring followed by the unavailable message.
For now I have the 2 SIP phones in the ATL context to test phone to
phone capa
Helo WipeOut,
I have found a solution for sending dtmf after
dial.
I use spooling. Take a look at the sample.call file
inside asterisk dir. You need to edit this file and dump it in
/var/spool/asterisk/outgoing. Asterisk will precess this file
automaticlly
I create the sample.call do som
We use it, but with no caller id.
Tan
telappliant.com
- Original Message -
From: "Dave Wilson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 05, 2003 1:53 PM
Subject: [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?
Hi all,
I can't seem to find an
my error .. the cards are X100P which is why I wrote FXO.
The Grandstream phones are on a LAN, the * server connects to the phonelines
via the X100P cards. When I call from the Grandstream phones onto the PSTN
there is a VERY big amount of echo, ie. I can hear myself in the earpiece.
cheers
Dave
What if someone adds your number to that list ?
Someone would have to moderate it.
regards
Martin
On Tue, 5 Aug 2003, McAughan, Matt wrote:
> Does anyone keep a known telemarketer caller id database? If not has anyone
> proposed an Asterisk community project to share this information? Sort
I have attached the output. It is just one test call that goes to
voicemail. You can see the NOTICE message several times.
There is one thing interesting to note. If I start * from the console
"asterisk -cvvv" on the server I can repreduce it almost always. But if I
start it from a remote X-Te
They work great, I have 3 up and running all with mixed fxo-8 and fxs-8
cards.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Tinchev
Sent: Tuesday, August 05, 2003 4:56 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Someone used ADIT 600 Channe
Hello all,
I have a cisco AS5300 which is register with a
gatekeeper and a Asterisk server also register with the gatekeeper.
PSTN >AS5300 >Gatekeeper
>Asterisk
I set up a conference room on the Asterisk sever
(Room No 1234).
I try to call from PSTN to AS5300, The AS5300 wil
On Tue, 2003-08-05 at 20:00, Chris Hirsch wrote:
> Ok I'm convinced..one last question will a dual PII-266 500Meg RAM
> have enough horse power?
>
I don't know what everyone else thinks, but I've run a test unit on a
PIII 400 with 128M, and at the moment there's a 133 with 92M beside me
with just
0.2.4. just upgraded to 0.2.4a. This was supposed to be fixed in some pre
0.2.4 afacr.
Snapped from #asterisk on irc.oftc.net (see levon's answer below):
ERROR[243751]: File chan_capi.c, Line 1078 (capi_request): no free b
channel on controller 2! will continue searching.
ERROR[243751]: File
Hi,
I have just been playing with the latest X-Lite.. It works fine with Asterisk..
As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only one that didn't
work.. not sure why..
But the bigger problem is that when I call another extension that is using a Snom200
the call connects but
Does asterisk work with Vonage? I see all this talk or are you guys just
plugging it into an FXO port?
bkw
On Wed, 6 Aug 2003, Steve Meyers wrote:
> On Wed, 2003-08-06 at 13:39, John Schmerold wrote:
> > I've canceled my Vonage service because of the requirement to prefix
> > every call with a
I am glad I caught this... Is this a permanent solution or are there cvs
updates that will address this?
I am deploying a RH9 Asterisk server this Thursday I hope there are no
surprises.
I placed the export in my init.d/asterisk init script.
On Monday 04 August 2003 15:39, Scott Stingel w
Hi Lee,
You need to specify the VM context that you are using..
so using your examples..
extensions.conf entry..
exten => 1000,1,Dial(SIP/1000,20)
exten => 1000,2,Voicemail2(u1000)
exten => 1000,102,Voicemail2(b1000)
exten => 1000,103,Hangup
should be..
exten => 1000,1,Dial(SIP/1000,20)
exten
these are taken as db right? 3.0 db = 100%
but, in some cases we've had to do txgain=9.0
is that bad, martin? are there any hardware limitations on this?
does zaptel really accept %? if so, then it should be taken as a
percentage, not pseudoDB (tm)
- wasim
On Tue, 5 Aug 2003, Martin Pycko wro
Quoting Michael Manousos:
> Michael Ulitskiy wrote:
> > Michael,
> >
> > With all due respect to both of you, it's not related to h.323 driver.
