Hello everyone.
The share is working and I'm now able to play audio files from a windows share.
Thanks everyone for the help!
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On 10 Jan 2013, at 02:10, Jai Rangi wrote:
I have removed yours right away.
Yes, I agree, But just like any company we have purchased/collected email
from different source. Also just like any company we are not perfect, we make
mistakes.
Then buy your addresses from different sources,
Joshua Colp wrote:
Kai-Uwe Jensen wrote:
On Wed, Jan 9, 2013 at 5:38 PM, Roy Abshire r...@coopvr.com
mailto:r...@coopvr.com wrote:
I have the transport=google-v1 too but restarting Asterisk always
solves my problem for a day...so how do you know that fixed it?
I don't.
If any of you can
Hi All,
I want to test dnd sevice using openims and asterisk. I have registered my
sip client with openims server. I have configured asterisk as an
application server on openims server. When I register my client with
openims server it sends subscription for reg event and ua-profile event for
Hi,
Have you experienced Asterisk 11 in production ?
What do you think of it ?
Which libpri version, if any, did you then associate with Asterisk 11 ?
Regards
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I don't presently have 11 in production, but in each case where I've put 11
in on top of 10.X the process has been relatively seamless, so I expect my
10.X boxes will go to 11.X sometime this year.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Thu, Jan 10, 2013 at 8:18 AM, Danny Nicholas da...@debsinc.com wrote:
I don’t presently have 11 in production, but in each case where I’ve put
11 in on top of 10.X the process has been relatively seamless, so I expect
my 10.X boxes will go to 11.X sometime this year.
**
Upgrading
On Thu, Jan 10, 2013 at 8:23 AM, RSCL Mumbai rscl.mum...@gmail.com wrote:
Hello,
Can asteriskCDR logs tell me if a call was disconnected by the caller
or the Agent ?
My call flow is as follows:
Caller Dials a DID Inbound Routes Play Greeting Call Enter
Queue Call sent to Dynamic
Here at our facility, we use QueueMetrics to analyze our call queues. One
of its standard reports is Disconnection causes, where it does list Agent
disconnected, Caller disconnected and Transferred as options.
If you wanted to try a hand at rolling your own solution to look at this,
check out
are you using cisco 79xx phones ?
We had a similar problem. Upgrading the sip firmare to 8.12 fixed it for us.
FWIW we're using 11 in a call centre, with 25k+ call attempts per day.
Rock solid. Not a single crash since Oct 15
Julian
On 10 January 2013 14:25, Christopher Harrington
On Wed, 9 Jan 2013, Jai Rangi wrote:
But just like any company we have purchased/collected email from
different source. Also just like any company we are not perfect, we make
mistakes.
Right. Because this 'other source' just happened to collect the address
that I only use on this list.
--
On Wed, 9 Jan 2013, chris wrote:
I have gotten hit with this twice so far. in March and Today:
Rohit Dhaka ro...@didforsale.com via mail.bingotelecom.com 3/8/12
DIDForSale donotre...@didforsale.com via mail.bingotelecom.com 1/9/13
didforsale.com (209.216.2.251) - bingotelecom.com
Nope, we're using Digium D40's.
On Thu, Jan 10, 2013 at 8:52 AM, Julian Lyndon-Smith aster...@dotr.comwrote:
are you using cisco 79xx phones ?
Nope, we're using Digium D40's.
We had a similar problem. Upgrading the sip firmare to 8.12 fixed it for
us.
FWIW we're using 11 in a call
On Thu, Jan 10, 2013 at 5:52 AM, Joshua Colp jc...@digium.com wrote:
Joshua Colp wrote:
If any of you can file an issue I'll get this sorted as soon as possible
(probably over the weekend, or maybe even sooner!).
https://issues.asterisk.org/jira/browse/ASTERISK-20916
--
Could this be why incoming calls to voice do not ring asterisk extensions
too?
On Jan 10, 2013 4:52 AM, Joshua Colp jc...@digium.com wrote:
Joshua Colp wrote:
Kai-Uwe Jensen wrote:
On Wed, Jan 9, 2013 at 5:38 PM, Roy Abshire r...@coopvr.com
mailto:r...@coopvr.com wrote:
I have the
Roy Abshire wrote:
Could this be why incoming calls to voice do not ring asterisk
extensions too?
When the issue occurs, yes.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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Hi All,
Is it possible to register a sip client to openims and subscribe DND to
asterisk? Here asterisk is used as AS for openims? Assist please.
thanks,
Isshed.
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Jai Rangi wrote:
I am sure we all get lots if spam emails every day.
