Victor Moreno wrote:
> Hi,
> voicemail are working ok, I receive message as attach via email.
> My question is :
> how can the user call asterisk and listen to his voicemessages ?
Set up a exten to voicemailmain passing the calling exten as the argument.
e.g.
exten => 121,1,VoiceMailMain(u${ex
Thank you Mr.Martin Joseph.Martin Joseph <[EMAIL PROTECTED]> wrote: On Jun 12, 2006, at 10:04 PM, Crazy Boy wrote:> Hi Friend,>> I heard about this word "echo" very much. Can you please tell me what > is this "Echo"?>Echo is when you say something and then hear it bounce back to you some brief time
Hi,
I'm still a newbie, but try to help you,
my voicemail works ok, I can also record messages ok.
My extension part is:
exten => s,1,Background(welcome-cisl)
exten => 1,1,Dial(Sip/vmoreno,10)
exten => 1,2,Voicemail(victor)
exten => 2,1,Dial(Sip/juliansip,10)
exten => 2,2,Voicemail(aajulian)
ex
Hi,
voicemail are working ok, I receive message as attach via email.
My question is :
how can the user call asterisk and listen to his voicemessages ?
thanks
Victor
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing li
Hello,
I have setup an Asterisk 1.2.7.1 system, with a working voicemail box:
/etc/asterisk/extensions.conf
exten => 83086921,1,Answer
exten => 83086921,2,Dial(SIP/stefan,5,r)
exten => 83086921,3,VoiceMail,u111
exten => 83086921,4,Hangup
exten => 83086921,1
Hi Koen Van Impe
Thanks for the meridian config and asterisk. I will
defenitly try them
And let every one know.
Just a few words and correct me if I am wrong
There are two things
1
E1 : the 32 channels once both th
On Jun 12, 2006, at 10:04 PM, Crazy Boy wrote:
Hi Friend,
I heard about this word "echo" very much. Can you please tell me what
is this "Echo"?
Echo is when you say something and then hear it bounce back to you some
brief time later...
This can be caused by many things, but the most comm
SIP is a UDP protocol, and telnet is TCP. You can't test it like that.
Have you tried connecting with a SIP client?
Peter
On 13/06/06, John Klimek <[EMAIL PROTECTED]> wrote:
I'm trying to setup Asterisk on my Linksys WRT54G router and it
appears to startup successfully (no errors) and it says
Hi,
shadowym wrote:
I am looking at ways to harden my asterisk install to prevent computer
related issues from happening. I am concerned about about disk write cache.
That seems to be a major source of hard drive corruption on power failure.
Hard Drive corruption is simply unacceptable for the
Hi Friend,I heard about this word "echo" very much. Can you please tell me what is this "Echo"?Thanks&Regards,Chandra __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
> Hi list!
>
> I have a Centos 4.3 box running Asterisk 1.2.9.1 with FreePBX 2.0.1
>
> I noticed that this setup is keeping a full asterisk log which, after
1
> month in production, has already grown to 1300 Mb in size. This is the
log
> location : /var/log/asterisk/full
>
> Why is this on by de
Not trying to be rude, but you will either need to invest many, many
hours learning how asterisk works and evaluating 3rd party billing
solutions, or possibly writing your own.
This will require light programming skills (agi, mysql, perl, etc), but
probably not C unless you really want to customiz
Hi list!
I have a Centos 4.3 box running Asterisk 1.2.9.1 with FreePBX 2.0.1
I noticed that this setup is keeping a full asterisk log which, after 1
month in production, has already grown to 1300 Mb in size. This is the log
location : /var/log/asterisk/full
Why is this on by default (I thoug
> There is a spec for echo cancellation on PSTN called g.168. I believe
> it's a
> suite of tests which put the echo canceller through its paces and if
you
> pass
> them you are certified to conform to g.168. None of the echo
cancellers in
> zaptel conform to this, whereas the Octasic, Tellabs an
Thanks again.
I found that there is a realtime load and realtime update in CLI. In
dial plan, I can use realtime() to load the value from table.
However, I am confused to use realtime update in dial plan. How to
implement realtime update in dial plan?
On 6/13/06, Damon Estep <[EMAIL PROTECTED]>
Your server is more than enough for 24 SIP users. Depends a bit on usage
patterns, though, you should be fine.
Erick Perez wrote:
I have this server I need to put to work.
The option I have is to make it work as a small office PBX with SIP
users and a Digium E1 Card for PSTN service.
