Not sure if this is the right place to ask, but what do we need to do to
get this patch merged? How can I help? I'm no dev, but I use LDAP with
Asterisk and I might be of some help.
Thanks guys.
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On 04/21/2010 03:08 PM, Warren Selby wrote:
On Wed, Apr 21, 2010 at 3:46 PM, Jay Vocaire jvoca...@innproc.com
mailto:jvoca...@innproc.com wrote:
Thanks for the tip, I did just that, and now I am more confused.
It does appear as though there is just one call ID (if my
assumption
On 04/21/2010 05:36 PM, bruce bruce wrote:
Here are result of dahdi_test:
[r...@ip-10-251-123-3 ~]# dahdi_test
Opened pseudo dahdi interface, measuring accuracy...
99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434%
-434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770%
On 04/19/2010 02:22 PM, Jay Vocaire wrote:
I have searched everywhere, but cannot seem to find anyone else talking about
this issue. Maybe I am just using the wrong search terms.
I am running Asterisk 1.6.2 and multiple Polycom phones all with 3.2.3 (the
latest) firmware on them.
I am
(sorry this is so long)
I could really use a helping hand. I have a 1.6.2.6 installation using
LDAP as the realtime engine for voicemail users, SIP users, queues, and
some custom hotdesking families. I'm also using ODBC voicemail storage.
The issue that I am having is that the UA's (Polycom
On 04/16/2010 03:39 PM, Nathan Clemons wrote:
I'm looking to find a test tool that will register with our Asterisk
(Trixbox) server here at work and place an outgoing call via our main
SIP trunk (BroadVoice) to confirm that things are working. I've looked
around but I can't seem to find any
On 04/16/2010 10:10 AM, Renato bianchini wrote:
Hi,
Anyone know why sometimes on CLI disappear parameters as sip, stop,...?
Thank you very much by reply.
Renato
AFAIK when a CLI option is not available it means that module isn't
loaded. Check the logs to make sure that module was
Is anyone successfully using DHCP option 66 to specify an FTP [sic]
provisioning of Polycom Sounpoint phones instead of TFTP? I know option 66
is typically used TFTP booting, but the Polycom doc doesn't appear to
specify that option 66 implies TFTP instead of FTP (since you explicitly
call
I have read 2 solutions
(a) Changing the Dial plan and capturing DNID and inserting it into
one of the existing column in CDR table.
(b) Copy new CDR related .c .h files which have added the
functionality of recording DNID into MySQL.
For this, CDR table structure needs to be
Hello all,
My folks would like to play a message to answering machines automatically after
hanging up the phone. So, when the caller dials the number of the callee,
hears an answering machine, they would like to enter a code on the phone and
hang up. After the hangup the message plays to the
I am seeing the exact same behavior on 1.6.2.5. Could this have
anything to do with issue #16816? I'm no developer here, the reason
that I think it might be related is that both apps depend on the
second leg of the call to be answered, and it appears that for some
reason the apps
- --[ UxBoD ]-- ux...@splatnix.net wrote:
- --[ UxBoD ]-- ux...@splatnix.net wrote:
Hi,
we are running Asterisk 1.6.1.14 and have a issue that when we use
followme the call is correctly placed to the mobile phone, the
mobile
rings, but when answered we do not
I'm having a problem with conferences both meetme and app_conference,
though I've done most of the testing with meetme.
What version of DAHDI are you running?
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Sean Brady wrote:
To get MeetMe working properly, I know some sort of timing device
provided by the zaptel package is required (even if it means the
zt_dummy). But, on a virtual machine I know that the Linux timing won't
work as expected. Is it possible to then dedicate a physical device
To get MeetMe working properly, I know some sort of timing device
provided by the zaptel package is required (even if it means the
zt_dummy). But, on a virtual machine I know that the Linux timing won't
work as expected. Is it possible to then dedicate a physical device
like a USB port or
On 5 Feb 2010, at 16:55, Greg Blakely wrote:
If so, how?
NFS or rsync?
S
Use ODBC voice message storage and realtime voicemail configuration.
- Sean
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On 1.6.2 I have also tried using a local channel for the outbound leg
with the originate looking like the following:
action:.Originate..
actionid:.1306903_89#AJ_ORIGINATE_25
timeout:.4
exten:.s
async:.true
callerid:..612
Hi, does anyone have an info into what could cause
[Nov 28 14:24:48] ERROR[11964] utils.c: write() returned error: Broken pipe
I have had the same issue with a PHP script that logs into the manager
interface. If you don't wait for the AMI response, then log off before closing
the connection
Hi everybody,
I would like to use realtime authentification with my LDAP.
It depends on what you are doing with LDAP. There is an LDAP realtime engine
for SIP/IAX peers, voicemail users, asterisk configurations and extensions with
a sample ldif included with the distro, although I
Looking at all the docs I can find Asterisks looks like it should be
able to do the job and a whole lot more.
This is for a small call centre so ideally we want all the features of
an average call centre, ACD, Call Recording, Queue's etc etc.
Any pointers on how to get started would be most
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