Sure, no worries. Will try that. What about advice on TTS setup. Would you
have any notes on how best to setup high-volume TTS environment, like maybe
a cluster of TTS servers and how Asterisk talks to those? Recommendations on
how to set that up? I'm thinking about trying Festival/FLite and maybe
asterisk by default listen on port 5060.You simply need open the file
/etc/asterisk/sip.conf and change these.>>
udpbindaddr=0.0.0.0:6080>>tcpbindaddr=0.0.0.0:6080save the file and open
asterisk console and execute "sip reload".
Muhammad Faheem
--- On Fri, 11/12/10, Baha @ SH wrote:
From:
Well,
I use many tts products because i work with diferents telphone
systems. Now for asterisk the best way for free is Festival and noon
free is Loquendo.
I'm not have notes to install debian on Sparc, i just only use debian
readme :-) It's too easy, debian work for you :D
Just download spa
Hi Luis,
Thanks for your comments. How / Why are you using that many TTS products? Do
you have a preference of one over the other?
Also, do you have any documentation / install/configuration notes that you
might be willing to share re: your experience with Debian on Sparc and the
TTS configuratio
On Thu, 11 Nov 2010 20:08:09 -0600, Russ Meyerriecks wrote
> On 11/11/10 7:23 PM, Carlos Chavez wrote:
>
> > I seem to be having the same problem with a new server. I am using a
> > TE220 with a VPM450 module. Using Dahdi 2.4.0 and Asterisk 1.6.2.13 on
> > a Dell server. All calls to the ou
I use Nuance, festival, Ibm tts and Loquendo.
Now in your case, i suggest use tts on the recommend tts
environment. Solaris is not standart system for tts products. Then you
can plug tts system into asterisk platform.
I use Debian for sparc and work excelent!! don't discard this option
may be
No, I want to use Solaris 10 on the Sparc platform. I've read a lot of
reports and tests/benchmarks conducted that sow Solaris 10 actually
performing better than all other Linux based Distros...not sure if that's
been the experience of others in the group.
I really want to know if someone has a hi
You try install debian in your sparc platform ?
On Thu, Nov 11, 2010 at 8:52 PM, RR wrote:
> Hello Group,
> I have been going through all the chit-chat about TTS and the various
> engines available to integrate with Asterisk incl. flite/festival, espeak,
> Nuance etc but I am wondering if anyon
On 11/11/10 7:23 PM, Carlos Chavez wrote:
> I seem to be having the same problem with a new server. I am using a
> TE220 with a VPM450 module. Using Dahdi 2.4.0 and Asterisk 1.6.2.13 on
> a Dell server. All calls to the outside have bad voice quality (echo
> and distortion). Internal cal
On Thu, 2010-11-11 at 18:28 -0600, Russ Meyerriecks wrote:
> On 11/11/10 5:44 PM, Jeff LaCoursiere wrote:
> >
> >
> > On Thu, 11 Nov 2010, Russ Meyerriecks wrote:
> >
> >>> On Tue, 9 Nov 2010, Daniel Tryba wrote:
> >>
> >>> I am curious about the tool "dahdi_maint"... what do the various
> >>> acro
Hello Group,
I have been going through all the chit-chat about TTS and the various
engines available to integrate with Asterisk incl. flite/festival, espeak,
Nuance etc but I am wondering if anyone's tried any or all of these to
compile on a Sparc based Solaris platform? If not, then what is the b
On 11/11/10 5:44 PM, Jeff LaCoursiere wrote:
>
>
> On Thu, 11 Nov 2010, Russ Meyerriecks wrote:
>
>>> On Tue, 9 Nov 2010, Daniel Tryba wrote:
>>
>>> I am curious about the tool "dahdi_maint"... what do the various
>>> acronyms stand for?
>> Yea there seemed to be a bit of confusion here as well so
On Thu, 11 Nov 2010, Russ Meyerriecks wrote:
>> On Tue, 9 Nov 2010, Daniel Tryba wrote:
>
>> I am curious about the tool "dahdi_maint"... what do the various
>> acronyms stand for?
> Yea there seemed to be a bit of confusion here as well so I patched
> trunk with some more descriptive error coun
> On Tue, 9 Nov 2010, Daniel Tryba wrote:
> I am curious about the tool "dahdi_maint"... what do the various
> acronyms stand for?
Yea there seemed to be a bit of confusion here as well so I patched
trunk with some more descriptive error counter labels :O)
>> FEC : 0:
Framing Errors
>> CEC : 0:
On 11/11/2010 04:48 PM, Marek Soha wrote:
> Uf...
>
> you are perfectly clean about that confusion...
>
> Only thing I want to do, is to route "stream" out of local asterisk - to
> connect final extension directly to "sender" - provider.
> So what I can do if I need i.e:
>
> 1) canreinvite=yes A
Uf...
you are perfectly clean about that confusion...
Only thing I want to do, is to route "stream" out of local asterisk - to
connect final extension directly to "sender" - provider.
So what I can do if I need i.e:
1) canreinvite=yes AND send T38 faxes through the same trunk?
2) maybe canreinv
Does it matter?
Phones are working correctly...I tried also portforwarding.
So corrected topology:
provider -> A (asterisk 1.6) -> B (asterisk 1.6) -> extension ->
-> NAT/FIREWALL -> (software fax, gateway whatever).