24 SIP us
There is not a formula, but I second the opinion that the config is
adequate if the linux build behaves on the hardware (correct drivers,
config, etc).
The limiting factor is the E1, you will be able to handle a full E1 of
traffic, with transcoding, with this box.
This is not based on a formula,
BJ, when you say it is more than adequate, what do you do to calculate?
there *must* be a way to at least tell if the motherboardboard/cpu
will achieve results.
I just don't want to install it and then after a 5th user going to
call someone the asterisk begin to crash due to lack of resuources.
So is the problem with your audiocodes or with the asterisk system?
if it is with the asterisk, what kind of calls are you trying route to
your box? SIP/IAX/other?
On 6/12/06, Mahilal Silva <[EMAIL PROTECTED]> wrote:
Hi All
I have been able to get MP 104 FXO to make outbound calls with my aster
Sox will do it, the syntax is a little tricky and I am not an expert
with sox.
Also, check to see if you are using "quietmp3" or the current equivalent
in your moh config file.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael
I ripped a rock-and-roll CD for a client's moh. But it's too loud. Is
there a simple way to reduce the gain without having to remix the tracks?
Thanks
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
_
Try "show application realtime" at the CLI
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of unplug
> Sent: Monday, June 12, 2006 9:00 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] get
Thanks!
Do you mean there is a realtime function available to get and set the
value in table? Can you give me some references (website) as I have
found nothing of this function.
On 6/13/06, Damon Estep <[EMAIL PROTECTED]> wrote:
The question has changed, but the answer has not
> Search the wik
I have ekiga registering to a voip provider (skypho) and receiving
external call
through the stun server.
I want to redirect inconditionally all these calls to my asterisk
server, but I can't understand how and what should I configure in
asterisk in order to accept the redirected call.
In asteris
I'm trying to setup Asterisk on my Linksys WRT54G router and it
appears to startup successfully (no errors) and it says it is
listening on 0.0.0.0 port 5060, but I am unable to connect to it.
I've tried "telnet localhost 5060" but it just says connection
refused. I've also tried connecting from a
The question has changed, but the answer has not
> Search the wiki for the application command realtime() if you are
> using realtime.
>
> www.voip-info.org
Accountcode is a channel variable that can be read at any time, if you
are trying to get information from the DB that is not held in a chan
shadowym wrote:
Any other recommendations/links for increasing the reliability of Asterisk
servers?
Separate the various use cases of the filesystem into different volumes
with LVM. The parts that are not written to except during upgrades like
/usr should be mounted read-only, and the various
Hi All
I have been able to get MP 104 FXO to make outbound calls with my asterisk box and polycom IP 500 phone.
However I cannot get the incoming calls to hit the asterisk box.
Any help will be appreciated.
Thanks,
Lal
___
--Bandwidth and Colocation pro
Thanks. I have set the ARA to using DB without problem. From the
example, I can get the value of account code from sip_buddies using
the following.
exten=>100,1,Answer()
exten=>100,2,NoOp(${ACCOUNTCODE})
I can't get the value of the field 'cancallforward' using the following.
exten=>101,1,Answer
The most significant change in the "02" versions of the SPA line is more
memory to handle larger firmware images.
They do not use the same firmware as the non "02" models and will reject
the older images.
As the firmware image evolves it gets larger, and the previous model
will end up being limit
We battled this same issue for a couple weeks, at about 30-50
simultaneous recordings the audio would get all screwy.
I looked at that solution but opted for something a little more
passive. I use orkaudio to sniff rtp streams and mux them. I have it
to perfect quality, the same as the mon
Steve Davies wrote:
On 6/12/06, Doug Crompton <[EMAIL PROTECTED]> wrote:
It seems that any firmware is usable on any hardware as my hardware is
2.x. I wonder if 3102 firmware could be used on the 3000. Is the size
the
same? I guess you would have to be willing to make a brick to find out!
Raid card with an onboard battery backup.
PaulH
--
Paul Hales
Technical Manager
AsteriskIT
www.asteriskit.com.au
bus: 03 8320 8100
mob: 0434 673 529
shadowym wrote:
I am looking at ways to harden my asterisk install to prevent computer
related issues from happening. I am concerned about
Or any polycom phone that has speakerphone like the IP501 and IP430.
