Duane wrote:
If they want a simple method
of allowing calls they should use enum, least then it's obvious that it
isn't a email address and that they would possibly need to enable a few
things to make it work.
Enum doesn't replace SRV records at all.
Enum records point to a SIP URI.
To resolve
I have sorted out the problem of compiling the CVS 040608. When launched
the server die with illegal instruction error.
Although the Makefile in asterisk is changed to PROC=i586. The Makefile
in codecs/ilbc still has a line of reference by using uname -m which
will come up with i686
unit, I hear the buzzing noise on the internal dialtone and also I hear
the buzzing noise on the internal network setup menu. (I even could
hear the buzzing dialtone when it was off the network)...
Try switching your computer and monitor off. Or your halogen lamp. Or your
cell-phone. Whatever
is it just me, or are the VoIP providers for Germany more expensive
than going via call-by-call?
sipgate.de lists a price of 1.76ct a minute, a couple of call-by-calls
are listed at 1.3ct-1.5ct a minute.
I see it exactly like you.
I'm probably NOT using Asterisk for outbound VOIP-Calls. The
Unfortunately I get an error in line 1205 when compiling chan_capi.c :
too many arguments to function: ast_dsp_process
Read the Readme or the Makefile.
Bang your head. :-)
Then enable the line that looks like CFLAGS += -DUNSTABLE_CVS
___
Actually there wouldn't be a need for an introductory course if the
Wiki wasn't so whacky.
Go and help and make the wiki better.
In the last days, several people made the wiki good, especially JazEzork.
It's a tremendous useful resource and anybody can make it better.
Does anybody is running spandsp with the latest cvs asterisk source. The
Makefile.patch that comes with spandsp doesnt work with cvs source code.
I would like to know how to path apps Makefile in order to compile
app_txfax and app_rxfax applications.
Pleae help
--
Manuel Marin Garcia
Has anyone managed to get Early-Dial working with the grandstream
phones?
It works for me on internal calls, not yet when I want to dial an external
call via chan_capi. Look for my bug notes in the wiki,
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+grandstream+budgetone
Dont use modprobe, use capiinit start
Doing that :-)
Get latest Firmware from:
ftp://ftp.in-berlin.de/pub/capi4linux/firmware/b1/3-11-03/
Done that now. I've also put this link into the wiki.
With some debugging statements I found out that it hangs right here
in chan_capi.c:
On Wed, 16 Jun 2004, Duane wrote:
Olle E. Johansson wrote:
[...H]owever sending mail to a host with only an
A record will still be delivered if the host is configured to accept
mail for the domain just like sip servers will accept the call if it's
configured to accept the call...
Yes it
Darren Edmundson wrote:
This is the correct way to do things. This is how asterisk should do them.
If you have a problem with the standard, the IETF SIP WG is the correct
place to air them, but I suggest the people there will be only too happy
to tell you why you're so very wrong too.
My argument
Maybe not. However, if the user is primarily interested in fax to email
then Hylafax can do that very well. A PBX is not an essential part of a
fax solution for many.
Iain
--On Tuesday, June 8, 2004 8:46 am +0800 Steve Underwood
[EMAIL PROTECTED] wrote:
Hi Iain,
Your response seems to
I try sniffing over asterisk but call must over asterisk and working goot
Sniffing network for voip call is very expensive and not working if call
is under ipsec packet
Any Czech firm working on this black box for voip sniffing and working
on h323 or sip
working on switche witn management and
I'm more interested in email to fax in as much as a user could send a
specifically formed email to a specific address and it be picked up and
faxed out. Similarly; inbound faxes being transformed into an email.
Matt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
My * shows me registered, but I can not place a call. Everything was
working fine, this has been a three day event.
I dial anything on their service, and i get an eventual timeout.
Did I miss something that went by perhaps? Something change? FWD and my
IAX2 inter-machine trunks are working
Anyone used this ?
I am having a bit of trouble got the right perms on makering.pl .
Should that file be somewhere in particular ?
use the reccommended command
sox inputfile -r 8000 -c 1 -t ul - rate | makering.pl ring1.bin
but i get
bash: makering.pl: command not found
Can ya help ?
In article [EMAIL PROTECTED],
Simon [EMAIL PROTECTED] wrote:
Anyone used this ?
I am having a bit of trouble got the right perms on makering.pl .
Should that file be somewhere in particular ?
use the reccommended command
sox inputfile -r 8000 -c 1 -t ul - rate | makering.pl ring1.bin
Aaron Martin [EMAIL PROTECTED] wrote:
Has anyone managed to get Early-Dial working with the grandstream
phones?
