Hi All,
I'm working on the following scenario:
VoIP Gateway -- Asterisk server -- Proxy server -- PSTN
|
XMLRPC
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Hi All,
I'm working on the following scenario:
VoIP Gateway -- Asterisk server -- Proxy server -- PSTN
| |
XMLRPC Radius
In this call flow a prepaid caller places a call over the VoIP gateway to
the
Asterisk realtime is what you are looking for. the subject is explained very
clearly including configuration examples and DB schema on the following
links:
http://www.voip-info.org/wiki-Asterisk+RealTime
http://www.asteriskdocs.org/modules/news/article.php?storyid=28
I won't go over the process
Asterisk supports it and the good news is that you don't have to do anything
for it to work.
On 5/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
We are currently trying to setup Asterisk with iBasis. One
question/problem we have is that Ibasis has told us to send the INVITEs to
one IP
to the INVITE IP using the Dial command and that's it?
Thanks
On Sat, May 19, 2007 12:29 pm, Yossi Ben Hagai [EMAIL PROTECTED] said:
Asterisk supports it and the good news is that you don't have to do
anything
for it to work.
On 5/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
We
Check rtpproxy from portone for media proxy and nat traversal.
http://www.voip-info.org/wiki/view/Portaone+rtpproxy
another option is the MediaProxy from AG projects:
http://www.voip-info.org/wiki-MediaProxy
Joss.
On 5/11/07, Jean-Marc Salsa [EMAIL PROTECTED] wrote:
Hi all,
I have been using
Regarding (2) - you can either provide a realtime query service supporting
web service interface which can be consumed using virtually any programming
language and it would be very easy to build an AGI script around it.
the second option would be to periodically update a flat file (csv) and
Hi Matt,
you didn't mention what type/bw of each site Internet connection, i suggest
that you try to split the scenario into smaller pieces:
- run long term pings between the server while you make a call and check for
packet loss.
- make internal calls between extensions on the same branch and
Hi Adriano,
I agree with Time Bandit - AGI is what you are looking for. I recently had a
similar scenario where I had the check the cid of every customer calling to
a support qeue and check the payment status against a windows CRM app. if
the customer has an unsettled debt the call is redirected
Check the Milliwatt() cmd here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Milliwatt
It sends 1000Hz, but you can derive from it.
Joss.
On 4/24/07, Jerry Geis [EMAIL PROTECTED] wrote:
Does asterisk have a way in the dialplan to generate tones?
Say I want to play a tone 300Hz
I second that. the PIX has SIP fixup which allows RTP traffic to pass
dynamically based on SDP information, so you don't need to create a rule for
the RTP range - just allow SIP UDP 5060.
On 4/25/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Apr 24, 2007 at 11:04:53PM -0400, Lee Jenkins
Looks okay to me. either the number you are testing with your VoIP provider
has an automated response which answers the call at the same sec you sent
the Invite request or the provider is sending False Answer Supervision...do
a sip debug and check while you make the call.
On 4/16/07, Adam KOSA
The Playback command is auto-answering the call. you can use
Playback(please_wait,noanswer) to fix it.
Joss.
On 4/16/07, Adam KOSA [EMAIL PROTECTED] wrote:
Hi, and thanks for the suggestions!
Matt wrote:
Sounds like your VoIP provider is incorrectly sending you an Answer
before
the call
It's possible, have the SIP clients use SRV records for server location and
use asterisk ARA to store SIP peers and extension.conf on DB. if the users
are not behind NAT it should work.
(open)SER is much better solution for high traffic / availability setups.
On 4/16/07, J. Oquendo [EMAIL
Here:
http://www.voip-info.org/wiki/view/Asterisk+log+queue_log
On 4/15/07, Rilawich Ango [EMAIL PROTECTED] wrote:
Where can I get the meaning of each field in queue_log?
On 4/15/07, Darryl Dunkin [EMAIL PROTECTED] wrote:
You will probably find what you are looking for here:
and populate the
variables?
Thanks
Bart
Yossi Ben Hagai wrote:
That's the correct syntax:
exten = s,1,Noop()
exten = s,n,MYSQL(Connect connid localhost root passw0rd dax)
exten = s,n,MYSQL(Query resultid ${connid} SELECT\ password\ FROM\
dnislookup\ WHERE\ dnis=${IVR-Exten})
exten = s,n,MYSQL
That's the correct syntax:
exten = s,1,Noop()
exten = s,n,MYSQL(Connect connid localhost root passw0rd dax)
exten = s,n,MYSQL(Query resultid ${connid} SELECT\ password\ FROM\
dnislookup\ WHERE\ dnis=${IVR-Exten})
exten = s,n,MYSQL(Fetch fetchid ${resultid} password)
exten = s,n,MYSQL(Clear
Hi Joe,
The debug trace you've enclosed is a NOTIFY message sent from * for the
message waiting feature - and is not related to the call.
You can however tell that something is wrong since the message is being
retransmitted since the server didn't receive 200 OK in reply - while it
could be due
And if they get you black-listed you can always signup with Verizon...
On 4/8/07, Dean Collins [EMAIL PROTECTED] wrote:
There's no way for them to tell if you have asterisk on the fxo port BUT
they will terminate your account if you hook it up as the outbound for an
office pumping call after
You can set up a simple mysql table with PIN-users this makes it more
extensible and you can create a simple web interface to change to pins/add
users.
after you have set up the table just use a simple IVR construct to prompt
for the PIN, fetch it from the table and authenticate it - something
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