, or should I simply
change to the AGI script approach?
Best regards,
--
Carl-Fredrik Enell
Tähteläntie 70B
FIN-99600 Sodankylä, Finland
-
URL: http://www.is.kiruna.se/~fredrik
Work URL: http://www.sgo.fi/~fredrik
11:09:12 AM Fotos 18/08..:
Imagens Anexadas..: DSC_0401.jpg - DSC_0402.jpg - DSC_0403.jpg
Videos Hotmail.com..: www.hotmail.com/videos.avi
_
Brrr... its getting cold out there Find someone to snuggle up with
http://a.ninemsn.c
dnesday 13 May 2009 17:55:41
> carl Lougher wrote:
> > Howdy,
> > How do i perform a lookup from a remote odbc database
> in the asterisk
> > dialplan?
> >
> > I can do it with mysql but not sure of commands for
> odbc connection.
>
> See func_odbc.co
Howdy,
How do i perform a lookup from a remote odbc database in the asterisk dialplan?
I can do it with mysql but not sure of commands for odbc connection.
Cheers!!!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
a
Hi,
I'm trying to get my Asterisk 1.4.24.1 server working with radius and aradial.
I have radiusclient-ng installed and asterisk radius cdr.
My cdr's fail to write to the database and i'm not sure how to authenticate
each call.
Anyone got this working or can offer any help. I've read all the r
have that possibility.
Mike
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of carl Lougher
> Sent: Thursday, April 23, 2009 4:54
> To: asterisk-users@lists.digium.com
> Subject: [aste
Hi,
Is there any method of getting call park working on different numbers for
different customers on the same asterisk server?
Currently running asterisk 1.4.23.1
Cheers!!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.co
Howdy,
Is it possible to send a reinvite after the media has connected?
Scenario:
Inbound call hits asterisk ivr then is sent out to an extension using the dial
command. We have to carry the rtp streams in this case as asterisk cant send
the reinvite after the ivr has stopped playing the messag
Howdy,
Scenario:
Asterisk server
Customer connected over internet using nat
Customer phones are Linksys 942 with Stun enabled
Issue:
Inbound and Outbound calls work fine. But when phones call each other
internally we have to carry the voice stream ie using t on dial commands.
Question:
Is there
Yeah but doesnt help for extensions that have or require call-limit=1.
--- On Tue, 31/3/09, carl Lougher wrote:
> From: carl Lougher
> Subject: Re: [asterisk-users] Call-limit=1 breaks attended transfer
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
ks attended transfer
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Date: Monday, 30 March, 2009, 10:50 PM
> carl Lougher wrote:
> > Howdy,
> > Was there ever a fix for this?
> >
> > I have Trix 2.6 running asterisk 1.4 and have to se
Howdy,
Was there ever a fix for this?
I have Trix 2.6 running asterisk 1.4 and have to set an extension with
call-limit=1. However that user can no longer do attended transfers from Linkys
962 ip phone.
Is there anyway around this?
Cheers,
Taff..
Howdy,
I have the following issue and would like to know if anyone has got around this
before.
IP Phones - Linksys 942
Sip server - Asterisk 1.4.13
Stun server - Vovida
Ok heres the issue. We have multiple client phones on their own network behind
a natted connection. We have setup the phones
]
> exten => s,1,Wait(1)
> exten => s,n,Answer()
> exten => s,n,Background(tentant2sounds/welcome)
> exten => s,n,SetMusicOnHold(tenant2)
>
> Use that with the previously supplied info.
>
> Darrick
>
> carl Lougher wrote:
> > Hi,
> > I
nya
>
> [moh-company-b]
> mode=files
> directory=/var/lib/asterisk/moh/companyb
>
> regards,
> nhadie
>
>
> carl Lougher wrote:
> > Howdy,
> > Is there a way to apply a music on hold class to
> different context user groups?
> >
>
Howdy,
Running asterisk 1.4
Is there a way to check the simultaneous sip calls in asterisk and display with
mrtg or some graphing app???
Also is there a way to segregate these based on extension or context?
Cheers,
Taff..