> > The result is the same whether h.323 channel participates in the
> > call or it's pure sip-to-sip call.
>
> Did you try it without the ztdummy and zapr
At 2:29 PM -0700 8/6/03, Steve Haehnichen wrote:
-=> On Wed, 6 Aug 2003 14:18:00 -0700, John Todd <[EMAIL PROTECTED]> said:
There was nothing in the FAQ that really referenced anything about
using the phone with other systems other than the IPTel-powered
platform that they apparently are using.
Dave Alan Caruana wrote:
hi ..
I have an asterisk system with three TDM100P (single port FXO) cards
and 10 Grandstream 100 phones connected to it ..
1st question:
when i phone out
or receive a call from one of the SIP phones onto the PSTN, there is
a LOT of local echo in the handset .. the PSTN
-=> On Wed, 06 Aug 2003 13:38:58 -0700, George Pajari <[EMAIL PROTECTED]> said:
> My question for the list is who will be the first to report on the
> compatibility and usability of the SIPphone with Asterisk?
Well, I put in my order for the SipPhone 2-pack this morning. It
seems to be the chea
Not better A LOT BETTER...
;-)
Dan
- Original Message -
From: "William Flanagan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, August 06, 2003 8:21 PM
Subject: Re: [Asterisk-Users] Windows IAX soft phone
> Hey all,
>
> Why does the IAX client have better reliabilty t
Well,
I've made a startup script for Debian working from init.d, for redhat
there's a script in the source tree somewhere if I remember well...
I addedd this functions on the script: Start, stop, reload, force-reload
(restart now), and console to enter the *CLI>
If someone needs it, feel free to
I run with asterisk-oh323 0.5.4 from inaccessnetwork.
Thanks
Rattana
- Original Message -
From: "Jeremy McNamara" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, August 04, 2003 5:39 PM
Subject: Re: [Asterisk-Users] H323 CallerID
> Which H.323 channel driver are you running
Ok I'm convinced..one last question will a dual PII-266 500Meg RAM have
enough horse power?
McAughan, Matt wrote:
Chris:
Try not to be so worried about sound card,
analog (FXO/FXS), digital (ISDN, BRI, PRI) and what is available by
connecting device. The channel drivers take
unsubscribe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olga und
Andreas Brodowski
Sent: Sunday, July 27, 2003 11:19 PM
To: [EMAIL PROTECTED]
Subject: AW: [Asterisk-Users] * behind ISDN pbx - Forwarding to
extensions with in primary pbx
Hi Peer,
at my s
It is done. Let me know if there is anything I can do to expedite all
this. Thanks!
Jim Friedeck
--
TC wrote:
Jim
check and make sure logger.conf logs debug to the log file
start * with -vvvgnc
get it to seg fault, & find the name of the core.pid file like c
> *This message was transferred with a trial version of CommuniGate(tm) Pro*
> Dunno what I'm doing wrong here but I just did an upgrade to the latest
> version and now I get no audio at all!
> I havn't changed a single thing. Is there anything special I need to do
> to get this to work again?
>
I am having trouble with the AgenCallBackLogin app. I can't seem to
define a context for the queue.
Here is the relevant configs:
queues.conf:
[general]
[default]
[q_lo_1]
music = default
strategy = ringall
context = c_in_1
timeout = 15
retry = 2
maxlen = 0
member => Agent/@3
agents.conf:
[age
Hi Dan,
I use panasonic DECT phones, when plugged into a TDM20B (2 port FXS card from Digium)
I get caller id passed through (name AND number) although i can't get callerid via the
pstn at the moment (located in nl) i do get it for VoIP calls. Plus when a pstn call
comes in and there is no clid
> Quoting WipeOut:
> > Hi,
> >
> > I have just been playing with the latest X-Lite.. It works fine
> > with Asterisk..