Yes, but it doesn't play nicely once calls are being gated to the
PSTN.
Early Dial works by attempting a call for each digit that is dialled.
Asterisk will try each such call across
Also check the first line of the file points to your location of perl.
Usually /usr/bin/perl or /usr/local/bin/perl
PS, chmod 755 makering.pl to make it executable.
Regards,
Adam
On Tue, 2004-06-08 at 18:52, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Simon [EMAIL PROTECTED] wrote:
Tony
That was a quick reply ( it came in before the copy of my message )
This is what i just did
sox file.wav -r 8000 -c 1 -t ul - rate | ./makering.pl ring1.bin
Then got
: bad interpreter: No such file or directory
Ta
Simon
-Original Message-
From: [EMAIL PROTECTED]
Hi!!
I try to install meetme2i follow instructions that i found in
http://www.areski.net/asterisk-meetme/about.php?s=0
but, when i modify the "Asterisk/apps/Makefile" and i run the "make"
command,
I have this type of error:
[EMAIL PROTECTED] apps]# makecc -pipe -fPIC
-DUSEMYSQLVM -c
.. Hylafax does that too.
Iain
--On Tuesday, June 8, 2004 9:15 am +0100 Matt [EMAIL PROTECTED] wrote:
I'm more interested in email to fax in as much as a user could send a
specifically formed email to a specific address and it be picked up and
faxed out. Similarly; inbound faxes being
On Tue, 8 Jun 2004, Duane wrote:
Darren Edmundson wrote:
My argument isn't about the standards or other software in general, my
argument is how asterisk (and in this case only asterisk) comes, that is
with SRV *disabled*, and the fact many people wouldn't understand what
it's for, or why they
# Numbers starting 0 are PSTN calls
exten = _0.,1,Macro(dial-pstn,${EXTEN})
Now suppose I want to call 01234 567890. Asterisk will return 484 for
0. However, when the 1 is dialled, the extension matches, and a
call will immediately be attempted to 01 (and fail), without me
having had
: bad interpreter: No such file or directory
Install perl. Then call the program like this:
perl makering.pl
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
Adam
We had done 755 and the path is correct.
still get error's
: bad interpreter: No such file or directory
Ta
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adam
Goryachev
Sent: 08 June 2004 09:59
To: [EMAIL PROTECTED]
Subject: Re:
This is called overlapdial in zaptel. It works on all zaptel cards i've
tested so far, also the zapbri cards. Chan_capi supports it aswell...
(called Early B3 iirc), and with the iaxy it is no problem either... (it
starts a call when picking up the hook)
So, if there is a problem... it is
Thanks to all for the help
The answered turned out to be some odd chr in the file that have for some
strange reason only just showed , they were not there when i first uploaded
makering.pl
Ta
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Holger
This is called overlapdial in zaptel. It works on all zaptel cards i've
tested so far, also the zapbri cards.
Good to hear this. For me, chan_capi + P2P is a dead end, so I'm waiting
till I get my zaphfc-supported card, probably on Thursday.
By the way, you don't HAVE to press send... if
app_meetme2.c:31:22: libpq-fe.h: No such file or directory
app_meetme2.c:32:19: mysql.h: No such file or directory
You need the mysql-devel packages for your distro.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I heard that asterisk support r2 signaling.
I'm try to test r2 using e100p.
How should I configure zaptel.conf, zapata.conf?
And if I want to modify source for customization, where should
I start?
Thanks.
The thing is I plugged the device in another room and I still could hear
the noise. I think its possible it's just a faulty device.. Why the company
I purchased it off of is sending me another unit.
Holger Schurig wrote:
unit, I hear the buzzing noise on the internal dialtone and also I hear
the
Tony
Ok got it to work now , thanks.
I have put ring1.bin in the tftpboot folder on my tftp server and rebooted
the phone but it has not taken the file.
Do i need to do anything special with the phone etc ?
Thanks
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
In article [EMAIL PROTECTED],
Holger Schurig [EMAIL PROTECTED] wrote:
app_meetme2.c:31:22: libpq-fe.h: No such file or directory
app_meetme2.c:32:19: mysql.h: No such file or directory
You need the mysql-devel packages for your distro.
And postgresql-devel, unless you comment out the
In article [EMAIL PROTECTED],
Simon [EMAIL PROTECTED] wrote:
Tony
Ok got it to work now , thanks.