___
-- Bandwidth an
Howdy,
Is there a way to apply a music on hold class to different context user groups?
I have multiple clients on my asterisk server and they each want different
music on hold.
Company A
Company B
Any help much appreciated..
Thanks,
Taff...
__
Howdy,
Running asterisk 1.4.13
Sometime when running a sip reload the clients are unable to make and receive
calls..
Any pointers?
No errors in debug or asterisk console so far..
Cheers,
Taff..
___
-- Bandwidth and Colocation Provided by ht
Howdy,
Whats the best way to change the callerid for internal
and external calls.
At the moment using callerid- Fred <04412345>
sends callerid as Fred 04412345 for internal calls
when his internal extension is 200.
How can i change the callerid for internal calls but
also keep the specifi
I've connected to Verizon BRI circuits and had major echo issues. Moved to a
Paetec PRI and bing all calls now work great.
- Original Message -
From: Klaverstyn, David C
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, June 06, 2007 1:47 PM
Subject:
I've noticed that there is the odd vnak message displayed in my asterisk syslog
traces. Would have to alert on those i'd assume..
- Original Message -
From: Matt
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, June 06, 2007 1:44 PM
Subject: Re: [as
Sent: Wednesday, June 06, 2007 12:07 PM
Subject: Re: [asterisk-users] Asterisk call quality detection
On 6/6/07, carl Lougher <[EMAIL PROTECTED]> wrote:
Hi Chaps,
Is there a way to detect/highlight poor quality voice
calls going through an asterisk server?
Was thinking of picking up a c
Hi Chaps,
Is there a way to detect/highlight poor quality voice
calls going through an asterisk server?
Was thinking of picking up a cdr record or some other
variable showing poor quality on calls when the
internet is having issues.
Is there any qos or poor audio quality variables
available?
Ch
Setup:
Asterisk server in NY.
Cisco 7960 IP Phones in NY and London.
Dedicated T1 from NY to Ldn.
T1:
Latency - 100ms
Qos applied
No errors
Default codec on Ldn IP Phones = g711alaw
Default codec on NY IP Phones = g711ulaw
Both codecs allowed on each phone.
Issue:
Calls on IP Phones from NY to L
time, does anyone else have
any suggestions? Are there some specific build options or kernel
flags we should try? Are there any other approaches that someone
might recommend?
Thanks in advance for your time.
Carl
___
--Bandwidth and Colocation provided
We are running the default asterisk package on Ubuntu Dapper. Our
connection to the PSTN is over an IAX trunk with our provider. We are
getting really bad call quality on calls over the IAX trunk--voice
seems to be garbled or out of order and often completely breaks up.
But on internal calls bet
nd say that our zttest
results are bad and that our server is probably the source of the
problem. However, our calls between extensions are superb, so I can't
believe that our server is to blame, since only calls to and from the
PSTN seem to have problems.
Thanks in advance
Upgrading to ver 1.2.10 fixed it.
--- carl Lougher <[EMAIL PROTECTED]> wrote:
> Hi,
> Need to get the following working:
>
> 1. User calls ext 750.
> 2. If no answer or busy go elsewhere.
> 3. If answered and press 1 accept call.
> 4. If answered and not pressed 1 or
Hi,
Need to get the following working:
1. User calls ext 750.
2. If no answer or busy go elsewhere.
3. If answered and press 1 accept call.
4. If answered and not pressed 1 or timed out then
send call to be redirected to the busy or no answer
option.
The issue is that the call gets accepted if an
This is a weird one.
Network:
Asterisk ver 1-0-9 on DL360.
10 Cisco 7960g phones with 3.8.2 SIP Load.
Gateway - Cisco 2811 router with 4 x verizon bri's.
Network - Private vlan with 1ms response times to all
devices.
Issue:
Intermittent echo on outbound/inbound calls. Users
hearing their own voic
Hi,
Running asterisk ver 1-0-9
Trying to send a call to a mobile phone and playback a
message to the user to press one to accept the call.
If 1 isn't pressed then the call needs to be re-routed
back into the asterisk dialplan.