> >
> > As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only
> > one that didn't work.. not sure why..
>
> Did you get Speex working? I've tried, but although I can g
Thanks for that,
I was looking at the extensions.conf, particularly the line in the general
section which is
TRUNK=SIP/???
Using this method would be easier.
How do you tell asterisk how many lines are available at the gateway
Dave
- Original Message -
From: "Martin Pycko" <[EMAI
Thanks Mark.
Any plans on implementing full redirect functionality?
Michael
On Thursday 07 August 2003 06:06 pm, Mark Spencer wrote:
> He should treat the first part as a local extension.
>
> amark
>
> On Thu, 7 Aug 2003, Michael Ulitskiy wrote:
>
> > Hi,
> >
> > Does anyone know if asterisk c
Hi,
Did you get something as caller id from the X100P card?
With the original code, in my situation the callerid information taken by
the X100P card was my own PSTN number.
With a very small change in the source code (commenting one line in
callerid.c source file) everithing now is ok.
This is tha
I agree...a php/forum based solution like for example
http://www.woltlab.info/en/produkte.php would be more effective and easier
to manage.
Ricardo Villa
http://www.telesip.net
- Original Message -
From: "James Taylor" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, August 09,
On Wed, 2003-08-06 at 13:39, John Schmerold wrote:
> I've canceled my Vonage service because of the requirement to prefix
> every call with a 1.
You could set up some simple extension rules in Asterisk that will
prefix 1 plus your area code on any numbers that don't start with a 1.
Just a though
We have clients coming in from all over the metropolitan area with
dial-up networking issues. The problem I have is we may set their
computer to dial 1 to reach their ISP, the client gets home & DUN
doesn't work because my guy forgot to remove the 1. Bad, very bad :-(
Just bought a channel ban
-=> On Thu, 07 Aug 2003 12:08:10 +1200, Richard Scobie <[EMAIL PROTECTED]> said:
> I have a ZyXEL Prestige 100 ISDN Router, a stand alone relic from
> when we used to access the Net via ISDN.
Like you said, this is a do-it-all router. I had an Ascend Pipeline
that was almost the same. As far as
On Fri, 8 Aug 2003, Richard Scobie wrote:
> I have been running busydetect=yes, using BUSYDETECT_MARTIN and am
> having hangups during calls.
If you use also BUSYDETECT_TONEONLY then you can detect tones that are
irregular, eg: 200 ms of tone, 200 ms of silence, 200 ms of tone, 500 ms
of silence
You might look into PAT to forward your RTP traffic to the asterisk
boxes.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut .
> Sent: Saturday, August 09, 2003 2:37 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] X-Lite - No sound
Make sure you are using G.711a, G.711u or GSM codecs.. I have not been able to get
iLBC to work and someone the other days couuld not get SPX working..
You will need to enable/disable the codecs in X-Lite..
If you also want to control the codecs that * uses then put the following in the
general
Try first to stop it :
asterisk -rx "stop now"
then killall -9 asterisk
On Fri, 8 Aug 2003, Jim Friedeck wrote:
> How do I stop asterisk when it is in a bad mood? It keeps dialing
> extensions and won't listen! I tried kill . No go. I don't want to
> have to reboot again. Thanks.
>
> Jim Friedec
On Fri, 8 Aug 2003, Maik Schmitt wrote:
[...]
> I just tried to use it with our 7960 (sip-version).
>
> I've set the services_url in SIPDefault.cnf to
> "http://xxx.xxx.xxx.xxx/xmlservices/vm/index.php&user=1234&pin=1234";
>
> It didn't work with ?user=...&pin= cause the phone then tried to
>
Hi,
How can I play Music On Hold on a channel for just a limited period of time.