I have put ring1.bin in the tftpboot folder on my tftp server and rebooted
the phone but it has not taken the file.
Do i need to do anything special with the phone etc ?
Not to make it
I've been playing with two pieces of hardware: a X100P and a TDM400P with an
FXO and two FXS modules. I had been using just the TDM
card; however, the TDM FXO module seems to hear things and answer the
telephone for no reason, and I wanted to compare the results
with an X100P card.
Thanks I will like in to them as for the switch problem I have tons of cisco
gig for that problem and my voip is running at 100mb so spanning to a gig
port solves it
Doug Block
Chief Information Officer of Efast Funding
713-983-4055 (Direct)
888-338-3863 x 4055 (Toll Free)
713-983-4555 (Direct
Bisker, Scott (7805) wrote:
If there is already an existing phone system in place, you could easily migrate to an
asterisk based solution if your internal phones are analog. The big question for you
is not number of phone lines, but peak utilization. Here's what I have.
Max concurrent calls
Tony
Have searched for the new fimware , can't find it no links on grandstream .
I am using
Software Version:Program--1.0.4.30Bootloader--1.0.0.13
HTML--1.0.0.20
No option to set ring tone in there.
Sorry for being a plank but i am just learning this stuff
Simon
-Original
dkwok wrote:
I have sorted out the problem of compiling the CVS 040608. When launched
the server die with illegal instruction error.
Although the Makefile in asterisk is changed to PROC=i586. The Makefile
in codecs/ilbc still has a line of reference by using uname -m which
will come up with
Matthew,
Dial works on a fall thru principle. Thus:
exten = 555,1,Dial(SIP/1000,30)
exten = 555,2,Dial(SIP/2000,30)
should suit your purpose (not taking into account vm), to add another exten just add
it on the dial 'list':
exten = 555,1,Dial(SIP/1000,30)
exten = 555,2,Dial(SIP/2000,30)
exten
Hi!
is it just me, or are the VoIP providers for Germany more expensive than
going via call-by-call?
What about other countries? Same thing?!
The thing is that only in Germany Call-By-Call providers have *that* low
prices. Anywhere else in Europe - at least in the small and middle-sized
Hi!
Did you try out the new ring tones? One of them contains a regular ring,
followed by a voice announcing the caller id of the calling party. VERY
neat.
Hehe - you're right! And is my impression correct that Asterisk users
know that particular voice very well that is announcing You have
Pah! my fingers are getting in the way today:
exten = 555,1,Dial(SIP/1000,30)
exten = 555,2,VoiceMail2(u3278)
exten = 555,3,Hangup
exten = 555,102,Dial(SIP/2000,30)
exten = 555,103,VoiceMail2(u3278)
exten = 555,104,Hangup
exten = 555,203,VoiceMail2(u3278)
exten = 555,204,Hangup
Andy
Hello
I have the follow situatuion:
ISDN
|
|
V
E100P
|| IAX2 / g729A || T100P
| Asterisk1 |- - - - - - - - - - - - - - | Asterisk2 | - - - - -
- |--|
| | | | | Zhone|
- -
Hi,
I'm trying to get a fritz working in our asterisk box, however I'm
getting the following error and was wondering if anyone could be of any
help:
-- Executing Macro(SIP/2002-5501, enum-call|call|0405152568) in
new stack
-- Executing Goto(SIP/2002-5501, call|0405152568|1) in new
In article [EMAIL PROTECTED],
Simon [EMAIL PROTECTED] wrote:
Tony
Have searched for the new fimware , can't find it no links on grandstream .
I am using
Software Version:Program--1.0.4.30Bootloader--1.0.0.13
HTML--1.0.0.20
No option to set ring tone in there.
Sorry for being a
Reid A. Forrest wrote:
Ummm X100P IS an analog adaptor.
Well, I thought he was talking about some of the ATA´s available on the
market. If an X100P is one analog adapter and has this problem, then
everything else has this problem too (except E1/T1/PRI boards that have
it´s own signaling).
Hi Philipp,
I'm not receiving my emails sent to the list. I thought my email was
not in the asterisk-users list anymore. But anyways..
About the DIAL application, I'm currently use the DIAL application from
an AGI program to connect call to Zap channels (i.e, the caller calls
an DID number, I
On 8 Jun 2004, at 15:02, Philipp von Klitzing wrote:
The thing is that only in Germany Call-By-Call providers have *that*
low
prices. Anywhere else in Europe - at least in the small and
middle-sized
countries - you'll surely give preference to VoIP due to better
pricing.