Tried various macros etc but if one isn't pressed the
call still gets
the dialplan.
On 6/16/06, Doug Lytle <[EMAIL PROTECTED]> wrote:
Carl Youngblood wrote:
> No, ${EXTEN} contains "i" at that point in the dialplan.
exten => 123,1,Set(_TMPEXTEN=${EXTEN})
exten => i,1,SayDigit({$TEMPEXTEN})
You need to read the document in the Asterisk source
Thank you! Thank you! I had been trying all sorts of convoluted ways
to get that information. That was very easy.
On 6/16/06, Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Variables
Use ${INVALID_EXTEN}
On 6/15/06, Carl Youn
ood and proper and take it from there.
Thanks,
Steve Totaro
Carl Youngblood wrote:
> Of course I'm trying to deal with the network problems, but it's nice
> to have another method of verifying that everything is working.
> Frequently there are people who don't complain, so
on.
> Is there a way I can erase the fact that an extension
was matched? Or
> is there some other way of accomplishing what I am trying
to do?
>
> Thanks,
> Carl
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-U
sion was matched? Or
is there some other way of accomplishing what I am trying to do?
Thanks,
Carl
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
e should be a record somewhere if a call was
terminated abnormally.
Thanks,
Carl
On 6/14/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
Carl Youngblood wrote:
> I have been getting occasional reports of dropped calls from the users
> of our asterisk system. Is there anything I can monito
I have been getting occasional reports of dropped calls from the users
of our asterisk system. Is there anything I can monitor in my logs or
in the console to see when a call is dropped? I'd like to see if
these drops coincide with network traffic problems.
Thanks,
Thanks. What is it in the 2.6.13-based kernel that improves timing?
Should I expect to see a significant improvement if I upgrade to it?
On 6/13/06, Mike Fedyk <[EMAIL PROTECTED]> wrote:
IAX trunking and meetme conferences are some of the heaviest users of
zaptel timing. I'd suggest if you don
Our asterisk system gains access to the PSTN through a voip provider.
We have no digium or other telephony hardware in our system. Do the
zttest results still matter to us? Our results were as follows:
--- Results after 1007 passes ---
Best: 100.00 -- Worst: 99.780273 -- Average: 99.975763
FYI, the digit timeout was simply too short in my IVR. After
increasing that everything worked fine. This problem only showed up
on cell phones, many of which don't allow you to type long digits so
your keypresses have more silence in between them.
On 5/12/06, Carl Youngblood <[EMAIL P
n that doesn't
> send a page to a phone that is currently in a call. But to do this I
> need a function that will tell me if a device is in a call. Any
> suggestions?
>
> Thanks,
> Carl
Snip.
___
--Bandwidth and Colocation provided by
old the value but mysteriously
doesn't.
Thanks in advance for your help.
Carl
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
page to a phone that is currently in a call. But to do this I
need a function that will tell me if a device is in a call. Any
suggestions?
Thanks,
Carl
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSU
Sorry to post on the asterisk users list, I know AAH is not exactly
related, but there is something wrong on their forum right now. I
can't post there, even though I'm logged into sourceforge.
Thanks,
Carl
___
--Bandwidth and Colocation prov
Is it possible to send an API command to dial an extension and playback a
specific announcement using application and appdata commands.
Scenario:
User adds different announcements daily (can't used fixed name for Playback
file).
Call command dials user and plays back specific announcement message.
works great now.
But instead of terminating with 50 Ohm at one end of the line, I
put 100 Ohm termination in both ends of the line...
Thanks for the help!
--
Greetings, Carl Andersson a.k.a Zaphod Beeblebrox
Email: [EMAIL PROTECTED] - ICQ: 1705837.
-=*=- FwD VOIP: 645040 - Iax
t;>
>> Is it only possible to use one device with a HFC card in NT mode or is
>> there something else I need to do first to make it work with two
devices?
>>
> Hi Carl,
> I just started yesterday afternoon with exactly the same setup so you
> are a bit ahead of me.
something else I need to do first to make it work with two devices?
