i am using sipsak to test asterisk. i use the command
$ sipsak -U -I -s sip:[EMAIL PROTECTED] -vv -x 1800 -e3
and i get the message
SIP/2.0 407 Proxy Authentication Required
as a response to INVITE message
(REGISTER was successful)
and
error: could not find To in the reply
does anyone have some
See below. Add a T to your Dial options.
CLI show application Dial
-= Info about application 'Dial' =-
...
't' -- allow the called user to transfer the calling user by hitting #.
'T' -- allow the calling user to transfer the call by hitting #.
--Luki
Hi, I receive fax using spandsp. It works, however the tif
file it stored has no good quality. Any method to configure
that?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On 5/12/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
Digium cards do not have a built in DSP. Neither do the Sangoma as far
as I know. I don't know about VoiceTronix.
As yet !
As for digium cards latest cvs commits suggest that there is some
ongoing development on hardware based
On Thu, 12 May 2005, Chris Coulthurst wrote:
Does someone have a link to step-by-step instructions to making the
Line-In on the console sound card a MOH source?
You can probably use the Remote MoH patch from
http://bugs.digium.com/view.php?id=3565
Peter
How do you dial 1800 number using FWD?
Ive tried (fwd prefix) 1800numberblah and I get congestion..
This is information clearly stated on the FWD site.
*1800
as in http://www.freeworlddialup.com/advanced/peering_numbers
___
Asterisk-Users mailing
I get these on a consistant basis for most of the providers I have
configured. Some less than others. I even get it from my phone at
home to my * box at our data center.
So do I, on both our DSL connections. It's a result of the
connectivity of the ISP network (or the rest of the internet
Could someone please comment on the current state of chan_capi,
chan_misdn and chan_modem channel drivers in terms of functionality
(echo cancelation, fax, latency etc) and stability. Specifically, which
channel driver would be best for a passive PCI HFC or W6692 ISDN card.
The chan_misdn wiki
On Thu, 2005-05-12 at 19:28 -0400, Nabeel Jafferali wrote:
I have a customer that's located in France and he wants french phones
if possible. So I'm wondering if there's any one out there that found
a phone that can be change to french.
I believe snom phones have the option.
Confirmed
Polycom will do the trick..
On Fri, 13 May 2005 09:00:35 +0200, Dave Cotton
[EMAIL PROTECTED] wrote:
On Thu, 2005-05-12 at 19:28 -0400, Nabeel Jafferali wrote:
I have a customer that's located in France and he wants french phones
if possible. So I'm wondering if there's any one out there that
In article [EMAIL PROTECTED],
Daniel Salama [EMAIL PROTECTED] wrote:
I agree. I also prefer 2.6.x. That's what I run on my Debian
machines. However, REL3 does not support 2.6.x. They don't even have
an RPM for it. I would have to get one from kernel.org and that would
void any support
I battled with chan_capi during the week, and it was not fun.
PaulH
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Louw
Sent: Friday, 13 May 2005 4:46 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] chan_capi, chan_misdn and
On 5/12/05, Charles Wang [EMAIL PROTECTED] wrote:
Hi, ALL:
When I use astcc to do the prepaid function, but if I want to enable
call forward.
The result of CDR seems not correct.
UA 1011 make a call to UA , and UA forwards this call to a PSTN
number.
I think we shall charge
hello All
I am reading information about VoIP technology
For that i am concentrating on SIP (Session Initiation Protocol) and
RTP (Real Time Transport Protocol).
I am interested in implementing RTP over TCP
I found that there are some disadvantages of TCP, some are
1) TCP doesn't support
On Fri, 13 May 2005, Paul Hales wrote:
I battled with chan_capi during the week, and it was not fun.
Since I'm working on chan_capi, I would like to know what problems exist.
Can you please be more specific on what problems you have encountered?
Armin
PaulH
-Original Message-
The unofficial GXP-2000 resource, bugs, and information page is at
http://www.aussievoip.com.au/wiki-GXP-2000
--Rob
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Thursday, May 12, 2005 5:53 PM
To: 'Asterisk Users
On Thu, 2005-05-12 at 12:40 -0600, Tim Pushor wrote:
I had trouble calling people who were using FWD/SIP from my FWD/IAX
account. I switched back to using SIP and could call SIP users, but not
IAX users. I've since de-registered myself for the IAX *beta* and can
now talk to everyone again.
Following is the errors when I tried to compile oh323 in FreeBSD 5.3.