Time Bandit wrote:
Can you, or anyone else comment on the speakerphone ability of the
GVX-3000
? We run the GXP-2000's and for the most part are happy with them,
but for
upper management we're looking at phones with better s
Christian Stredicke wrote:
> If you ping on the SIP port the message has to go through the
> application layer - which takes some time considering it is an embedded
> system with a small CPU. That part should be ok.
>
> It the phone becomes choppy, that problem is probably related to the RTP
> si
With the help of the Zulty's support, I was able to get the WIP2 working
properly.
The trick was:
progressinband = yes
And that was it.
later,
Paul Hales
--
Paul Hales
Technical Manager
AsteriskIT
www.asteriskit.com.au
bus: 03 8320 8100
mob: 0434 673 529
_
I put reinvite=yes in my sip.conf.
For starters, it's canreinvite=yes. Then do a "sip show peer" on the
peer and make sure it says that it can reinvite.
Reasons why no reintives are even attempted include the transfer flag
in the dial application and if the channel is monitor-ed (for obvious
rea
On Monday 12 June 2006 17:55, shadowym wrote:
> Believe me, you can drive yourself insane trying to come up with some
> magical formula that JUST works because it usually won't happen that way.
> Software echo cancellers are simply not good enough for many situations.
Actually that's untrue. I th
On 6/12/06, Roger Schreiter <[EMAIL PROTECTED]> wrote:
Hi,
I put reinvite=yes in my sip.conf.
For testing, I restricted the codecs to alaw.
I have no modifiers in my dial command.
Thus, there should be no reason not to reinvite.
Call (sip, authenticated) comes in and is forward
via SIP (not au
Well, it wasn't so much of a command line ping on the SIP port, but the
times reported under Status when qualify is set to yes.
It should be far less time than that as the time from that PBX up to my
server out on the public Internet is only 10 ms away. I have servers in
other parts of the co
I am looking at ways to harden my asterisk install to prevent computer
related issues from happening. I am concerned about about disk write cache.
That seems to be a major source of hard drive corruption on power failure.
Hard Drive corruption is simply unacceptable for the 99.999% uptime
requir
I'm slightly confused about how SIP security and authorization works.
I've looked at the Wiki (http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer)
, but it's, um, flawed:
> As of Asterisk 1.2, there is no reason to actually use 'user'
entries
> any more at all; you can us
I just got my 1st batch of GXV3000's. I can attest that the speakerphone is
every bit as bad as the GXP2000, perhaps even a little worse. Nowhere near as
good as Cisco. The other phone I personally found to be good for speakerphone
use us SNOM.
On Wednesday 31 May 2006 11:53, Paul C wrote:
> C
I'm experimenting with Dock-n-Talk. It can connect to several different phones and has an analog line connection to it. There is quite a bit of delay between the time the call is passed to it and you hear ringing.
I'm using it with a Motorola E815 on Verizon. Couldn't comment on the sound q
Go to one of the numerous voip distribor sites and browse their
selections, there are several
Yes you can plug into you wiring, just don't forget to disconnect
your local telco line - bad things would happen
On Jun 12, 2006, at 5:19 PM, John Klimek wrote:
Ahh, thanks! That's what I
Hi,
I put reinvite=yes in my sip.conf.
For testing, I restricted the codecs to alaw.
I have no modifiers in my dial command.
Thus, there should be no reason not to reinvite.
Call (sip, authenticated) comes in and is forward
via SIP (not authenticated) to another asterisk box.
Unfortunately, med
Title: ICLID or CNAM calling name and number through a cisco isdn gateway
All,
I need to run this by everyone and see if someone has any idea's. I have a asterisk server setup and currently am receiving the inbound calling number where the name should be. My setup is
One pri te
Pete,
I have it all running ok now with the exception of music on hold, for some
reason Asterisk stops MOH on the Vega trunks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Doyle
Sent: Monday, June 12, 2006 7:21 PM
To: Asterisk Users Mailing Lis
That could be an issue. Would recording to a ram drive solve the problem?Thanks.On 6/12/06, Steve Totaro <
[EMAIL PROTECTED]> wrote:Recording many simultaneous calls can cause this behavior too.
Thanks,Steve Totaro
___
--Bandwidth and Colocation provided
Issac,
I think the "destroying calls" part is coming from having the
registration fail. Instead of using [13] in sip.conf, try [08] (based
on what you just sent). THEN, in extensions.conf, make sure you can
handle extension 13. Extension 13 probably has to be in the same
context as your sip.conf
Thanks Damon, I have now turned off all echo cancelation on both the lines coming in for fax and the channels to the channel bank that have fax modems/machines. The fax is clearer, but not as clear as it was prior to *. I have an error showing up of WARNING[2465]: chan_zap.c:3925 zt_handle_event:
Ahh, thanks! That's what I thought but I wasn't sure because I
thought ATA boxes were only for specific VOIP providers.
