2012/2/9, Maximilian Grobecker m.grobec...@portunity.de:
Hello,
I know about the german phone system that the sense of an anonymous call
is, that the called party has no way to get the caller's number in any way.
The last switch honours the anonymous bit and removes the phone
numbers before
Hi
To the best of my understanding this is the correct behaviour. When you
add a peer to the database and a device configured for that peer
registers, it enters that peer into the RealTime cache.
When you do a sip reload you fully clear that RealTime cache so the
asterisk process will lose
Hi, I'm working on a small php program for just this. I guess from your
question that you have Asterisk writing to a CDR database table, in which case
you should be able to use my .php code fairly easily. It's nothing fancy but
does give me a graphical presentation of calls/15minute segments.
I hope I'm not flogging a dead horse here, but the discussion around the whole
scalability issue in Asterisk have opened my eyes to a whole lot of issues,
making me increasingly confused!
We have a fully functioning and stable installation where we offer PBX services
to some 15 small firms
Hi, I forgot to add that you are free to use my code, I'll mail it later today.
Fra: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] På vegne af Brynjolfur
Thorvardsson
Sendt: 10. februar 2012 09:47
Til: Asterisk Users Mailing List - Non-Commercial
Hi,
I'd like to implement some way of controlling remote SIP clients while
in a call, to execute remote commands.
The call topology (think of a PA system) is this:
* the caller is in a MeetMe() conference room
* the callees are Page()d, then the dynamic conference room is connected
to the
Hello,
this is a know problem when you are writing the voicemails over a nfs
link. you have to start asterisk with the -t option to write voicemail
records to the local /tmp and copy it to the final destination after it
is finished.
as far as i remember the first 10 seconds are ok and then the
This is a FreePBX question as the Asterisk dialplan is managed by it.
I suggest to use 'extensions_override_freepbx.conf' (details in
extensions.conf) and place there your modified [macro-dialout-trunk].
HTH,
Ioan
On Fri, Feb 10, 2012 at 1:13 PM, ing.Achim Alexandru
alexandru.achi...@gmail.com
I see this on some peers every time I do a sip reload and I am not using
real-time. I use qualify and every time a sip reload occurs the device goes
unreachable. I have shortend the register time to 5 min so the device
comes back with-in about two min but it is very annonying to me and my
2012/2/10 Brynjolfur Thorvardsson bi...@itanet.nu:
I hope I'm not flogging a dead horse here, but the discussion around the
whole scalability issue in Asterisk have opened my eyes to a whole lot of
issues, making me increasingly confused!
We have a fully functioning and stable installation
--- On Fri, 2/10/12, Leandro Dardini ldard...@gmail.com wrote:
mysql multimaster replication and
asterisk realtime.
Just a word of caution: I've had terrible luck with MySQL NDB tables in a
multimaster setup. I'm not a big expert but v.5.0 and 5.1 have given me lots of
reliability issues
Shaun,
Follwing more tests :
1) In production box asterisk works with 2.4.X dahdi tree + kernel
2.33.x tree. I can put the trunk up and recieve some calls .
2) In the same box I had tested dahdi 2.6.X + same kernel, when run
dahdi_cfg this error messages comes , in dahdi_cfg running and
Is it possible to distribute QUEUE information among several Asterisk nodes in
a multimaster or load balancing setup?
I haven't tried this yet but if one uses realtime with a clustered multimaster
database and the queue agents/members are fixed SIP channels (eg. SIP/100) then
I guess that the
I'm in a similar situation. However, most of my buildings were re-wired
around 1994 to provide Cat5 or 5E to the desktop for data, and 2-pair
Cat3 for voice, all in a star topology. I can move my voice
infrastructure to an IP-based one running 10Mbps, utilize existing
wiring infrastructure,
2012/2/10, Jason W. Parks jason.w.pa...@gmail.com:
I'm in a similar situation. However, most of my buildings were re-wired
around 1994 to provide Cat5 or 5E to the desktop for data, and 2-pair
Cat3 for voice, all in a star topology. I can move my voice
infrastructure to an IP-based one running
Hi Brynjolfur,
Yes, this is exactly what I am looking for - hopefully in English :-)
Date or range selection would make this perfect. I have been looking for
something like this for quite a while but there is none. I would really
appreciate it if you share this with me.