--
Greetings, Carl Andersson a.k.a Zaphod Beeblebrox
-=*=- FwD VOIP: 645040 - IaxTel VOIP: 17006906266 -=*=-
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
N0906.JPG
The circuit board layout was made with an old version of Tango PCB, but I've
included Postscript versions of both the silkscreen and the back of the board.
--
Greetings, Carl Andersson a.k.a Zaphod Beeblebrox
-=*=- FwD VOIP: 645040 - IaxTel VOIP: 17006906266 -=*=-
_
Guillermo.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Jamie Carl
Resident Ge
is strictly prohibited. If you have
received this communication in error, please notify Brendata immediately on:
+44 (0)1268 466100, or email '[EMAIL PROTECTED]'
Brendata (UK) Ltd
Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK
Registered Office as above. Registered in England No
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Jamie Carl <[EMAIL PROTEC
trieve the call? Or at least where there is some documentation on this 'feature'?
TIA
--
Jamie Carl <[EMAIL PROTECTED]>
Resident Geek
Achieve Corp.
+61262648200
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://
'm currently running 1.0.7-stable and think it might be a version
problem. Is the supervised transfer feature available in 1.0.7 or do
i need to suck down a new version from CVS?
Otherwise, apart from setting up features.conf, is there anything else
i'm missing?
TIA,
Jamie.
on like that.
Well if someone have BCM + i2004 + firmwares, they can send me a dump of the
network traffic. If it's ok, the firmware update will be available on the
next version of chan_unistim.
--
Carl
___
Asterisk-Users mailing list
Hello,
Cedric Hans has released an UNISTIM channel driver for asterisk (stable).
You can download it at :
http://mlkj.net/asterisk/chan_unistim-0.9.2.tar.bz2
Copy of README :
This is a channel driver for Unistim protocol. You can use at least Nortel
i2004 phones with it.
Only few features are sup
or PRI? Both of
> these use the cpu for the signal processing and are a lot cheaper.
Yes, except if your main usage is faxing (in/out), in this case, use
chan_capi + patch/hylafax. You'll get a reliable solution based on hardware
DSP. If it's a voice only usage,
On Tuesday, 15 February, 2005 02:46 : Stefan Gofferje
<[EMAIL PROTECTED]> wrote:
> Carl Sempla schrieb:
>> On Monday, 14 February, 2005 02:30 : George Cohn
>> <[EMAIL PROTECTED]> wrote:
>>
>>
>>> Stefan Gofferje wrote:
>>>
>>>
d...
>
> I just came back from a Nortel roadshow and was told it's H.323.
Nop. This phone uses a proprietary protocol except for the audio part (RTP
G711, G723 and G729).
Some work are in progress right now for an UNISTIM support in asterisk.
--
Carl
___
>>
> 1 is too high you can safely lower it to sleep(1) there's a while
> over there
> otherwise it will lock the channel for 10 seconds.
> that's code from chan_capi fax patch right?
Nop, it's in the origina
ba8) at chan_capi.c:563
563 usleep(1);
And past the results of these commands.
Good luck
--
Carl
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UN
Wed Oct 27 18:55:32 2004
@@ -556,7 +556,7 @@
}
}
// wait for the B3 layer to go down
- while (i->state != CAPI_STATE_CONNECTED) {
+ while ((i->state != CAPI_STATE_CONNECTED) && (i->state !=
CAPI_STATE_DISCONNECTED)) {
usleep(1);
detected.
It may be possible to alter the code and use the asterisk dsp code instead.
Regards,
--
Carl
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update optio
with Eicon you must configure it and load a
firmware before.
--
Carl
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
NING[11275]: channel.c:472 ast_channel_walk_locked: Avoided initial
deadlock for 'CAPI[contr1/173720007]/7', 10 retries!)
--
Carl
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCR
>> You need a kernel support for you card and you also need to load a
>> firmware for some cards.
>> If you have a message like "CAPI not installed!", check your kernel.