The "Musiconhold" application plays indefinitely.
Thanks,
Dan
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
First of all I would like to thank Mark for getting roundrobin to go
roundrobin. Good job.
Now we have some options here for leastrecent and fewestcalls strategy. It
needs some work on the logic and Mark recommend that I ask the list and
get some input before he makes any changes to it.
fewestca
Jim
check and make sure logger.conf logs debug to the log file
start * with -vvvgnc
get it to seg fault, & find the name of the core.pid file like core.1234
cd /usr/src
asterisk -vvvgnc
gdb ./asterisk/asterisk ./core.1234
bt
paste the bt output a new bug rpt to bugs.digium.com
along with the la
Title: RE: [Asterisk-Users] queue / agent documentation
http://www.digium.com/asterisk_handbook/agentlogin_queues.html
And you should take a look at queues.conf for some comments detailing the various queue distribution algorithms, ringall, roundrobin, leastrecent so on so forth.
-Matt McAu
Eric-
Thanks for answering. Maybe a Stupid Question. If you have two IAX
softphones connecting to the same destination, how does it manage it on the
receiving end. It seems like that would work great for point to point
connections, but if you have more than one session up (and I would think for
Can anyone else that is trying to get Agents & Queues
into production shape report all issues to bugs.digium.com
& use the new Project ACD so that we can see all issue
related to this new code
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://list
Mark,
if the capability for line reversal detection is in the hardware (X100P) then does
this mean that the detection of DTMF style caller-id as used in the following
countries would ber trivial? or am I hoping too much...
Finland, Denmark, Iceland, the Netherlands,India, Belgium, Sweden, Brazi
Hello Michael,
Here is the BackTrace of the program which i forgot
to attach
BACKTRACE OF Asterisk -vvc
#0 0x42074d60 in _int_realloc () from
/lib/tls/libc.so.6
#1 0x420738c4 in realloc () from /lib/tls/libc.so.6
#2 0x47c7da89 in PAbstractArray::SetSize(int) () from
/home/sip/pwlib/lib/libpt
Sometimes you need to start asterisk from
/usr/src/
or /usr/src/asterisk/
and also in console mode for g729 code to work.
But I've heard that safe_asterisk also works.
regards
Martin
On 8 Aug 2003, Eric Wieling wrote:
> I'm getting the following message when I start Asterisk:
>
> WARNING[1024]:
Hi,
It seems that even the protocol is similar (for some European countries),
the frequency used for FSK modulation differ between US and Europe.
See the link for France:
http://matthieu.benoit.free.fr/freqcli.htm
They are:
Receiver FSKMin./Typ/.Max./Unit.
Transmission Rate118
On Fri, 8 Aug 2003, Dave Cotton wrote:
> The x100p does get the CID in France. It is now a question of how to break it down.
>
> I changed callerid.c line 278 to :-
>
> ast_log(LOG_NOTICE, "Got this:- %s\n", cid->rawdata);
>
> and the result on August 8 at 10:06 from 0490233081 was:-
>
> File call
Hi! I am new to Asterisk too, I got the similar problem and I would like to
know how to get * to work behind NAT.
When I have the SIP Debug turn on, I got the error 479 from FWD when * try
to register with FWD, it looks like * is using the local IP (192.168.x.x) in
the Contact field. I have put
Dear Colleagues
I am a newbie on Asterisk and
am having difficulties to find documentation about how to configure the H323
and Sip services. Could somebody of you have the kindness to send me functional
samples of conf files to my personal e-mail ?
I’m testing two VoIP clients: H.32
try this:
http://www.loligo.com/asterisk/current/
Lubo
Adelino Baena wrote:
Dear Colleagues
I am a newbie on Asterisk and am having difficulties to find
documentation about how to configure the H323 and Sip services. Could
somebody of you have the kindness to send me functional samples of co
Hi Bruce,
On Wed, 6 Aug 2003, Bruce Ferrell wrote:
[...]