Asterisk is updated from cvs.
asterisk# gmake
for x in wrapper asterisk-driver; do gmake -C $x build || exit 1 ; done
make: illegal option -- -
usage: make [-BPSXeiknqrstv] [-C directory] [-D variable] [-d flags]
I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as
described in README file.
Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11
I tested it with following phones:
-- XLite (SIP softphone)
-- QMix SIP IP phone (PA168F)
-- SJPhone (H323
Following is the errors when I tried to compile oh323 in FreeBSD 5.3.
Asterisk is updated from cvs.
asterisk# gmake
for x in wrapper asterisk-driver; do gmake -C $x build || exit 1 ; done
make: illegal option -- -
usage: make [-BPSXeiknqrstv] [-C directory] [-D variable] [-d flags]
Hello -
I recently offloaded some of the SIP traffic on to a separate Asterisk box and
interconnected our main Asterisk system with the new system via IAX. The SIP
clients are running 7960's.
When a call is put on hold, often times when the call is pulled off hold, there
seems to be no RTP
bonjour,
tout d'abord merci de bien vouloir m'initier à la configuration de serveur
Asterisk.
J'expose mon soucis, je tourne sous windows XP, donc g pri asterisk version
windows == Astwind 0.1.1
Je ne pe pas changer d'ordinateur.
Mon objectif : Lancer mon serveur sans qu'il y aille d'erreur,
Hi all,
I'm trying to get IAX2 trunking between two * boxes and am having
extreme difficulty :) What happens is when the sending * server (the one
initiating the call) receives the ACCEPT back from the receiving server
it immediately replies with INVAL. I've checked the code and it seems to
be
Maybe in English you get more answer
Ismael.
Tutu Lord [EMAIL PROTECTED]
Enviado por: [EMAIL PROTECTED]
05/13/2005 11:14 AM
Por favor, responda a
Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
Para
asterisk-users@lists.digium.com
cc
On Fri, May 13, 2005 at 09:55:58AM +0200, Armin Schindler wrote:
On Fri, 13 May 2005, Paul Hales wrote:
I battled with chan_capi during the week, and it was not fun.
Since I'm working on chan_capi, I would like to know what problems exist.
Can you please be more specific on what problems
I had to use google translate to answer your question, so I'm going to reply
with my answer in the same way, the extra time it took me to decipher the
question will probably have to be reinvested:
Le safe_asterisk est juste un manuscrit pour remettre en marche le serveur si
il se termine
On Thu, 12 May 2005, Juanjo Portela wrote:
I was using iaxtel to make calls to 1-800 phones for free, but
unfortunatelly it is no working ...
freenum.org or e164.org ENUMs.
SIP/[EMAIL PROTECTED] seems to be the main provider providing ENUM
free phone coverage for +1.
I am using Asterisk-oh323 v0.7.1 with GNUGK. Please advise what must
be done to make FastStart work with SIP phones. Thanks.
On 5/12/05, VoIP Newbie [EMAIL PROTECTED] wrote:
Hi all,
When I enabled faststart in oh323.conf, calls from H323 endpoint to
SIP phones could not complete. The
On Fri, 13 May 2005, Louis-David Mitterrand wrote:
On Fri, May 13, 2005 at 09:55:58AM +0200, Armin Schindler wrote:
On Fri, 13 May 2005, Paul Hales wrote:
I battled with chan_capi during the week, and it was not fun.
Since I'm working on chan_capi, I would like to know what problems
Should I believe that at this time there is no DSP capable cards working
with Asterisk?
- Original Message - i
From: izo [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 13, 2005 6:11 AM
Subject: Re:
hello armin,
Armin Schindler schrieb:
- cleanup in chan_capi.c (I noticed some errors)
- add native bridging using CAPI Line-Interconnect
this would be very nice
I also was thinking about an application for receiving fax over CAPI, but
I'm not yet familiar with the current asterisk fax support,
I've just installed Astrisk with AMP. All work well but one thing is not
clear. I wanna add users to allow calls between SIP phones. I've added
extension but seems not to be enought.
How i can add SIP users and allow calls between they ?
Thanks ! Oz
--
O-Zone ! No (C) 2005
Hi !
Does anyone managed to send multipage faxes (in single TIFF file) with
app_txfax from spandsp package (i'm using 0.0.2pre18, libtiff 3.7.1)?
If so, I'm interested in format of TIFF file that has been sent sent
succesfully (tiffinfo fax-filename).
I'm having problems with app_txfax, sending
Hi!