>>
>>
> I use the linux 2.6.7 kernel which came with the knoppix distro I've
> installed on the box. I have checked with make x
> "chan_capi.c:2603 load_module: Unable to load config capi.conf"
You need to create this file /etc/asterisk/capi.conf
with the following content :
[general]
nationalprefix=0
internationalprefix=00
[interfaces]
msn=50
incomingmsn=*
controller=1
softdtmf=0
accountcode=
context=incoming
;echosquel
(tried 1, 2 and 1,2,3,4, doesn't matter). I
> always get:
If your devices= is to low, for example =1, then when you receive a 2nd
call, you'll get the following error :
ERROR [3075]: chan_capi.c:1696 capi_handle_msg: did not find device for msn
= 1234xxx and the caller get a busy signal.
files created by Record() ?
You can also read raw GSM with winamp.
http://winamp.com/plugins/details.php?id=142107
--
Carl
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
On Tuesday, 26 October, 2004 15:46 : Kostur, Andre <[EMAIL PROTECTED]>
wrote:
>> Say... no chance that you could post your server code somewhere?
>> We've got some i2004's kicking around doing nothing
The source code is available at :
http://www.mlkj.net/U
port my standalone code into asterisk, it would be great.
--
Carl
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hello,
Can I use the Diva Server PRI/E1-30 card with asterisk (with chan_capi or
i4linux) ?
If asterisk detect a fax, how divert the call to the onboard DSP ?
How can I handle the fax with hylafax (capi or pseudo serial device ?)
Thank you,
--
Carl
SJPhone from SJLabs.
www.sjlabs.com
Also, a lot of simple questions like this can be answered by looking at
www.voip-info.org
There is a large section there on different soft/hardware phones.
Regards,
Jamie Carl
Chief 'Stuff' Officer
J-Code International
Email: [EMAIL PRO
alling=pri_cpe
context=incoming
channel => 1-15,17-31
signalling=fxo_ks
context=internal
channel => 32-33
Thanks,
--
Carl
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or upd
installed? Usually they'd
install the NTU and not put the voice router onsite until sometime
later.
Normally though you would be provided with E1s by Optus which the TE401p
should work fine with.
Regards,
Jamie Carl
Chief 'Stuff' Officer
J-Code International
Email: [EMAI
>
>
> Jamie Carl [EMAIL PROTECTED] wrote:
> > Just a quick question. Are there any known issues using the zaptel
> > drivers on different linux distros? ie: is a 2.4 kernel the only
> > requirement?
> >
> That should read "2.4 or later". I&#x
symptoms as setting it to ground start
signalling instead of loop start on a loop start PSTN line. However it
is definately set to fxsls and even fxsks has been tried.
I've purchased another X100P just in case it IS a hardware issue.
tia...
--
Regards,
Jamie Carl
Chief 'Stuff&#
Victor Rini wrote:
Hello,
After poking and prodding at Asterisk and Zaptel for over a couple
years now, I've dedicated some time to actually reading the code and
trying to figure it out.
It's been fascinating. With the driver source on one part of the
screen and a pdf of "Linux Device Drivers"
Bob Knight wrote:
I have MIBs for whatever version I am running that I am more than
happy to share. Anyone know where I can place these for public access.
Sort of like the freedomphones site for Polycom. We could then
put pointers on the wiki.
Thanks for the info tho. If mbrowse is console based
Bob Knight wrote:
There is a linux package called mbrowse that you can use with your
mediatrix mibs.
I can get and walk everything in my 1204's.
For some reason I have not had any success with writes, but I have not
spent
that much time on it.
I don't even have the MIBs which is half the problem
Thanks to everyone for their help and comments on this. You've all been
very helpful. I've actually got outbound calls working on it fine right
now without having to change the configuration on the Mediatrix box at
all, as I don't have the Unit Manager Software at the moment. Outbount
seems
Hi all,
I just picked myself up a Mediatrix FXO SIP gateway to play around with
and hook into Asterisk but have no documentation.
Are there default passwords or IP's that I need to know if I do a
factory reset?
Or better still, would anyone have a User Manual they could send my
way? Any help
ng I'm
missing. This used to work fine. Has something changed in the zaptel
driver? Are there any undocumented settings I can tweak to possible get
this working again?