Which ATA with an FXS port would you recommend for around (or under) $50?
Also, can I simple plug the ATA into an existing RJ-11 jack so that
ALL of the phone jacks in my hous
Recording many simultaneous calls can cause this behavior too.
Thanks,
Steve Totaro
Gary Richardson wrote:
We're not using any zaptel hardware though. I didn't think the echo
cancellers would be doing anything? We're digital and sip from end to
end. Do I need to disable echo cancellation in so
On 6/12/06, Erick Perez <[EMAIL PROTECTED]> wrote:
I have this server I need to put to work.
The option I have is to make it work as a small office PBX with SIP
users and a Digium E1 Card for PSTN service.
24 SIP users and one E1 card in an Intel 945board (533 Front side bus)
with 1GB DDR 533mh
In the case of the Sangoma, A200D, it adds about $300US. Does not matter if
you have 2FXO ports or 24. That's probably a significant hit for a home
user with only a couple lines. If a business cannot afford that extra cost
then that is a good indication that an Asterisk system may no be a good f
I have this server I need to put to work.
The option I have is to make it work as a small office PBX with SIP
users and a Digium E1 Card for PSTN service.
24 SIP users and one E1 card in an Intel 945board (533 Front side bus)
with 1GB DDR 533mhz of ram, one Pentium Dual Core 2.66 ghz (FSB
533MHZ
This ALL pretty much depends on YOUR particular install because the loop
length and signal level to your local exchange will vary from site to site.
Believe me, you can drive yourself insane trying to come up with some
magical formula that JUST works because it usually won't happen that way.
Soft
Hello,
We have an office [EMAIL PROTECTED] system which vwe ran through a
Dlink DLS router. The Dlink has developed a fault and we are now trying to use a
3com router.
With the Dlink the port forwarding was simple and worked. We routed 5004-5082
and 1-2 to the asterisk box. Things w
This seems to be only for prepaid calling cards. Is there something that can also handle prepaid and postpaid multi-tenant SIP phones?- DanielOn Jun 12, 2006, at 5:00 PM, William Piper wrote:www.asterisk2billing.org On 6/12/06, Wasif <[EMAIL PROTECTED]> wrote: Hi,I need to use Asterisk as a switc
www.asterisk2billing.org
On 6/12/06, Wasif <[EMAIL PROTECTED]> wrote:
Hi,I need to use Asterisk as a switch which can handle wholesale traffic withbilling. Please advice me how I can I implement this.
Thanks___--Bandwidth and Colocation provided by Easy
Min Qiu wrote:
> Arh... I did experence sound quality issues and I pointed
> my finger toward my VoIP provider;-) Good to know. Can
> you pass a pointer to where I can get 1.50?
>
> Thanks a lot,
>
> Min
>
Atcom support will send you one by email, or I can email you a copy.
__
Hi,
I need to use Asterisk as a switch which can handle wholesale traffic with
billing. Please advice me how I can I implement this.
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or updat
Analogue Telephone Adapter(s)
Linksys/Sipura range SPA-3102; SPA2100; SPA1001 to name but 3
On 12/06/06, John Klimek <[EMAIL PROTECTED]> wrote:
Ok, I've done some more research and I don't think I want an FXO box...
What I'd like to do is use BroadVoice (with their BYOD plan) and then
run Aster
add an ATA with an fxs port or 2
On Jun 12, 2006, at 2:39 PM, John Klimek wrote:
Ok, I've done some more research and I don't think I want an FXO
box...
What I'd like to do is use BroadVoice (with their BYOD plan) and then
run Asterisk on my WRT54G router. I'd also like to use my regular ho
If you ping on the SIP port the message has to go through
the application layer - which takes some time considering it is an embedded
system with a small CPU. That part should be ok.
It the phone becomes choppy, that problem is probably
related to the RTP side. Maybe you have different pack
NAT and STUN are supported on the SPA-941. G.711, G.726, G.729A and G.723.1
are also supported.
Cory Andrews
Executive Vice President
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice - 800.398.VoIP X3402
fax - 716.630.1548
e - [EMAIL PROT
any body know this phone? support NAT? and standart codecs of asterisk ?