> Is there a difference between what asterisk is and a softswitch? Can
> someone explain it in small words and phrases for me?
Sure, there is. First of all, * is much cheaper.
But technically, * does much more than a soft switch:
AIUI, a
Well the code that actually blows up is the rawwriteformat assignment,
but yes the channel pvt struct somehow is getting destroyed or corrupted
somehow
Jeremy McNamara
John Fortman wrote:
I don't understand the reasoning here so could somebody please help me
out?
chan_h323 is causing a se
Thank you. I am sorry. I've fixed the problem
Serge
From: Brian West <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Need help with installation of H323 chanel
driver
Date: Sat, 9 Aug 2003 20:19:54 -0500 (CDT)
Apparently you didn't read the RE
Julien,
try adding defaultip= in your sip.conf for each phone definition
Andy
*** REPLY SEPARATOR ***
On 10/08/2003 at 16:28 Julien wrote:
>Yes, the voice mail is at 2999 , but it doesnt work when i call it from
>the
>ata .I talked about the 600 (echo test) but i removed it fr
Quoting Paul Cheng:
> Hi,
>
> We have our * box configured to receive inbound SIP calls from FWD and
> enter into an autoattendant where someone can enter an extension
> directly.
>
> However, the inbound DTMF is not being correctly detected in most
> cases. Entering 8050 results in a correct detec
did you happen to run ztcfg after you setup the configs?
On Sat, 9 Aug 2003, Barry Porch wrote:
> I am attempting to set up an Asterisk box which I am only concerned with
> getting a single T1 working. I have this T1 connected to my PBX and I
> am looking at using Asterisk as a conference bridge
On Sun, 2003-08-10 at 10:18, Andrew Joakimsen wrote:
> On Sun, 2003-08-10 at 09:02, Andrew Joakimsen wrote:
> > Does anyone have Asterisk working with Iconnect here for incoming
> and/or
> > outgoing calls?
>
> have a look at:
>
> http://www.loligo.com/asterisk/example-configs.2003-04-24/extensi
Can someone find this?
On Sat, 9 Aug 2003, Dave Cotton wrote:
> On Sat, 2003-08-09 at 15:07, Siggi Langauf wrote:
>
> >
> > Well, I'd say: just strip the date and time off!. You made that call on
> > 2003-08-08, 10:06h local time, didn't you?
>
> My question was how to change the code.
>
> As an
>
>
> You shouldn't run the gatekeeper and asterisk on the same machine.
>
> Jeremy McNamara
>
>
Why is that?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi,
There is any Actiontec IPW owner on this list who can give me a
specification for the device drivers?
I mean just for the special functions like: ring, dial, tone, etc?
I intend to integrate it with the WinIAX client which for me works a lot
better than any other one.
Thanks,
Dan
__
Hi,
We have our * box configured to receive inbound SIP calls from FWD and
enter into an autoattendant where someone can enter an extension
directly.
However, the inbound DTMF is not being correctly detected in most
cases. Entering 8050 results in a correct detection, but enter 9003
results i
Hi WipeOut,
softdtmf=1 will use asterisk's dsp functions to detect and
generate the DTMF tones. softdtmf=0 will use your capi controller
to do the detection/generation. unless you have an active
card i would suggest to use softdtmf=1.
early B3 connects will let you hear the inband call progress
t
The chipset used in the X100P - at least the one I have - is designed for
the US/Japan market only. The reference design in the datasheet for the
chipset does not include facilities for the detection of line voltage
reversal. Hence the only way to detect caller ID sent before ringing would
be
>
> I always have a chuckle when I see this.
>
> it probably could if someone sorts it out, but its reqally starting to
> expect a lot.
It'll just take someone with the masochistic tendencies needed to do the
realtime DSP code for reception. For transmission, however, things are a
bit simpler. T
W:
checked the disk space and there is plenty of room
What sequence did you follow for debugging?
where does * put the E-mails before transmitting?