That's very clearly the
If you wan't to create a ringtone with makering.pl for firmware 1.0.50,
be sure to create it as ring.bin and then rename it to ring1.bin /
ring2.bin or ring3.bin. This seems to be the only change between the
format from 1.0.4.68.
Regards,
Maron
___
On Tue, 8 Jun 2004, Simon wrote:
We had done 755 and the path is correct.
still get error's
: bad interpreter: No such file or directory
That is usually due to DOS linefeeds. Try to open it in vi and the run
:set ff=unix
:wq
and try again
-Original Message-
From: [EMAIL PROTECTED]
Hi * :-)
I found in the online WiKi docs some information on how to integrate
Asterisk with old PBX...
http://www.voip-info.org/wiki-Asterisk+legacy+integration
...but I couldn't find anything on integration with a Siemens HiCom
150E. Later on we'll migrate to a HiPath 3750 so information
Yes I've been told how much you love enum
a few hints
o i was one of the gang who came up with the idea of stealing
the enum hack from tcp.int (may have been the first, but
the brain, such as it is, fades)
o i helped the sippers work out the enum naptr etc hacks, in
fact pushed
Aaron Martin wrote:
Has anyone managed to get Early-Dial working with the grandstream phones?
On my older phones running firmware 1.0.3.X it works fine, but it doesnt
work on the newer versions..
Don't waste time trying. I'm even surprised that you could get it to
work with 1.0.3.x. In my
Simon Dorfman wrote:
I'd love to hear a review of any Snom Phones. I'm waiting for the Snom 190
before I buy my first hardware VoIP phone. It's supposed to be around $150
or less.
I've already read what voip-info has to say:
http://voip-info.org/tiki-index.php?page=Snom%20Phones
Hey Vasyl,
Yup, this is exactly what I have... :) the hdlc stuff works great... It is the routing of the network that is driving me nuts since the set up that Sprint gave me was:
209.26.25074 pointopoint 209.26.250.73,
adtran router 209.26.224.33,
usable IP's as you said, 209.26.224.34/29,
pesb [EMAIL PROTECTED] wrote:
I know this question is kind of stupid. But, I don't know
anywhere else to ask. I've received some answers when I asked about the
need of having a zaptel interface to make the meetme application work,
that said that it was better to have a real hardware then the
I recommend a PRI E1 link into the Hicom from *
Jason
At 16:05 08/06/2004 +0200, you wrote:
Hi * :-)
I found in the online WiKi docs some information on how to integrate
Asterisk with old PBX...
http://www.voip-info.org/wiki-Asterisk+legacy+integration
...but I couldn't find anything on
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tor Houghton wrote:
| On Sun, Jun 06, 2004 at 04:25:32PM -0400, Tim Sailer wrote:
|
|On Tue, Jun 01, 2004 at 07:49:29PM -0500, Yelson Vivas wrote:
|
|Hi everybody
|
|i have a problem trying to connect an incomming phone call from pstn
to my
|(soft
If you're in the current directory with makering add a ./ in front of it.
Otherwise you can try moving it to /usr/bin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon
Sent: Tuesday, June 08, 2004 4:40 AM
To: Asterisk-Users
Subject: [Asterisk-Users]
Is there a clever way to camp on an extension in asterisk? What I need is a
way to answer my extension (not just a ringing ZAP channel) from any other
phone. If I'm in another office and hear my phone ringing, I want to be
able to quickly pick it up from that extension. The list revealed the
Jason A. Pattie wrote:
|
| One workaround is to use Firefly, but that may not be for everyone?
True. I almost got it working under Wine, though. Kept dumping files
into C:\. Probably just means I don't have the necessary dependencies
or Wine doesn't have the capabilities needed to run this
All,
I am setting up my first * box and am trying to configure SIP Softphone to SIP
softphone dialing.
When I dial ext. 2001 it rings once (Very short) then immediately goes to voice mail
for ext. 2001.
I keep seeing the Got SIP response 482 Loop Detected back from 192.168.1.252,
immediately
exten = 555,1,Dial(SIP/1000,30)
exten = 555,102,Dial(SIP/2000,30)
exten = 555,103,Dial(SIP/3000,30)
exten = 555,104,Voicemail2(u3278)
exten = 555,105,Hangup
exten = 555,2,VoiceMail2(u3278)
exten = 555,3,Hangup
...should be
exten = 555,1,Dial(SIP/1000,30) ; Unanswered = 2, Busy = 102
Hi there,
I have the following scenario:
I want user A to make a call to user B. If, any of these users transfer the
call to user C, then asterisk should generate a CDR for the first leg of the
call. And after the transfered call is finished, asterisk should generate a
second
Does anyone know if the 1-800 iaxtel gateway is down?