To me, it seems like Asterisk are involved in alternating the sound/voice
running through it.
One thing is that it mutes DTMF digits.
I also got an Adit 600 channel bank connected via MGCP, which _might_ have
something to do with it,
but I can't find any settings in it, regarding DTMF
Thanks for this precision !! Certainly, a good news
for Asterisk users community.
- Original Message -
From:
Wiley
Siler
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, May 12, 2005 10:16
PM
Subject: [Asterisk-Users] IPVolution
Hi,
When trying to compile ASTCC i am getting the following error:
[EMAIL PROTECTED]:/usr/src/astcc# make install
mkdir -p /var/www
mkdir -p /var/www/html/_astcc
mkdir -p /var/www/cgi-bin/astcc-admin
chmod 755 ./astcc.agi
chmod 755 ./astcc-admin.cgi
echo | ./astcc.agi /dev/null
Can't locate
I apologise in advance if this is a silly question, as legacy
telephone technologies are really not my forte.
Is there an E1 card that can provide clock source? (E.g. Make my
asterisk server look like a telco to my legacy PBX system?).
What I am trying to achieve is the following:
Assuming that I am broadcasting 'legal' content, not having an external
live source to play will unsell the concept to many businesses that have
already purchased an external MOH source and want to integrate it.
Also, sometimes it is legal to broadcast radio (ie, you have paid the
Peter Valkov wrote:
I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as
described in README file.
Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11
I tested it with following phones:
-- XLite (SIP softphone)
-- QMix SIP IP phone (PA168F)
--
--- Kevin Bockman [EMAIL PROTECTED] wrote:
May 12 22:27:05 VERBOSE[2442]: -- Executing
[1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m,
[1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack
May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640
May 12 22:27:05
John Daragon wrote:
Peter Valkov wrote:
I have build asterisk from latest CVS HEAD-05/09/05 with H323 support
as described in README file.
Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2
kernel-2.6.11
I tested it with following phones: -- XLite (SIP softphone)
-- QMix SIP IP
I installed Asterisk CVS-NHEAD-05/13/05-01:59:30 and placed few call
in and through successfully. I was trying to set up the Realtime -
picking the sip.conf and extensions.conf from mysql. I was going
through some wiki pages, but what i don't understand is - which
configuration change makes
I am not an expert yet ;)...but VoipJet is very picky.. in your exten =
when I tried with the default it didn't work, when tried as they ask in the
FAQ's it workedyou must keep the exact format with the account
number...
[Voipjet]
exten = _1NXXNXX,1,SetCallerID(4153574000); Set your
Hi.
I'm new to asterisk and, one way or the other, I manage to get it working
for me.
But I'm having a hard time getting calls going to and coming from the
same provider, since the definition of the peer in sip.conf seems to be
different AND not compatible for incoming and outgoing call.
Personally I thought that VOIPJET has the best service and
documentation including simple up to date CDRs also.
They do not offer incoming, at least not to me
If you follow the instructions on their site it will work, if you are
useing AAH then maybe you should look into editing the files by
VoipJet are not too bad, little pricey though.. theres better around.. a
matter of looking :-)
Regards,
Sahil Gupta
VoiceValley
On Fri, 13 May 2005, Andrew Latham wrote:
Personally I thought that VOIPJET has the best service and
documentation including simple up to date CDRs also.
They do not
Hi,
this is a rather ugly solution I devised.
Create a script called 'ast-playlinein' (or whatever) in /usr/sbin, as
follows:
#!/bin/bash
/usr/bin/arecord -q -c 1 -r 8000 --buffer-size=2048 -f S16_LE -D hw:0,0
-t raw
In musiconhold.conf:
[classes]
default =
You need the asterisk perl module. Check here:
http://asterisk.gnuinter.net/
Darren Wiebe
[EMAIL PROTECTED]
Robson Ribeiro wrote:
Hi,
When trying to compile ASTCC i am getting the following error:
[EMAIL PROTECTED]:/usr/src/astcc# make install
mkdir -p /var/www
mkdir -p /var/www/html/_astcc
I need help
configuring my Cisco 7940G's for my office. I have [EMAIL PROTECTED] running on the server. Right now
all my phone is saying is "Defaulting CM to TFTP Server". I have 5 Cisco
7940G's, a Cisco ATA186, and a Zyxel 2000W Wi-Fi Phone. Right now, my VOIP lines
are coming in on the
Hello
I was just
stuck around as to how I configure my Asterisk to access extensions from Mysql.
