Hello All,
I'm planning to use Asterisk only for voicemail Application and Recording
will be done at different server.
When user changing his personal greeting or leaving voicemail Call need to
throw to external Voicemnail recording server over SIP til the time
recording complete.
While throwing
Hi
I want to can we use asterisk as TTS server. Which can support mrcpv2 and
ssml.
Im looking for tts server with above requirement will asterisk 1.8 is
useful for me. Any configuration available.
Any opensource tts available.
Amit--
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On Mar 13, 2013 10:16 PM, "Amit Salunkhe" wrote:
> Hi
>
> I want to know asterisk 1.8 as text to speech server.
>
> If we can use as TTS server then it support SSML.
>
> Any sample configuration available for this requirement. Plz help me with
> suppor
Hi All,
I'm using 1.8 Asterisk and i havet set DTMF mode=rfc2833 in SIP global
default settings.
but when user sending DTMf event with SIP info method my asterisk accepting
that DTMF. If default or global setting is rfc2833 then how come asterisk
accepting SIP info dtmf event? what to check pleas
Dear All
Can anyone let me know where i can free sound file whcih i can use for
system monitoring alrams.
Regards
Amit--
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Dear All
Can anyone let me know where i can free sound file whcih i can use for
system monitoring alrams.
Regards
Amit--
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Dear All
Among all the readers anybody have ever work on Granstream device GXE2504A
which act as ippbx and having GUI to configure and maintain.
We are facing one problem with this device, thsi device reply or adding
codec like ilbc,G.721 which is not supported by our Asterisk server or our
SBC.
Dear All
Can you let me know is this possible to if we are using Asterisk version 1.4
or 1.6 for incoming voicemail we can send as email in text formta. Means
voice mesage converted into text message & send it to resp. email ids. is
this possible.
If yes. we can do the same with help of Asterisk
Dear ALl
Can we use Asterisk for only for transcoding?. if yes how many concurent
call we can transcode with help of Astetrisk?
For this we only need to config SIP.conf or any other file too.
Thanks
Amit--
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Hi All
For recording inbound call we are using following line in dial
plan.But we wish to set file name which describe who attend the call or lets
say extension of the call attendant.
Current line in dial plan to set file name is like this-
*Set(MONITOR_FILENAME=${STRFTIME(${EPOC
Hi All
we are using Asterisk 1.4.21 users having analog phone connected
with Audio codes g/w Mp124.
How we can put caller on hold when we receive call on Analog phone
(Panasonic). Any dial plan application or feature.conf need to use for this.
Audio code g/w having option which we ca
Hi All
we are using Asterisk 1.6.0.9 version.try to use Minivm for
voicemail, but having following problem.
1.How any extension let's say 7001 can access his voicemail box from his
deskphone, any config or dial plan example is there. What kind of config
require in extension.conf,minivm.co
Hello,
I am using OpenSIPS to register all the users and planning to use asterisk
for Auto Attendant, Queues, Voicemail and Conference Bridge.
I have a scenario where the signaling does not happen properly:
1) A user from Opensips dials an extension 7000 which is an
auto-attendant extension
Hi
We are using Asterisk + java based Pd Dialer. Cisco 7040 IP Phone we are
using as extensions or Agent phones.
currntly we set NAT keep alive time less as possible & registartion time= 25
sec instaed of 3600. But following issues we are facing & im not sure whther
its due to internal netwokr
w. anybody can provide me config exmple?
I am using Asterisk 1.4.9. Plz help me
Regards
Amit Salunkhe
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Hi All
Can anybody help me for dial plan to barge or Spy(ExtenSpy)
specificor selective extemsions among 20 extension in my office.
lets say my office extension range is 301-320 & i want to barge only 3
extension say 320, 302,314.
is this possible to barge specific extension? . Plz he
Hi All
Can anybody help me for dial plan which can barge inbound call
groupwise.
Because when i am trying to barge inbound calls which is coming on my DID
number i can hear 1st 3 digit of my Inbound provider IP address instaed of
extension which pick that calls.
I tried Chanspy as well as
Hi All
I am trying to implement Inbound call Barging using ChanSpy &
ExtenSpy.Actual requirement is i want to spy or barge Inbound calls received
by specfic group or queue.
For that i use Set(SPYGROUP) concept but as its inbound calls it playing
Inbound channel(whcih is 1st 3 digit of inbo
Hi
i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling
client so i can make VOIP calls thru that phone. Aslo that Phone easly able
to register with Asterisk Pbx to recive inter-office calls.
i try to search from web & also from Nokia site but they only mention this
features a
HI
I is there any Application cmd which we can use for Asterisk Meetme
confrence for Adminstartor. so, Admin can Manupulate confrence like mute all
users & kick any user out of the confrence.
I tried with Meetme cmd but its not work for me. so is there any idea or
config details for this.
i
Hi
is there any Application cmd which we can use for Asterisk Meetme
confrence for Adminstartor. so, Admin can Manupulate confrence like mute
all
users & kick any user out of the confrence.
I tried with Meetme cmd but its not work for me. so is there any idea or
config details for
Hi All
I need to set my Asterisk conference such way that , during
confernce Admin Can kick 1 or all user , Same for mute fuction.As well as
Admin can increase or decrease conf & user volume.
for that i used MeetMeAdmin like this
exten
Hi
any body have idea about asterisk call waiting with SIP if we use
asterisk1.4.5 & hard phone with extensions.
i need dial plan logic for this which capable to activate & deactivate such
feature.
with ATA & IP phone it is possible but with normal hardphone & SIP in
asterisk is it possibl
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