--
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/aste
Ok, I've done some more research and I don't think I want an FXO box...
What I'd like to do is use BroadVoice (with their BYOD plan) and then
run Asterisk on my WRT54G router. I'd also like to use my regular home
phones without having to use a special "SIP" phone... (eg. I like my
Vtech normal co
Eric "ManxPower" Wieling wrote:
There is a standalone app included with Cepstral. I think it's called
"swift".
Doug Lytle wrote:
I've also used the swift application to convert whole documents with
good results.
Doug
Correct.
--
Ben Franklin quote:
"Those who would give up Essenti
There is a standalone app included with Cepstral. I think it's called
"swift".
Doug Lytle wrote:
Doug Crompton wrote:
Ok Thanks. I just registered 'Diane' also. She seemed to have the best
voice. I am curious if you added the Cepstral app or used the festival
method described in the Cepstral
Stephen Bosch wrote:
I
thought these were just voices for use with third party applications...
It is a 3rd party app, but you can find the link to it by searching the
wiki for app_cepstral.
The installation instructions can be found here:
http://www.oldskoolphreak.com/tfiles/voip/install
Doug Crompton wrote:
Ok Thanks. I just registered 'Diane' also. She seemed to have the best
voice. I am curious if you added the Cepstral app or used the festival
method described in the Cepstral FAQ. I recompiled with Cepstral app and
saw that later. App seems to work fine here.
I'm playing
Hmn. Very nice! It works!
On the matter of timing --
Asterisk appears to wait two full PSTN rings before it dials the SIP
extension. Is there any way we can tighten up this interval? Is that
done in the Zap configuration? The driver? The dialplan?
Asterisk is waiting for the CallerID, which is
Stephen Bosch wrote:
Hi, folks:
Okay, so here's an idea.
I have a TDM-400 card with an FXO card in it connected to the PSTN and a
Polycom IP 501 phone.
Observe the following simple dialplan for illustration:
[incoming]
; incoming calls from the FXO port are directed to this context from zapa
Hi all,
Your help would be greatly appreciated, I have been struggling for days with inst / config of TDM01B.
I have installed TDM01B.
Using Asterisk 1.2.9.1 on RH Ent 4.00
cat /proc/interrupts shows card wctdm on int 10.
Green LED is on (module is in port 4).
I have my /etc/zaptel.conf
Walid Azab wrote:
> Hi everyone,
>
> Is there a way to get Asterisk read from MySQL using Festival Text to
> speech engine?!!
>
> Thanks
>
>
>
>___
>--Bandwidth and Colocation provid
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Thomas Kenyon
> Sent: Monday, June 12, 2006 2:06 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] use AT320 international call
>
>
> Min Qiu wrote:
I don't know everything that's going on as someone
else has been working on the project, but it hasn't really been going anywhere,
so I had some questions.
We've got some Snom 320s with Asterisk 1.2.9.1 (I
believe). All was well (with a previous release), but the phones started
to get rea
Is anybody using http://freevoip.gedameurope.com ?
I've a problem with dial out.
-from help menu---
In the help setup menu there is an example :
exten => _683X,1,Answer()
exten => _683X,2,Dial(IAX2/6840369/${EXTEN})
The second allows you to call FreeVoIP members by dialing 683
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Monday, June 12, 2006 2:09 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] use AT320 international call
>
>
> On 6/12/2006, "Min Qiu" <[EMAIL PR
Hi,On 6/12/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Hello,I have to setup a IP/SS7 gateway on a Sun Ultra 20Debian Sarge for AMD64
Can we compile asterisk on AMD64 processor ?Yes, you can. Asterisk works fine on AMD64 (at least in my experience).regards,jmacz
Harry
Hi Kanishka,Several months ago I had a server running * and zaptel 1.0.9 over a Sun Fire V20z with Slackware compiled for 64 bits, and I have absolutely no complaints about it: lower CPU consumption than with 32bits and a good voice quality.
The server received 4 E1s through a pair of TE21OP wildc
Hello John ,
What about debian sarge ?
Harry
--- John Millican <[EMAIL PROTECTED]> a écrit :
> I have 2 servers currently running 64 bit SuSE 10.x
> on AMD Opteron processors
> both of which are working very well.
>
> John Millican
> Senior Partner
> Director of Technology
> Sentinel Communicat
Doug Crompton wrote:
> Ok Thanks. I just registered 'Diane' also. She seemed to have the best
> voice. I am curious if you added the Cepstral app or used the festival
> method described in the Cepstral FAQ. I recompiled with Cepstral app and
> saw that later. App seems to work fine here.