I have made all the necessary changes in the extconfig.conf, the extensions.conf,
res_mysql.conf, res_config_odbc.conf,res_odbc.conf as they have mentioned on
the site www.voip-info.org.
There is a good section in the handbook on setting up
Cisco phones.
you well need to get 7.4 SIP firmware from cisco. it
sounds like you are running MGCP now.
http://asteriskathome.sourceforge.net/handbook/index.html
--- Adam Collard [EMAIL PROTECTED] wrote:
I need help configuring my Cisco
Hello!
Is it possible to make the console autoanswer incoming calls to some
extensions?
Something like this:
; Dial Console with user pickup
exten = 123,1,Dial(Console/dsp)
; Dial Console with autoanswer
exten = 321,1,Autoanswer(Console/dsp)
I want to be able to place calls through the
Then I hope to receive some reports on what is buggy/not working, wishlist
and hopefully also some reports on what works well.
There are at least two anoying bugs:
1. The Busy-Applicatzion does not work, there seems to be no was to
singnal Busy to the caller is no SIP-Phone is ready to answer the
I agree. I've been using voipjet since before their formal launch..I
have account number 63. They've been amongst the most reliable in my
experience. If they offered DIDs in my area I'd have those as well.
Michael
On Fri, 13 May 2005 07:42:32 -0500, Andrew Latham wrote:
Personally I thought
I installed a new * server and copied the sip.conf and extensions.conf
from my existing setup to the new box. I created my outbound trunk
with a different broadvoice account and am able to dial out without
issue. I am able to dial all extensions but I go straight to voicemail
without any ringing,
Hi all.
I'm curious as to the current status and development of a way to monitor
incoming voicemail in Asterisk. IE: The screen calls with the
answering machine feature -- the ability to listen to and break into a
currently-recording voicemail if you want to.
This feature would be very
can you post your mgcp.conf file.
From the debug output it looks like * can not find the gateway in the mgcp.conf
(* goes on to tell you it can not match the endpoint, because it first has to
find the gateway device...)
- Original Message -
From: Ben Dugdale [EMAIL PROTECTED]
To:
Hi,
time to clear some things up. :)
The new version of chan_capi (0.4.0) is still work in progress (no, I
have not dropped chan_capi in favour of BRIstuff). I harmonized the
dialstring syntax with chan_zap, so you can just use CAPI/g1/...
instead of those strange constructions with the outgoing
I'm #11 but I have notice of late a few problems but nothing major
given the price differences assuming you don't have the volume to
commit to another carrier directly for the destinations you are
after.
-- William
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Asterisk-Users mailing list
hello,
iax trunking not working we actually testing dial 500(Digium) two or
three calls simultaneously but bandwidth graph shows 95 to 100kbps not
match the results shows on wiki iax bandwidth pages i enable trunk=yes
in iax.conf is there any tweaking or optimization because i
desperately need
I use AAH with VoipJet and it works perfectly. Setup was a breeze with
absolutely no hand coding of configs required.
VoipJet is without a doubt the best outbound provider I have come
across. No problems at all yet. knock on wood
And the call quality has been awesome.
Anyone having trouble
What codec are you using?
-Original Message-
From: Adnan Ahmed [mailto:[EMAIL PROTECTED]
Sent: Friday, May 13, 2005 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] iax trunking not works!
hello,
iax trunking not working we actually
1.3 cents minute dialing? That is one of the lowest prices out there.
Maybe for you in Australia but in North America, it is a very nice deal.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sahil
Gupta
Sent: Friday, May 13, 2005 5:49 AM
To: Andrew
Mamadou Lamine KA wrote:
Should I believe that at this time there is no DSP capable cards working
with Asterisk?
That is correct as far as I know. The entire DESIGN of Asterisk is to
do the DSP work in software. Rumor has it that Digium is coming out
with a DSP version of their cards (or a DSP
It is not that they are not working with Asterisk... It is that there
are none available.
Go check out the link that was sent to you before... Here it is
again
http://www.zapatatelephony.org/
As you can see, Zapata (which drives Asterisk) was originally designed
to be a chipless DSP
Almost positive iLBC is not allowed Use uLaw...
This is directly form the install instructions...