Question here, does the
Hello Folks;
I know this is a non-commercial discussion group, but I am looking for some
open-source software suggestions
We are going to be setting up a prepaid PBX service with the following
features:
*
- Email to Fax and Fax to Email
- Inward DID local and 800 services
-
- Original Message -
Yes, this is exactly what I am looking for - hopefully in English :-)
Date or range selection would make this perfect. I have been looking
for something like this for quite a while but there is none. I would
really appreciate it if you share this with me.
- Original Message -
Hello Folks;
I know this is a non-commercial discussion group, but I am looking for
some open-source software suggestions
We are going to be setting up a prepaid PBX service with the following
features:
• Email to Fax and Fax to Email
• Inward DID
I assume that solution was A2Billing?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Friday, February 10, 2012 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
No, that doesn't do the job I specifically asked and installation
instructions are all over the place...
Thanks though.
On Fri, Feb 10, 2012 at 11:36 AM, Tim Nelson tnel...@rockbochs.com wrote:
- Original Message -
Yes, this is exactly what I am looking for - hopefully in English
There is none. We are looking to develop our own currently and in the
process of hunting down best developers. We have a great deal of experience
with billing systems but doing a fron-end for this purpose just requires
multiple developers. You can e-mail me in private if interested in a shared
Yes, I like the look of that.
Researching it too - the commercial one looks nice too, but I don't know if
there is a budget.
G
On Fri, Feb 10, 2012 at 11:52, Terry Brummell te...@brummell.net wrote:
I assume that solution was A2Billing?
-Original Message-
From:
What have you looked for yet? There are no commercial ones that do all that
in one.
On Fri, Feb 10, 2012 at 11:57 AM, James Wystead szilvertho...@gmail.comwrote:
Yes, I like the look of that.
Researching it too - the commercial one looks nice too, but I don't know
if there is a budget.
G
Thanks for this - but I am looking really for a software type solution.I would venture say that he means he wants it for free.Christian SavinovichVoIP Telephony Consultant646-982-3572
Original Message
Subject: Re: [asterisk-users] Question for the group
From: James Wystead
Hello,
I'm trying the Call Completion system. All work fine.
I still don't undesrtand how Asterisk work.
Does Asterisk use sip signaling or other protocol to send notifications?
On the Asterisk Wiki seems that the system is based on
draft-ietf-bliss-call-completion-04 but this draft talk
On 02/10/2012 11:32 AM, bakko wrote:
Hello,
I'm trying the Call Completion system. All work fine.
I still don't undesrtand how Asterisk work.
Does Asterisk use sip signaling or other protocol to send notifications?
On the Asterisk Wiki seems that the system is based on
Great subject, regardless of the name you post from or the list you post
to.
I'm sure this is the only question that will ever be asked of either
group.
Better subject = better replies = better value for future archive
searchers.
--
Thanks in advance,
I am facing an issue with Peer registration in my asterisk server .
I am using asterisk version 1.8.5.0 and using SIP real-time
architecture.when i am doing registration it registered fine on asterisk
as peer is available in Database.
But now i am doing 'sip reload' or 'reload' due to some
Thank you for your answer Kevin.
Effectively, I'm using CCSS in generic mode.
I hope to try the draft between two Asterisk Server soon.
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Hi Leandro, that's a really good suggestion. Thanks a lot, I'll certainly give
it a try.
-Oprindelig meddelelse-
Fra: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] På vegne af Leandro Dardini
Sendt: 10. februar 2012 14:03
Til: Asterisk Users
I'm trying to implement a very simple call queue for a small, low volume
helpdesk. We have 2-5 agents, and rarely does the queue get more than 1 or 2
callers deep. I'm using the ringall strategy and I want calls answered in FIFO
order.
Say caller A calls the queue, and there is one member
May I know how the compile RPM from Digium Repo gets to install DAHDI so
easily on the VM? Can you please point me to how this compilation is done
so I can have my own RPM of Asterisk with all options added on (e.g.
ooh323, jabber, etc...)
Thanks
On Mon, Jan 16, 2012 at 1:57 PM, Kevin P. Fleming
On Feb 10, 2012, at 14:37 , Phil Frost wrote:
I'm trying to implement a very simple call queue for a small, low volume
helpdesk. We have 2-5 agents, and rarely does the queue get more than 1 or 2
callers deep. I'm using the ringall strategy and I want calls answered in
FIFO order.