Software fax ends with DIS sent, 9600Bbps
Joel, dňa 11. novembra 2010 ste napís
On 11/11/2010 04:21 PM, Marek Soha wrote:
> Hi all.
>
> I have an issue with T.38 and re-invites.
>
> Topology:
> provider -> A (asterisk 1.6) -> B (asterisk 1.6) -> extension ->
> -> (software fax, gateway whatever).
>
> When between A and B trunk is canreinvite=no everything is working
> smoot
NAT? Firewall?
On Thu, Nov 11, 2010 at 3:21 PM, Marek Soha wrote:
> Hi all.
>
> I have an issue with T.38 and re-invites.
>
> Topology:
> provider -> A (asterisk 1.6) -> B (asterisk 1.6) -> extension ->
> -> (software fax, gateway whatever).
>
> When between A and B trunk is canreinvite=no every
Hi all.
I have an issue with T.38 and re-invites.
Topology:
provider -> A (asterisk 1.6) -> B (asterisk 1.6) -> extension ->
-> (software fax, gateway whatever).
When between A and B trunk is canreinvite=no everything is working
smooth. When I switch canreinvite to yes, it stop working.
Do you
On Tue, 9 Nov 2010, Daniel Tryba wrote:
> On Mon, Nov 08, 2010 at 02:44:26PM -0500, Jeff LaCoursiere wrote:
>>> It could be the echo canceller, I had this kind of problem with OSLEC. I
>>> also thought the PRI provider was sending clipped audio. I switched to
>>> the VPM450 daughterboard and sinc
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Watkins,
Bradley
Sent: Thursday, November 11, 2010 10:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VoiceMail cust
The Asterisk Development Team has announced the release of Asterisk
1.6.2.14. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.14 resolves several issues reported by the
community and would have not been possib
The Asterisk Development Team has announced the release of Asterisk
1.4.37. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.37 resolves several issues reported by the
community and would have not been possible wi
I did some more tests, and it's not really a problem with linphone: the
rtp capture shows empty packets sent by Asterisk.
Since the channel which is doing Playback() is in a MeetMe conference, I
tried also to speak on another phone on the same conference: well the
rtp capture shows the stream fr
>-Original Message-
>From: asterisk-users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
>Benoit Panizzon
>Sent: Thursday, November 11, 2010 11:29 AM
>To: asterisk-users@lists.digium.com
>Subject: [asterisk-users] VoiceMail customizing
>
>Hello
Hello
We would like to customize the voicemail menues.
So the intro should not be played if some user has recorded an own greeting
message and we would also like to remove some options from the menue.
Is this all hardcoded or is it somehow possible to redefine the voice menues
and the order ho
On 11/09/2010 03:20 PM, Gareth Blades wrote:
Jonas Kellens wrote:
On 11/09/2010 02:12 PM, Gareth Blades wrote:
Jonas Kellens wrote:
On 11/08/2010 09:50 PM, Jonas Kellens wrote:
Hello,
SIP DNS SRV records are not working.
My Grandstream uses the SRV records to f
Paulo Santos wrote:
> Hello,
>
> Following my first mail about this issue [1], I think I know now what
> the problem is.
>
> When I have both lines being used and a third call comes in, the person
> calling doesn't get a busy tone, he gets something like line unavailable.
>
> I've been debugging
Hi,
I dial on A* from a linphonec to a Playback() extension, then suddenly
the sound stops after a while, without any notice.
I enabled debug both in linphone and A*, and the RTP packets are sent
from A* and received from linphone. It doesn't matter whether I choose
alaw, ulaw, gsm as codec (bes
Hello
How can I run the sip service on asterisk on another port beside 5080?
I mean asterisk will still take sip requests on port:5080 and another custom
port, lets say port:6080
Thanks for any help
--
_
-- Bandwidth an
Found the problem already :
Dial(SIP/test6,,L(11000,5000,5000))
Correct syntax is :
Dial(SIP/test6,,L(11000:5000:5000))
semicolon...
Jonas.
On 11/11/2010 10:43 AM, Thorsten Göllner wrote:
Take a look at /var/log/asterisk/main or full /if enabled. Perhaps
there is a file not found. try:
e
On Thu, Nov 11, 2010 at 3:43 AM, Thorsten Göllner wrote:
> Take a look at /var/log/asterisk/main or full /if enabled. Perhaps there is
> a file not found. try:
>
> exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
> exten => _367,n,Set(LIMIT_WARNING_FILE=/path_to_your_audiofiles/file) # do
> not ad
Take a look at /var/log/asterisk/main or full /if enabled. Perhaps
there is a file not found. try:
exten =>
_367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
exten =>
_367,n,Set(LIMIT_WARNING_FILE=/path_to_your_audiofiles/file)
# do not add any extension!
Hello,
Limiting the call duration with the L-option of the Dial()-command is
working fine, however the announcement is not played.
Dialplan :
exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000))
The call lasts for 11 seconds, but 5 minutes befo
Hello,
All i have one issue regarding caller id, once i received a call from my SIP
provider it always set caller id with append 1 into
original callerID if a call from USA then there is no problem , but if i
receive a call from other country like INDIA i have also
found callerID part as 191 w
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