Looking a
Post this to asterisk-biz, NOT asterisk-users
/M
--
"Those that sacrifice essential liberty to obtain a little temporary
safety deserve neither liberty nor safety." -- Ben Franklin (1759)
___
--Bandwidth and Colocation provided by Easynews.com --
A
Time Bandit wrote:
>> > [incoming]
>> > ; incoming calls from the FXO port are directed to this context from
>> zapata.conf
>> >
>> > exten => s,1,Answer()
>> > exten => s,2,Dial(SIP/polycom)
>
> Try this
>
> exten => s,1,Dial(SIP/polycom,20)
> exten => s,2,Hangup()
>
> I think this way, * won't
What about using a search engine like google before ...
http://www.google.se/search?client=firefox-a&rls=org.mozilla%3Aen-GB%3Aofficial_s&hl=sv&q=nokia+e60+asterisk&meta=lr%3Dlang_en%7Clang_no%7Clang_sv&btnG=Google-s%C3%B6kning
and you will find several examples that is working for you!
/Mats
On 6/12/2006, "Min Qiu" <[EMAIL PROTECTED]> wrote:
>Hi all,
>
>The firmware I used is pa168s_iax2_us_151011.bin.
>
>My problem is the handset dial before I finished key in all
>the numbers, no matter how fast I managed to press the keys.
>It appeared it always dialed immediately, for example "01
Min Qiu wrote:
> Hi all,
>
> The firmware I used is pa168s_iax2_us_151011.bin.
>
> My problem is the handset dial before I finished key in all
> the numbers, no matter how fast I managed to press the keys.
> It appeared it always dialed immediately, for example "011862",
> when I actually ment
Hi everyone,
Is there a way to get Asterisk read from MySQL
using Festival Text to speech engine?!!
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.
Turn off echo can for those calls.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Monday, June 12, 2006 10:55
AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] TDM Fax
Problems
I am running into er
Search the wiki for the application command realtime() if you are using
realtime.
www.voip-info.org
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of unplug
> Sent: Monday, June 12, 2006 10:47 AM
> To: Asterisk Users Mailing List - Non
I have 2 servers currently running 64 bit SuSE 10.x on AMD Opteron processors
both of which are working very well.
John Millican
Senior Partner
Director of Technology
Sentinel Communications
PO Box 9
Wentworth, NH 03282
On Monday June 12 2006 1:39 pm, Kanishka Somaratne wrote:
> Hey
> Does aster
> [incoming]
> ; incoming calls from the FXO port are directed to this context from
zapata.conf
>
> exten => s,1,Answer()
> exten => s,2,Dial(SIP/polycom)
Try this
exten => s,1,Dial(SIP/polycom,20)
exten => s,2,Hangup()
I think this way, * won't answer the line until your SIP phone
answers. If
Colin Anderson wrote:
> no from the Asterisk perspective it will work regardless of the number
> dialled as long as it matches the 55512XX pattern. As others have pointed
> out though, it's just easier to have a single DID and your provider allow
> multiple channels or instances of the same number
Hi all,
The firmware I used is pa168s_iax2_us_151011.bin.
My problem is the handset dial before I finished key in all
the numbers, no matter how fast I managed to press the keys.
It appeared it always dialed immediately, for example "011862",
when I actually ment to dial 0118620. Th
Colin Anderson wrote:
> the caller is out his money anyway when you call any phone and voicemail
> kicks in, although i think on a payphone they give you a 2 or 3 second
> window to hang up.
That assumes that you are routing to voicemail. That doesn't always apply.
Also -- the payphone behaviour
Hello,
I have to setup a IP/SS7 gateway on a Sun Ultra 20
Debian Sarge for AMD64
Can we compile asterisk on AMD64 processor ?
Harry
__
Do You Yahoo!?
En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible
contre les me
Hey
Does asterisk works well on an AMD 64 bit processor server.
are there any issues with this ?
Regards
Kani
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://list
Ok Thanks. I just registered 'Diane' also. She seemed to have the best
voice. I am curious if you added the Cepstral app or used the festival
method described in the Cepstral FAQ. I recompiled with Cepstral app and
saw that later. App seems to work fine here.
Doug
On Mon, 12 Jun 2006, Doug Lytle
1 - 100 of 188 matches
Mail list logo