I've been trying to use it all day today and asterisk says it's ringing:
Channel (ContextExtensionPri ) State Appl.
Data
IAX2[iaxtel]/1 ( s1 ) Ringing AppDial
(Outgoing Line)
SIP/2201-a253 (home
On 08/06/2004 at 11:15 John Fraizer wrote:
exten = 555,1,Dial(SIP/1000,30)
exten = 555,102,Dial(SIP/2000,30)
exten = 555,103,Dial(SIP/3000,30)
exten = 555,104,Voicemail2(u3278)
exten = 555,105,Hangup
exten = 555,2,VoiceMail2(u3278)
exten = 555,3,Hangup
...should be
That's why
Hi all,
A new version of DIAX (0.9.8a) is ready to be downloaded from the following
locations:
http://www.laser.com/dante
or
http://www.geocities.com/tdanro
What's new in 0.9.8a:
- unconditional autoanswer or based on CallerID (user configurable);
- use any Ericsson/SonyEricsson GSM/PCS to
Hi All,
I'm running the latest asterisk CVS code (from experimental), and
hardware-wise an AVM Fritz!PCI Card, together with chan_capi 0.3.3.
The problem is that any inbound/outbound calls result in echo on MY end
(the asterisk end). I've played with the echo settings in capi.conf
(mainly
pesb wrote:
Hi there,
I have the following scenario:
I want user A to make a call to user B. If, any of these users
transfer the
call to user C, then asterisk should generate a CDR for the first leg
of the
call. And after the transfered call is finished, asterisk should
Do you have r on your Dial line? If so, then Asterisk will override
whatever should you SHOULD be hearing and provide you with a ringing
sound.
On Tue, 2004-06-08 at 10:24, Mark Musone wrote:
Does anyone know if the 1-800 iaxtel gateway is down?
I've been trying to use it all day today and
I have the same situation - i.e. three different extensions scattered
about. But I don't try them each individually. When a call comes in my
asterisk attempts to ring up to four different devices at the same time.
To do this using your dial plan is easy - i.e.
exten =
Duane wrote:
The documentation, well what documentation there is, simply isn't
coherent enough, or detailed enough to explain these things, and the few
lines in the config file certainly doesn't explain anything either...
;srvlookup=yes ; Enable DNS SRV lookups on outbound calls
Does anyone know if the 1-800 iaxtel gateway is down?
I've been trying to use it all day today and asterisk says it's ringing:
Channel (ContextExtensionPri ) State Appl.
Data
IAX2[iaxtel]/1 ( s1 ) Ringing AppDial
(Outgoing Line)
SIP/2201-a253
It seems to be down, I even tried dialing for
example 1-800-555-TELL. I tried yesterday
and again today.. Just get dead air.
Stephen Rosebush
Mark Musone wrote:
Does anyone know if the 1-800 iaxtel gateway is down?
I've been trying to use it all day today and asterisk says it's ringing:
Down here.
It seems to be down, I even tried dialing for
example 1-800-555-TELL. I tried yesterday
and again today.. Just get dead air.
Stephen Rosebush
Mark Musone wrote:
Does anyone know if the 1-800 iaxtel gateway is down?
I've been trying to use it all day today and asterisk
how can you create your own ring tone?
- Original Message -
From: Maron Kristófersson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 08, 2004 6:57 AM
Subject: [Asterisk-Users] grandstream ringtones - makering.pl usage for
1.0.50
If you wan't to create a ringtone with
I do, see sip.conf below. My network is multi-NAT, so all DHCP'd
machines have one external IP to share, and a small number of servers
(including *) get a static internal IP which maps to a static external
IP.
[general]
port = 5060 ; Port to bind to
bindaddr = 192.168.254.204
Hi,
I have a problem to make call behin NAT with bewan router, the phone in the
internet can call the phone behind bewan router but the phone behind the
bewan can't make a call.
Any Idea are welcome.
Best Regards.
Zouhair.
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus
In sip.conf, make sure you set nat=yes and canreinvite=no . I had similar
problems, and those two settings made a big difference.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 8 Jun 2004, Echchelh Zouhair wrote:
Hi,
I have a problem to make call behin
How do I patch an * file if all I have is the .diff
file?