URiel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Simon
Woodhead
Sent: Saturday, August 09, 2003 4:02 PM
To
It works now ...sorry but it was my linux box ... I had Sip express router
installed on this machine :-\
So my ip phones loged on S.E.R and not on asterisk ;)
My voice mail works fine :)
Just a last question, if i configure G723 in my ATA, i can't call the
voicemail for exemple. I've seen that
All,
Has anyone had any thoughts/discussion on providing a malicious call trace feature
within Asterisk. Most legacy PBX's support this feature which allows a handset user to
indicate using DTMF during a call that it's a malicious call which instructs the PBX
to send a specific Q931 message ove
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armand A.
Verstappen
Sent: Sunday, August 10, 2003 5:29 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Iconnecthere
Hi Andrew,
On Sun, 2003-08-10 at 19:39, Andrew Joakimsen wrote:
> -Original M
what hassles?cvs update
Jeremy McNamara
Linus Surguy wrote:
Hi all,
We're using an older version of *, built a couple of months ago and before
we go through all the hassle of updating source files and checking latest
dependancies on other kernels etc, I'd like to know if the following is a
Check out X-Lite. http://www.xten.com
Only the older (pre 5.x) versions of Messenger support
SIP. If u can get a copy of v4.7 that will work fine.
Instructions on setup can be easily found by doing a
search of this mailing list. (look for the keywork 'MSN').
I do however suggest X-Lite, for
Hi All,
Is there any way to connect (register, initiate and receive calls) with
Asterisk to FWD through NAT? Since I own my router port forwarding is not a
problem.
I tried with
Register => @fwd.pulver.com:@fwd.pulver.com
but since Asterisk still use internal IP in some SIP fields I got "479 W
On Sun, 2003-08-10 at 21:25, Andy Hester wrote:
> Perhaps there is another way to cut down on increased traffic...
>
> Specifically, I would go back to the suggestion of a collaborative website
> for documentation. Collecting info and organizing into Howto's would reduce
> the number of times peo
So in other words... Asterisk can do what the Avaya Conversant can do if
you have a full understanding of it? Please pray tell how to do these
things. I am all ears.
Steve Lane
Vision Communications
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
At the console while a call is in progress run "sip show channels" and look in the
format column..
> How can I tell what codec a SIP session is using?
> --
> Paul
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailm
I'll admit it. I'm a asterisk newbie (but no stranger to telephony).
The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone
specials) on a private segment calling to a Linux box acting as the
segment's firewall with a leg on our public network. The phones are
setup as SIP/phone1 (x1
We do the replacement adapters for £12+VAT if interested. You'll still need
the US-to-UK adapter though. Contact me offline if you need one.
You could go to cpc as was suggested. www.cpc.co.uk
Tan
telappliant.com
- Original Message -
From: "WipeOut ." <[EMAIL PROTECTED]>
To: <[EMAIL PR
Does anyone have any info on weather or not GS plan to support some of the open
codecs, Mainly iLBC, GSM and Speex??
Having to use G.711 with Asterisk sucks.. its just too hungry on the bandwidth..
--
__
http://www.linuxmail.org/
Now with e-mail forwar
On analog ports you need to Answer
Ringing
Wait,2
and then do something .
That should detect faxes.
regards
Martin
On Mon, 11 Aug 2003, Tilghman Lesher wrote:
> On Monday 11 August 2003 03:26 pm, Eduardo Goncalves wrote:
> > On Mon, 11 Aug 2003 15:15:08 -0500
> >
> > Tilghman Lesher <[EMAIL
Yes, all does.
regards
Martin
On Mon, 11 Aug 2003, Michiel Betel wrote:
> Simple Q but I can't find the answer in the archives (and am too lazy to
> look in the source, but then its 32 Celcius here...
>
> Do all digium cards provide the zapata timing? e.g. also the analogs
> (including the X100P
they might be too lound
regards
Martin
On Tue, 12 Aug 2003, Lee Goodman wrote:
> Can anyone explain why this is happening?