Say
I can't see those IPs in the /var/log/asterisk/full. I can't event see
parts of the IP address as I try *grep -o 23.20.189 full. *That is still
nothing.
I am wondering what is wrong here. This is my regex filter file:
failregex = Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Wrong
On 27 January 2012 04:49, asterisk jobs asteriskcod...@gmail.com wrote:
Hello everyone,
I have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen
this or can explain this?
Chain fail2ban-ASTERISK (1 references)
num target prot opt source destination
1
Hi,
I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer
simply stopped functioning. I can downgrade and make it work, upgrading
kills it again. There obviously is a difference in how the newer firmware is
treating this auto answer sip header.
Can anybody tell me if
On Fri, 10 Feb 2012 16:08:28 -0500
Phil Frost p...@macprofessionals.com wrote:
On Feb 10, 2012, at 14:37 , Phil Frost wrote:
Now, caller B calls. Asterisk rings the member. Now the member's
handset is showing two incoming calls.
[...]
Now, the member is unbusy, so he answers a call. But,
On Fri, Feb 10, 2012 at 10:30 PM, Mike l...@net-wall.com wrote:
Hi,
** **
I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer
simply stopped functioning. I can downgrade and make it work, upgrading
kills it again. There obviously is a difference in how the newer
Did the 4.0.1b update overwrite sip.ld on these phones? If I recall
correctly you have to tweak that file to make auto-answer work correctly.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian ipt
Sent: Friday, February 10, 2012
Not really, it did fix the phantom ringing I had (phone continued to ring
when connected to a caller), which was the main reason to upgrade, but I
believe so would upgrading to 3.3.4. Some pluses for me are:
- It does make booting up MUCH faster
- There is a Warning message
It does update the sip.ld file, yes. So does all upgrades, no?
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, February 10, 2012 5:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Mike. Yes sip.ld is the firmware.
I wanted to jump in because i saw you had the phantom ringing problem as well.
I am running 3.3.1 and thought upgrading to 3.3.2 would solve that problem did
you still have the problem in 3.3.2? I thought I saw in the release notes for
3.3.2 that was
Sorry for the top post, but I am using a silly mail client. I havent talked
about ndb tables, just multimaster setup. It is really stable if done with
just two mysql servers. I am running a couple of asterisk servers sharing a
common cdr and cnam database for at least 3 years without problems.
Hello everybody
someone in this list, has installed asterisk, in a virtual server like
proxmox? I'm thinking install some asterisk servers in a machine dell
xeon 64 processor, but I'm not sure, about virtual Server software.
I heard, about proxmox, but I don't know if works fine.
Regards
That was just another weird IP showing up.
On Fri, Feb 10, 2012 at 4:50 PM, dotnetdub dotnet...@gmail.com wrote:
On 27 January 2012 04:49, asterisk jobs asteriskcod...@gmail.com wrote:
Hello everyone,
I have noticed getting wired IPs blocked by Fail2ban. Has anyone else
seen this or can
On Fri, Feb 10, 2012 at 9:18 PM, Carlos Rojas crt.ro...@gmail.com wrote:
Hello everybody
someone in this list, has installed asterisk, in a virtual server like
proxmox? I'm thinking install some asterisk servers in a machine dell xeon
64 processor, but I'm not sure, about virtual Server
On 12-01-26 11:49 PM, asterisk jobs wrote:
Hello everyone,
I have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen
this or can explain this?
Chain fail2ban-ASTERISK (1 references)
num target prot opt source destination
1DROP all -- 0.23.20.189
I run two off virtuozo vps boxes - but capacity will always be the defining
value
Sent from my iPhone 4S
On Feb 10, 2012, at 9:18 PM, Carlos Rojas crt.ro...@gmail.com wrote:
Hello everybody
someone in this list, has installed asterisk, in a virtual server like
proxmox? I'm thinking
I run my Asterisk system on a quad core Opteron system running VMWare ESXI 5.
On Feb 10, 2012, at 21:18, Carlos Rojas crt.ro...@gmail.com wrote:
Hello everybody
someone in this list, has installed asterisk, in a virtual server like
proxmox? I'm thinking install some asterisk servers
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