Yes, I do. Now ask yourself why I was able to immediately identify
where my address was harvested from for this particular piece of spam.
The answer has to do with DIDForSale's business practices as observed
on the Asterisk
+1
On Jan 9, 2013 8:59 PM, Don Kelly d...@donkelly.biz wrote:
Jai,
It should not be necessary for me to remove my email address from your
list. It should not be on there to start with—we do not have, and have
never had, a relationship that justified you sending me email.
--Don
Don Kelly
On Thu, Jan 10, 2013 at 10:18 AM, Matthew J. Roth mr...@imminc.com wrote:
You already have all of our addresses. Please unsubscribe us.
It's the only way to partially redeem yourself on this list.
Do you really want to be pissing off some of the most active Asterisk
users? You've already
I just ignore spam if I'm not interested and flag them so they go right
into my trash folder.
I think its more exhausting debating the issue on this forum.
I got the email too from DIDForSale but now I'm getting alot more from
this thread.
It really didn't bother me as much as reading all the
+1
On 10/01/2013 1:04 PM, Roy Abshire wrote:
I just ignore spam if I'm not interested and flag them so they go
right into my trash folder.
I think its more exhausting debating the issue on this forum.
I got the email too from DIDForSale but now I'm getting alot more from
this thread.
It
On Thu, Jan 10, 2013 at 11:04 AM, Roy Abshire r...@coopvr.com wrote:
It really didn't bother me as much as reading all the posts but that's
just me...now back to Asterisk issues :)
Sorry to add another, but for me, the main point is that this activity
speaks to the character, ethics, and
That does not solve any asterisk issue that I have.
On 10/01/2013 1:32 PM, Carlos Alvarez wrote:
On Thu, Jan 10, 2013 at 11:04 AM, Roy Abshire r...@coopvr.com
mailto:r...@coopvr.com wrote:
It really didn't bother me as much as reading all the posts but
that's just me...now back to
After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a
Segmentation fault.
[root@localhost asterisk-11.1.2]# asterisk -vvc
Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk comes with ABSOLUTELY NO
First thing to *ALWAYS* check is if you have any Asterisk version specific
modules (Fax for Asterisk, G.729, etc). Ensure these are not loaded (noload in
modules.conf, or simply move them out of the asterisk modules dir).
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
I see asterisk is finding res_jabber.so not compiled for your asterisk
version. As Tim just said, remove all the modules from
/usr/lib/asterisk/modules and reinstall asterisk.
[2013-01-10 14:20:10] WARNING[27062]: loader.c:804 inspect_module: Module
'res_jabber.so' was not compiled with the same
Just ignore the very clear subject line.
On Thu, Jan 10, 2013 at 1:04 PM, Roy Abshire r...@coopvr.com wrote:
I just ignore spam if I'm not interested and flag them so they go right into
my trash folder.
I think its more exhausting debating the issue on this forum.
I got the email too from
So what asterisk issue do you have? Let's fix it.
On Thu, Jan 10, 2013 at 1:49 PM, Ron Wheeler
rwhee...@artifact-software.com wrote:
That does not solve any asterisk issue that I have.
On 10/01/2013 1:32 PM, Carlos Alvarez wrote:
On Thu, Jan 10, 2013 at 11:04 AM, Roy Abshire
Hopefully it's not, What is the best DID provider for Asterisk...
On Thu, Jan 10, 2013 at 1:37 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
So what asterisk issue do you have? Let's fix it.
On Thu, Jan 10, 2013 at 1:49 PM, Ron Wheeler
rwhee...@artifact-software.com wrote:
That
A tier one provider.
On Thu, Jan 10, 2013 at 3:44 PM, Carlos Alvarez car...@televolve.com wrote:
Hopefully it's not, What is the best DID provider for Asterisk...
On Thu, Jan 10, 2013 at 1:37 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
So what asterisk issue do you have? Let's
Although people complaining of spam may be valid, the one part of complaining about spam that bothers me, is that some people should look at themselves in the mirror and ask the question aloud "Why does it bothers me to see another Asterisk professional compete with me for jobs?", "Why do I get
On Thu, Jan 10, 2013 at 3:09 PM, C. Savinovich
c.savinov...@itntelecom.comwrote:
Although people complaining of spam may be valid, the one part of
complaining about spam that bothers me, is that some people should look at
themselves in the mirror and ask the question aloud Why does it bothers
On 10 Jan 2013, at 22:09, C. Savinovich c.savinov...@itntelecom.com wrote:
Unfortunately, there is a fine line between being a forum where people can
exchange ideas, and being a forum where people can find asterisk consultants,
and both don't seem to co-exist well together.
