Displaying 17 results from an estimated 17 matches similar to: "app_dbodbc segfault"
2004 Jul 22
1
app_dbodbc URGENT
I have been searching for the last two days and I cannot seem to set
Asterisk to work from a database, can someone please tell me what I am doing
wrong here? Here are my files
[app_dbodbc.so] => (Database access functions for Asterisk extension logic)
== Parsing '/etc/asterisk/odbc.conf': Found
> app_dbodbc: dsn is MySQL-asterisk
> app_dbodbc: username is
2005 Aug 03
1
app_dbodbc for asterisk stable 1.09
Hi,
Has anyone manage to comile app_dbodbc or ast_data with the latest
stable release (1.09). If so can you give some guidence on howto do it
as I have trouble getting either working.
Umar
2004 May 14
4
app_dbmysql and ODBC Voicemail
I have done a little work on asterisk and database integration. Below is
a link to app_dbmysql, modeled after Brian's app_dbodbc but for pure
MySQL.
I also ported the mysql-vm-routines.h to ODBC in case anyone is
interested.
You can get both of these from:
http://www.cheapnet.net/~mike/asterisk
They were working as of yesterday CVS, but today CVS will not compile
and I have not looked
2004 Jul 22
4
VSP? Looking for advice.
Has anyone tried using BroadVoice for VSP? I have Asterisk configured
for a home office & I've been trying to decide which VoIP provider to go
with for a little while now. I had heard you could get sub $.01 calls
but I have not found that to be true yet (not saying it's not possible,
I just haven't found it!).
Also I'm not sure if BV will support multiple lines. Any
2005 Aug 04
5
newbiew extensions.conf question
I am newbie trying to setup about 12 Polycom Ip500's
on an asterisk server. I am working on my
extensions.conf and am trying to make it so that all
my extensions can dial each other. My extensions are
number 720, 721, 722, 723 ..etc
in my from-sip context I began doing entries such as:
exten => 720,1,Dial(SIP/720,20)
exten => 720,2,Voicemail(u720)
exten =>
2008 Sep 08
2
Pointers to replace astdb
Hi listers,
We want to implement one call center with asterisk. The idea is it should be
scalable, with openser as an dispatcher and bunch of asterisk servers to do
ACD, Queues, Agents things... Easy to say :(
Look closely to the current asterisk, we do see some problem:
- SIP registrations was stored in astdb.
- And queue members also was stored in astdb.
- ...
asterisk was built as
2012 Aug 26
1
One leg in a conference and adjusting stream volume of other leg
Hi all,
I'm looking for some serious help. :) I couldn't find a better
description for my problem... I think it is quite complex! Here's what I
would like to achieve:
A SIP caller dials into to my Asterisk 10. He will automatically listen
to a specific MP3 stream.
Other SIP callers dial also into my Asterisk. They all will
automatically listen to the same MP3 stream.
All
2004 Dec 11
2
long list of prefixes
if a phone number starts with one of 50+ prefixes,
i want to send the sip call to gateway X. if it
is in any other prefix, i want to send it to gate
Y.
i am not excited about a looooong list of extens,
but will do it if i have to.
i suspect there is a database hack, but i lose all
database contents if i reinstall the port (this
may be a feature of the freebsd port), and i have
not figured out a
2004 May 28
11
Asterisk Database
I'd like to be able to add additional fields to the the Asterisk
database. I'm using Mysql for most of my data lookup and manipulation,
and it seems to work pretty well. In keeping with what I know how to do,
it would be very handy to be able to insert say a "call forward number"
into a customer record. That way, I could automatically route calls to
extensions to a forwarded
2005 Feb 28
4
Recommendation for dialplan in case of DDoS atta cks?
I'm trying to formulate a strategy for our interconnected Asterisk IAX peers
to failover to the PSTN in the event of a DDoS. We currently use them like
this:
DID--->PRI--->Primary Asterisk--->IAX--->On-site Asterisk--->SIP
This works fine, and everyone is happy. One of my concerns, though, is if we
get DDoS'd - which happens probably once every couple of years. I'd
2008 Mar 08
3
replace astdb with a cluster-capable sql database engine
I've been searching the Internet for information
regarding the replacement of astdb with a modern sql
engine.
There are several reasons one would like to do this.
First of all, external applications have a hard time
reading/writing to the now-old astdb format.
Also (and this is what interests me most), the sql
astdb could easily be clustered throughout several
servers (I'm looking for a
2005 May 10
2
RE: Writing To Multiple MySql Tables
Ive got realtime and mysql.cmd to read from databases but apart from cdr
how else can * write? I need to write to 2x tables and cant do this with
cdr?
Any advice appreciated thanks.
---
Rafal Kaniewski
--
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2004 Jun 29
3
incoming cid translation tables
How does one do translation for calls that come in from other pbx's
where the incoming caller ID is an internal extension number on their
pbx? Eg. when I get a call from Free-World-Dial the CID shows up as
"429102" which is essentially their internal extension number sans any
routing prefix. To dial the number back I need to dial the extension
with FWD's routing prefix
2004 May 10
1
DNS load-balancing & SRV records
Let's say I have a third-party device acting as a sip<-->pstn gateway, a
cluster of three asterisk servers, and a teensy bit of dns knowledge.
Let's now say those asterisk servers are a1.company.com at 192.168.0.1,
a2.company.com at 192.168.0.2, and a3.company.com at 192.168.0.3.
1. If I setup round-robin dns like so:
asterisk.company.com. IN A 192.168.0.1
asterisk.company.com. IN
2004 Apr 13
2
T100P E&M Wink Trunk
I am setting up a box with a T100P. Everything is going well. The
company I am working with has their one phone switch gear. They
provisioned me a E&M Wink T1. Cannot do PRI unfortunately. We chose E&M
so we could pass an unlimited number of DIDs to the trunk as apposed to
FXS loopstart signaling. I can make outbound calls no problem, but I am
having problems with the dial plan for inbound
2005 Jul 28
8
dialplan defenition
Hello list,
Im writing my dial plan, in witch every SIP phone begins with 74 and has
more 3 numbers (like 74XXX).
So, I want to route all 74XXX calls to my sip channel. For this I wrote
this line:
exten => s,1,Dial(SIP/74118@193.136.252.5,30,r)
but this way all calls go to 74118@193.136.252.5 .....
Then I tried:
exten => s,1,Dial(SIP/${EXTEN}@193.136.252.5,30,r)
but this way, the
2005 Mar 24
0
AGI commands STDOUT problem
i have a problem with AGI in Asterisk 1.0.5, the problem occurs either
with PHP or C AGI scripts/programs. Well, its simple,
either asterisk is not sending correctly the command responses to the
standard output, or for some unknown reason to me the
scripts/programs are not able to read it from standard input.
I have the next C test program for AGI:
#include <stdio.h>
main()
{
char