Step 3A (recommended): Set your codec to G.711 ulaw for optimal sound
clarity and minimal transmission delay. In iax.conf (found in
/etc/asterisk) locate the codec section and include the
Videotel !!! : French software, Video hard phone, Excellent browser...
see it at : http://www.call.fr
Works fine with Asterisk.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de Martin Roy
Envoye : vendredi 13 mai 2005 00:52
A :
I am having trouble with zttest on a Tyan board, dual AMD Opteron's on
FC3. Tried different kernels, no IRQ sharing, everything looks in
order. My zaptel modules load fine, but if I run zttest, it just
hangs. Below is the strace output and you can see where it stops.
Anyone have any ideas?
Hello all.
I have an ISDN passive card (HC HFC 2BDS0) using HiSAX driver (since
this is a passive card, I can't use the CAPI driver... ). All I want
is recieve faxes and store them on diferent folders depending the
destination number. So far I was able to do a similar thing with voice
(vbox),
My home asterisk seems to work- I can call from
one internal phone to another. However, just leaving my system idle always
generates an error message relating to a NOTIFY. See the log below.
Any ideas?
Thanks,
Mike
--MESSAGE
FILE-
to
I want to know if I buy a Tyan Transport GX28 (B2881) will it work
with a TDM400 card? As the expansion slots are only (2) 64-bit
133/100MHz PCI-X. I never tried PCI 2.2 compliant card in a PCI-X
slot so I don't know if it can even fit in the slot and if it does
will it be seen?
Thanks
Hi,
I am having trouble with dropped calls in Asterisk. I've done a bunch
of searching but all I could find was setting busydetect and
callprogress to yes in zapata.conf to help combat the problem, but I did
this to no avail. The following is the output from
/var/log/asterisk/full at the
Look a little closer...
WARNING[2442]: Call rejected by 66.246.220.19: No such context/extension
that certainly does not imply an incorrect codec!
Almost positive iLBC is not allowed Use uLaw...
This is directly form the install instructions...
Step 3A
I own and operate a number of franchised Sylvan Learning Centers where
I recently upgraded to an all VOIP phone system (Asterisk) with one
VS-1 and about 25 extensions scattered around the country. I had
originally setup a Dell 420 SC but the Dell had incurable buss issues
with single span
On Fri, 13 May 2005, Frank Sautter wrote:
hello armin,
I also was thinking about an application for receiving fax over CAPI, but
I'm not yet familiar with the current asterisk fax support, so I need to
learn more here. Maybe some else can inlight me here...
chan_capi currently supports
On Fri, 13 May 2005 07:59:09 -0700
Wiley Siler [EMAIL PROTECTED] wrote:
Almost positive iLBC is not allowed Use uLaw...
They do allow for iLBC. From their FAQ page:
Codecs. Carriers with primarily business customers should
use the G.711 codec when sending VoIP traffic to VoipJet.
This
I am interested in implementing RTP over TCP
Why? If you want to permit operation through a firewall
that blocks UDP, there are packages that provide VPN
tunnels over TCP or even HTTP. You could then run
any VoIP system over that VPN. As you said, delay
performance would sometimes be awful.
I'd really like more info on how to correctly format a tiff for tx_fax too.
The only Tiff's we've been able to send using tx_fax with consistant
success are the ones that rx_fax creates when it receives an incoming
fax.
Here's the Tiff info from one such if its any help.
TIFF Directory at
On Fri, May 13, 2005 at 12:33:10PM +0200, Armin Schindler wrote:
On Fri, 13 May 2005, Louis-David Mitterrand wrote:
On Fri, May 13, 2005 at 09:55:58AM +0200, Armin Schindler wrote:
On Fri, 13 May 2005, Paul Hales wrote:
I battled with chan_capi during the week, and it was not fun.
Hi Mike,
Probably the same problem i had i while
back.
The ATA-box dont support message waiting indicatons from
asterisk and therefore dont respond to the message, asterisk restries 5 times
before giving up with a warning in the log.
Iresolved it by removing the mailbox= in sip.conf for
I guess you don't know how to read.
Failed to connect database server asterisk on. Check debug for
more info.
Holy cow! You failed to connect to your database! Imagine that. I wonder why
it isn't working. Hmm. Could it be that? Did you check the debug for more
info? Probably not seeing as you
Robert Webb wrote:
On Fri, 13 May 2005 07:59:09 -0700
Wiley Siler [EMAIL PROTECTED] wrote:
Almost positive iLBC is not allowed Use uLaw...
They do allow for iLBC. From their FAQ page:
Codecs. Carriers with primarily business customers should use the G.711
codec when sending VoIP traffic to
Good catch. Did not see the FAQ.
Robert, are you the one having problems getting this running in AAH?