Isn't this
Hello,
I am playing with the manager interface and it seems I cannot catch the
event of a phone subscribing to an hint. Is there a way to catch this kind
of event using the manager interface? I use custom device states, so when a
phone subscribe to a hint, the device is created on the fly. I'd
Isn't this precisely the raison d'être for [asterisk-biz]? Oh my goodness!, the asteriz-biz? nooo, they will kill you if you try to post anything offering your services!... that list ceased to provide any value and died a long time ago precisely because its members ran each other away from it. A
There is a big difference between publicly posting offering services
to the list and harvesting all the email addresses and them contacting
everyone privately
On Thu, Jan 10, 2013 at 5:32 PM, C. Savinovich
c.savinov...@itntelecom.com wrote:
Isn't this precisely the raison d'être for
On Thu, 10 Jan 2013, C. Savinovich wrote:
Are you saying that if they would have posted in the regular forum
offering their services, then it would have been okay with you?
Yes.
On the -biz list.
--
Thanks in advance,
With all my respect guys, I do have my asterisk mailing list setup as
send-as-soon-as-their-is-a-message.
I'm getting too many email from this thread that I seriously don't care
about, and that should be taking out of here.
If you guys want to discuss, I suggest you email between each other,
We are about to announce the availability of a product that was written
to solve data integration problems in Learning Management implementations.
It might help people who need to merge Asterisk CDR data with trunk
provider's data and feed a CRM or billing system.
I would like to think that I
I'm getting too many email from this thread that I seriously don't care
about, and that should be taking out of here.
Do you understand the meaning of the word forum? How can a debate on the
purpose of this list can possibly not be related to this list?
-Original Message-
From: Frank
On 10/01/2013 5:34 PM, chris wrote:
There is a big difference between publicly posting offering services
to the list and harvesting all the email addresses and them contacting
everyone privately
We have to understand that we are going to be approached by people who
think that we need their
This is something I've seen with some key systems and PBXs. When the
phones are on-hook, they can play music throughout the office instead of
having an overhead speaker system do it. Never heard of it being done with
VoIP, but figured I'd ask if anyone else has. I don't see any way to do
this
I've seen this implemented on polycom phones where a secondary extension is
on the phone that is setup to auto answer and they have something on the
PBX side that is configured to call some or all of the secondary extensions
On Jan 10, 2013 8:28 PM, Carlos Alvarez car...@televolve.com wrote:
On Thu, Jan 10, 2013 at 7:31 PM, chris tknch...@gmail.com wrote:
I've seen this implemented on polycom phones where a secondary extension
is on the phone that is setup to auto answer and they have something on the
PBX side that is configured to call some or all of the secondary extensions
On Thu, Jan 10, 2013 at 6:42 PM, Christopher Harrington ch...@acsdi.comwrote:
Wow, that seems wildly bandwidth inefficient. Is it possible to do
multicast VoIP?
Depends on whether the phones are local to the server. Unless you're
looking at hundreds of phones, a 100MB network running 80k to
Lol yes it was all local on a gigE network even :) I also didnt say it
was the most elegant solution but it seemed to work well with them
they even had grouped it into extensions and I'm sure you could even
write some logic to make sure the calls are local
On Thu, Jan 10, 2013 at 9:31 PM, Carlos
On Thu, Jan 10, 2013 at 7:53 PM, RSCL Mumbai rscl.mum...@gmail.com wrote:
Hello,
Can asteriskCDR logs tell me if a call was disconnected by the caller
or the Agent ?
My call flow is as follows:
Caller Dials a DID Inbound Routes Play Greeting Call Enter
Queue Call sent to Dynamic
On Fri, Jan 11, 2013 at 10:29 AM, Satish Barot satish4aster...@gmail.comwrote:
On Thu, Jan 10, 2013 at 7:53 PM, RSCL Mumbai rscl.mum...@gmail.comwrote:
Hello,
Can asteriskCDR logs tell me if a call was disconnected by the caller
or the Agent ?
My call flow is as follows:
Caller Dials a
What I get on asterisk is the following line.
Failed to authenticate user 720001
sip:720...@open-ims.test;tag=1447049631447050637
for SUBSCRIBE
On Fri, Jan 11, 2013 at 10:11 AM, isshed isshed@gmail.com wrote:
Hi Franz,
I am attaching a trace file. Please consider user 720001
Am 11.01.2013 02:42, schrieb Christopher Harrington:
Wow, that seems wildly bandwidth inefficient. Is it possible to do
multicast VoIP?
Snom phones[*] do support multicast streaming. You can setup an
IP port combination that the phone will accept audio at; once
stream data starts arriving,
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