Hello everyone. I have a client with specific requirement, here's the
scenario:
Call comes in
Ivr menu, press 1 for new record 2 for existing 3 for operation blabla..
on pressing 1, list of 5 categories A,B,C,D .. when customer selects a
category a 4 digit pin needs to be generated and a recording
Hello
Can anyone suggest sip phones with headset for use in call centers . They
should be fully inter operable with Asterisk over sip .
Thanks
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The destination numbers are valid in almost all cases . But i do think it
might be when someone is on call and on client side internet connection
goes off .. I am really not sure about this one but i just saw that maximum
such records are from one of my customer who has a very bad connection .
On
Strangely enough i ordered another number on their website yesterday hoping
that they might have filtered numbers which are to be disconnected from the
pool but after a hour of registration i get the same email for new DID i
bought . This is pathetic and frustrating now . I have sent an e-mail to
t
that either :P ) .
On 09/08/2007, Stephen Bosch <[EMAIL PROTECTED]> wrote:
>
> Mail list wrote:
> > Just got mail from them saying my NY DID will be deactivated in few days
> > . Funny thing is their site is still showing orderable DID's of same
> > area code . An
Just got mail from them saying my NY DID will be deactivated in few days .
Funny thing is their site is still showing orderable DID's of same area
code . Anybody else got this ?
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ast
I am looking for a retail DID provider which should provide unlimited
incoming calls something around 4-5 bucks . Nufone seemed like a good choice
at $5 but they are down again :( . Any suggestions please .
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2007/7/13, Noah Miller <[EMAIL PROTECTED]>:
> > > > I have a strange comportment of the MOH system on my asterisk.
> > > > When i respond to a call and after fews second i set this call in hold
> > > > mode the correspondent listen the music fine.
> > > > When i re-take my correspondent at T0 insta
2007/7/12, Noah Miller <[EMAIL PROTECTED]>:
> Hi -
>
> > I have a strange comportment of the MOH system on my asterisk.
> > When i respond to a call and after fews second i set this call in hold
> > mode the correspondent listen the music fine.
> > When i re-take my correspondent at T0 instant the
Hi list,
I have a strange comportment of the MOH system on my asterisk.
When i respond to a call and after fews second i set this call in hold
mode the correspondent listen the music fine.
When i re-take my correspondent at T0 instant the music is paused. And
when i re-hold him at T60 (60 second la
6I am using asterisk 1.4.5 and storing cdr in mysql . Here's example
of one cdr generated
2007-06-26 00:44:28 682345xxx 6823456xxx s
macro-dialout-trunk SIP/343684-09544f20 SIP/provider-0938de98
Dial SIP/provider/1386734|300| 28 5 60 1
0 ANSWERED
Hello
Anyone here uses Vitelity as voip provider ? Their pplans looks good but i
need some feedback from existing customers if any here .
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lol yeh all will miss you :D . . its like stopping to use internet if google
is down sometime .
On 15/03/07, Richard Lyman <[EMAIL PROTECTED]> wrote:
wrote:
> *snipped
> If I can't be confident enough in an important source of information
like
> this then I can't be confident enough to provide
Is this site good ? They ship on time ? Any reviews on their card .
Thx
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did u try modprobe zaptel first ? also check makefile if ztdummy is marked
for compilation or not
On 19/02/07, Chris Blunt <[EMAIL PROTECTED]> wrote:
Hi List,
I am having some trouble with installing the latest version of ztdummy on
a CentOS Kernel 2.6 system.
I have installed a few Aste
ng* <[EMAIL PROTECTED]> wrote:
Mail list wrote:
> Yes i read that on voip-info wiki but i have bindport = under device
> (extension) which should make that extension work on other port but its
> not working . :(
No, bindport= under the device section is ignored b
This went little bit offtopic :P but i dont mind . Here's how i solved it (
just if some one in future goes in this problem and end up here by googling
) .
Port 5060 was blocked by isp incoming as well as outgoing for few
extension's isp . On server i did
iptables -t nat -A PREROUTING -i eth0 -p
Is there any good iax2 softphone capable of attended transfer ( like sjphone
for sip ) . ? I tried iaxcomm and idefisk both seems unable to handle
attended transfers.
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Actually port block is on softphone side and not on asterisk server's
internet connection .I put this in iptables of asterisk server
iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j REDIRECT
--to-port 127.0.0.1:5060
server is listening on port 5060
Now strange part is everything s work
dport= and it must be in [general]
Mail list wrote:
> I am using a month old svn version of asterisk 1.2 . I have set
> bindport=5091 for a sip peer ( type = friend) and nat=yes .. in sip show
> peer it shows port 5091 for peer but asterisk isnt listening on port
> 5091 at all . I
Hello my isp has blocked outgoing and incoming connection for port 5060 . I
have ssh access to server so i want to send all traffic from port 5091 to
port 5060 of asterisk .so i tried
iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to
127.0.0.1:5060
Now my softphone is abl
I am using a month old svn version of asterisk 1.2 . I have set
bindport=5091 for a sip peer ( type = friend) and nat=yes .. in sip show
peer it shows port 5091 for peer but asterisk isnt listening on port 5091 at
all . I tried both port=5091 as well as binport=5091 but asterisk does not
listen
yum install subversion
On 09/12/06, Kovar Petr <[EMAIL PROTECTED]> wrote:
svn is application called "subversion", you should download and install
it first.
- Original Message -
*From:* Ed Nuñez <[EMAIL PROTECTED]>
*To:* Asterisk Users Mailing List - Non-Commercial
Discussion
*Sent:*
that site also has g729 codecs for asterisk but is it legal to use them ?? (
digium charges $10 each g729 channel )
On 08/12/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:
g723 codec isn't problem, you can obtain for all asterisk versions from:
http://kvin.lv/pub/Linux/Asterisk/
PJ
Jean-Michel H
is there anyone working on VXML or CCXML integration for asterisk?
I've integrated OpenVXI 3.4 (the latest one) with Asterisk for a
client. It is now in production, interpreting their VXML pages
using Asterisk for SIP/IAX telephony (but could use anything).
They don't require ASR for now, so I
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