I'm about to chuck the card and go for a SIP or MGCP gateway but if I
can not spend the cash, I will. Anyone with ideas?
on another non-PSTN FXS interface. Still working on it tho
so I may get it working again soon.
Regards,
Jamie Carl
Chief 'Stuff' Officer
J-Code
Web:http://www.j-code.net
Email: [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
ice is a little bit slow
> but they seem to be pretty decent. I'd recommend checking them out.
> www.nufone.net
>
> Darren Wiebe
> [EMAIL PROTECTED]
>
> Carl wrote:
>
> >Ha ha I get the picture :-)
> >I've tried Voicepulse but can't manage to get
?
Do you need to add info to extensions file to
point to context in the h323.conf file?
How do u send an account number with the call so
that the third party gatekeeper can verify?
Your help will be much appreciated!
Carl.
---Outgoing mail is certified Virus
Free.Checked by AVG anti
Same as mine. Strange!
I'll keep trying. Cheers.
- Original Message -
From: "Girish Gopinath" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, February 28, 2004 9:53 PM
Subject: Re: [Asterisk-Users] DTMF Issues with SJPHONE
>
> >What ver of SJPHONE?
>
> SJPhone Evaluation Ve
erconnects?
- Original Message -
From: "John Fraizer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Cc: <[EMAIL PROTECTED]>
Sent: Sunday, February 29, 2004 2:43 AM
Subject: Re: [Asterisk-Users] Iconnect behind NAT
>
>
> carl wrote:
> > Anyone got an example of
Anyone got an example of sip and extensions confs
for Iconnect outgoing calls behind NAT.
What ver of SJPHONE?
Thanks for the voicemail stuff :-)
- Original Message -
From: "Girish Gopinath" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, February 28, 2004 7:48 PM
Subject: RE: [Asterisk-Users] DTMF Issues with SJPHONE
>
> >Has anyone had a similar issue with Aster
Has anyone had a similar issue with Asterisk
Voicemail being unable to detect the digits sent from an SJ Phone connection. I
have included dtmfmode=inband and it works fine when calling other phones though
not with Voicemail. Voicemail doesn't regonise the password.
Is there a way to not se
Hi,
Whats involved in getting H323 working on Asterisk with Redhat 9???
Cheers,
Carl.
: Redhat v9
PC : AMD K2 512
Cheers,
Carl.
_
Store more e-mails with MSN Hotmail Extra Storage 4 plans to choose from!
http://click.atdmt.com/AVE/go/onm00200362ave/direct/01/
___
Asterisk
When I call Voicemail I get a very slow underwater sounding voice for the
first few seconds then it corrects itself. Any idea?
Output from Console:
-- Executing VoiceMailMain("SIP/2101-20db", "") in new stack
-- Playing 'vm-login' (language 'en')
Feb 26 14:45:58 NOTICE[393234]: sched.c:218 s
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wednesday 31 December 2003 03:24 pm, asterisk wrote:
> Here's the deal:
> It does almost anything. I can make it open my garage door. My
> installation records all conversations and then archives them as
> timestamped stereo MP3s. Our VB windows
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Thanks, to the two.
But I can't tell you how weary and sick I am of this kind of reflexive
criticism. It is -always- counter to true advancement, and burns those
delicate few who actually cause progress, in addition to (pitiable) n00bs.
I run a smal
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 29 December 2003 11:18 pm, Brian West wrote:
> All good here also.. this has got to be in the to 10 stupidest things
> posted to the mailing list today.
>
> bkw
Very nice Brian, thanks for the adversarial 'welcome'.
Is this the kind of trea
Tried to DL using CVS this eve, and it says:
Unknown host cvs.digium.com.
Has Asterisk development stopped?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
in general, so I could be wrong.
Carl
ProvoCityPower wrote:
>Did DVD players have to accommodate VHS tapes? Did VHS players have to
>accept beta?
>Why does VoIP have to deal with an accent protocol that can't handle
>lossy audio, nor irregular delays?
>Also why should we be so
1 - 100 of 180 matches
Mail list logo