>
> I have a server attached to a phone line that will play a .wav file, then play all
> the dtmf digits (after it answers the call). If I place a call from a SIP device
>
Hi Mark,
Short of taking my board out of my * box is there any way to check what revision of
the TDM400P I have? It was purchased in May of this year. Is the pricing likely to be
the same or similar to the add-on FXS ports? Does this also mean that as we'd be able
to get away with not using the
I have the same problem that Michael describes below does anyone have any recommendations? Jay __ Hi folks, I’m using chan_h323 to dial out to a gateway which connects me to the PSTN.In order to use a
Did you try:
register => @fwdnat.pulver.com:@fwd.pulver.com
On Tue, 2003-08-12 at 13:24, Borut Senicar wrote:
> Hi All,
>
> Is there any way to connect (register, initiate and receive calls) with
> Asterisk to FWD through NAT? Since I own my router port forwarding is not a
> problem.
>
> I tri
Try adding:
exten => fax,1,Dial(blah)
Where Blah is the zap or SIP port your fax machine is connected to.
On Mon, 2003-08-11 at 15:26, Eduardo Goncalves wrote:
> On Mon, 11 Aug 2003 15:15:08 -0500
> Tilghman Lesher <[EMAIL PROTECTED]> wrote:
>
> > On Monday 11 August 2003 02:46 pm, Eduardo Gonc
my point was your logic regarding 'calculating magic/metric' for
extremely long call times shouldn't be part of the 'logic' it
SHOULD be 'inhouse' policy where the mgr gives agentA a nice long
chat about how to sell/service the client more effectively.
i know there is at least one other out there
These list messages might be useful:
http://lists.digium.com/pipermail/asterisk-users/2003-June/013918.html
http://lists.digium.com/pipermail/asterisk-users/2003-June/013946.html
On Tue, 2003-08-12 at 13:22, Steve Lane wrote:
> I am trying to do the same thing you are doing. I am new to asterisk
Generally speaking, something like this would require a regulated power
supply,
as the internal components are most likely needing 5VDC, instead of the
phone
taking over 5V and regulating the power itself (which drops the voltage in
the
process, hence taking over 5V...). I'm guessing that you will
hi,
i have a external usrobotics modem, i want to use it with asterisk to
interact with the pstn,
what i have to do?
thanks,
--
santiago josé ruano rincón
administración servidores y servicios de internet
red de datos
universidad del cauca
-BEGIN PGP MESSAGE-
Version: GnuPG v1.0.6 (
Can anyone explain why this is
happening?
I have a server attached to a phone line that will
play a .wav file, then play all the dtmf digits (after it answers the call). If
I place a call from a SIP device (like a Cisco 7960 phone) through Asterisk and
on to the test server, via PSTN, the
On Tue, 2003-08-12 at 15:29, Steve Lane wrote:
> Would the firewall pose a problem? I thought Asterisk had the solution
> for working behind a firewall?
If is has it's one of the best kept secrets.
--
Dave Cotton <[EMAIL PROTECTED]>
___
Asterisk-Users
Title: ANI/DNIS call routing
Can someone in Asterisk'land subscribing to 800 service explain to me how to setup extension.conf to route calls based on the incoming DNIS/ANI. For example I want to route 3 incoming 800# coming across a trunk group to all land in the same queue. So I guess I am a
I've been taking another approach to this codec/bandwidth
problem. Instead of trying to get more codecs into
Asterisk (which is always hard due to licencing) I've been
trying to get vendors to implement GSM in their products.
SNOM do GSM.
D-Link gave me the good old, "we have plans to support
Hi,
Quick question to all the electronics gurus out there..
I unpacked my second GS phone yesterday (had it for about a month!) and set it up..
This morning the power supply is dead..
I have looked for a new one online (In the UK using Maplin let me know if you know a
better place.) becasue it
1 - 100 of 592 matches
Mail list logo