Achilles Bochoris wrote:
Hello,
I am relatively new to Asterisk and I need to compile the G.723.1 codec
for Asterisk. I downloaded the ITU source code, placed it in the codecs
directory, but apparently Asterisk needs a rather different library than
the one provided from ITU.
As I've seen in
Type:
user$ man patch
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Devine
Sent: Tuesday, June 08, 2004 11:28 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] HOW-TO DIFF
How do I patch an * file if all I have is the .diff file?
Cd ../asterisk
patch your_diff_filename.diff
Cheers,
Rich
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Ed Devine
Sent: Tuesday, June 08, 2004 12:28
PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] HOW-TO
DIFF
How do I patch an * file if all I
Hi,
Having searched through the mailing list archives and the wiki, I still
don't know how to solve the following problem:
Call is received, phone rings once, then the caller gets the voice menu.
What I want is for the call not to actually ring, but to go straight to
the voice menu.
How can I
Hi,
I had the same problem until I changed iax.conf to not have a callerid=
field in it for the context you are using.
All I have now is.
[guest]
type=user
context=default
I have several servers all talk to each other, and get caller/extension ID
from them all.
Dave
-Original
same with their 700 network
w
- Original Message -
From: Mark Musone [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 08, 2004 11:24 AM
Subject: [Asterisk-Users] iaxtel 1-800 gateway down?
Does anyone know if the 1-800 iaxtel gateway is down?
I've been trying to use it all
On 08-06 18:06, Duane wrote:
Darren Edmundson wrote:
My argument isn't about the standards or other software in general, my
argument is how asterisk (and in this case only asterisk) comes, that is
with SRV *disabled*, and the fact many people wouldn't understand what
it's for, or why
Actually, I think the format of the file has changed in version
1.0.50.
I did some sniffing, and came up with an approach that worked for me.
I've attached the modified version of makering.pl that I've
used (thanks to Tony and Stephen), as it may work better for others.
It also includes a
[incoming]
s,1,Answer
s,2,Wait,1
s,3,VoiceMailMain(u1234)
If the caller still hears a ring, it's probably due to the incoming
channel being configured for caller-id. CID info is passed between ring
1 and ring 2.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I have to correct myself here -- this is no longer working. I know I
had it working from my cell-phone at first. What's interesting is that
I can HEAR the DTMF go through, both from cell-phone and other sip
lines. I changed it back to inband, but no luck.
-Original Message-
From:
On Tue, 8 Jun 2004, John Campbell wrote:
Hi,
Having searched through the mailing list archives and the wiki, I still
don't know how to solve the following problem:
Call is received, phone rings once, then the caller gets the voice menu.
What I want is for the call not to actually ring,
On Tue, Jun 08, 2004 at 01:52:44PM -0400, [EMAIL PROTECTED] wrote:
You are using analog lines? If so, asterisk has no way of knowing the
phone is ringing, until it rings.
Even if the called end is digital (ISDN or PRI), isn't it possible that
the phone system on the calling end could indicate
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi!.
I am having problems with getting asterisk to detect when someone hangs up.
I have a TDM400P with one FXO module connected to my telco, and also a
FXS-module connected to my phone.
The FXS-module detects hangups just fine, but I can't get the
Hello,
I understand these licensing issues very well. I
don't reside in the US, so I assume that there is no problem, especially for
testing/development, and not commercial use.
What I was asking however, is whether there is an
alternative G.723.1 library which compiles with asterisk other
Title: Call centers using Asterisk
I have a local company that wants to use Asterisk for a small call center. I've told them that they can do so but they'd like to actually talk to somebody that is. Does anybody know anyone using Asterisk in a call center that they could talk to?
If you're
Jeremy White wrote:
Actually, I think the format of the file has changed in version
1.0.50.
I have gotten a bit more clarity on the ringtone issue from Grandstream
support.
It is a fact that there are two hardware revisions of the phone, which
they call Rev A and Rev B.
Only Rev B phones are
I have been lookin for someplace to lean how to dump all of the call
transactions into a sql database. Can anyone provide me any assistance?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi,
I have a current call into BroadVoice regarding lack of DTMF
on incoming calls. They are aware of the situation but have
yet to respond.
Michael Swan
Neon Software, Inc.
At 12:48 PM 6/8/2004 -0500, you wrote:
I have to correct myself here -- this is no longer working. I know I
had it working
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