W
-Original Message-
From: Robert Webb [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 12, 2005 11:22 PM
To: asterisk-users@lists.digium.com; Wiley Siler
Subject: Re: [Asterisk-Users]
On 5/13/05, Atul Thosar [EMAIL PROTECTED] wrote:
hello All
I am reading information about VoIP technology
For that i am concentrating on SIP (Session Initiation Protocol) and
RTP (Real Time Transport Protocol).
I am interested in implementing RTP over TCP
I found that there are some
You are completely correct.
I see by the called number that the user is in Phoenix? I am too.
Call me at 4804230118 ext. 1003 if you want some off list assistance
with this.
I have mine running just fine with AAH 0.09.
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
It would be nice if you post how you set this up to either the wiki or right
here. Just a few lines would do nicely. There seems to be allot of people
who use voipjet and aah and both are good products.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Does anyone know of a BYOD provider that terminates calls to NCFA
numbers (UK 'national rate'). I enjoyed broadvoices unlimited to those
numbers, but this is getting silly now, it doesnt work and no answer if
after switching to a new provider it will ever work.
Can anyone suggest an
On Friday 13 May 2005 10:13 am, Nathan Pralle wrote:
Hi all.
I'm curious as to the current status and development of a way to monitor
incoming voicemail in Asterisk. IE: The screen calls with the
answering machine feature -- the ability to listen to and break into a
currently-recording
hello
i want to insert delay into callfile execution.
UA6000(callbackNumber) this will create call file
UA---asterisk(callfile)
how to insert delay into this callfile execution.
thanks
Kamran
__
Do you Yahoo!?
Make Yahoo! your home
On Fri, 13 May 2005, Louis-David Mitterrand wrote:
On Fri, May 13, 2005 at 12:33:10PM +0200, Armin Schindler wrote:
On Fri, 13 May 2005, Louis-David Mitterrand wrote:
On Fri, May 13, 2005 at 09:55:58AM +0200, Armin Schindler wrote:
On Fri, 13 May 2005, Paul Hales wrote:
I battled
But I'm having a hard time getting calls going to and coming from the
same provider, since the definition of the peer in sip.conf seems to be
different AND not compatible for incoming and outgoing call.
Sometimes what is needed can be provider-dependent. Every provider
I've seen gives an
I have the same problem, any ideas people?
Gustavo Alvarez
Sander crombeen at rommelweb.nl
Sun May 1 12:17:31 CDT 2005
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On Fri, 2005-05-13 at 09:02 -0700, trixter http://www.0xdecafbad.com
wrote:
Does anyone know of a BYOD provider that terminates calls to NCFA
numbers (UK 'national rate'). I enjoyed broadvoices unlimited to those
numbers, but this is getting silly now, it doesnt work and no answer if
after
Armin Schindler wrote:
please look for the neccessary patches at:
http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2
That is exactly what I was thinking about.
I did not have a close look into the patch yet, but this archive seems to be
incomplete. Only changed files are part of the
hallo klaus-peter,
Klaus-Peter Junghanns wrote:
The new version of chan_capi (0.4.0) is still work in progress (no, I
have not dropped chan_capi in favour of BRIstuff).
that was my assumption, as there was no progress so many months.
i'm very happy, that you are back on developing chan_capi!
I
Wel, that is the real issue. There is no secret method.
You literally just add it to the trunks then set it in your outbound
routing
being careful to make sure you have dial patterns set that match
correctly.
So
Here is the trunk definition
Hackish, but it works. I've used it myself several times. This, of course,
assumes you are using Zap channels for incomming calls. If not, then you'd
need to find another way to listen to incomming calls - perhaps ChanSpy, tho
i've not been able to get that to work - crashes my * box with
TCP is too slow for Real time Apps. If you have packet errors TCP will
try to resend the packet. This will create latency issues. This is why
UDP is used for Voip. 1 or 2 missing packets is not going to be missed.
If you look at your Stats. you'll see a few of them.
Stewart Nelson wrote:
Works great for me also.
On Mon, 2005-05-09 at 14:33 +0200, Altus Snyman wrote:
How well does the sangoma cards work with fedora core 3
Im doing the research on what hardware/os I need to use
Please help and advice
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I had the same problem, there were a couple problems, mostly with my
dialplan:
AAH 1.0 Config
Trunk Settings
Trunk Name: voipjet
Outbound Caller ID: Tech Data Pros 9495027819
Maximum Channels: 4
Dial Rules: 1949+NXX ; you need to add local area code
Outgoing Settings:
Trunkname: voipjet
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