Hi All,
I am new to VoIP world and trying to set up asterisk, linphone, and
jssip webrtc.
Settings:
- transport_wss (127.0.0.1, apache ws_tunnel)
- transport_tls (public ip port 5060)
- use_avpf=yes
- ice_support=yes
- dtls enabled (letsencrypt)
- rtcp_mux=yes
-
We've had similar problems with realtime and ODBC on Asterisk 1.8 and 1.11. We
found the issue to be caused by unixodbc and not Asterisk.
When we had problems, it was with unixodbc 2.3.0 and 2.3.1. We've since
downgraded to 2.2.14 and it's been fairly smooth sailing since then.
-H
On
I'm having an issue with my Asterisk 1.8.21 server and hairpinning a call. Any
help would be appreciated.
My Asterisk server sends a call out to my proxy. The proxy then routes the
call back to Asterisk because it recognizes
that the destination is on that same Asterisk server.
When the call
It¹s not clear to me if you¹ve done troubleshooting to determine where the
quality issues are occurring. Try testing outbound/external calls
separately from internal calls (i.e., calls that stay on your network and
don¹t go out over the trunk to the carrier).
If the problem is on internal calls,
Hello everyone
I want to know if it is somehow possible for asterisk to consider new
registration attempts instead of matching them with old nonce
Correct auth, but based on stale nonce received from 'test
sip:3247@1.1.1.1;tag=79a401979bffd0d9o0'
I see messages like the one above, I
I¹ve gotten PlayTones to work, however it stops playing the tones as soon as
the call is answered. I would like to use PlayTones during the call because
I want to have a tone/beep played in the background while call recording is
going on.
Anyone know a way to get PlayTones to work while call is
Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] PlayTones while in call
Henry,
Both Montior and MixMonitor have a 'B' option that plays a periodic tone.
B([interval]): Play a periodic beep while this call is being recorded
I¹m trying to use Playtones to have a tone played periodically throughout
phone calls. Unfortunately, I can¹t seem to get PlayTones to work. I never
hear the audio tones.
Here is the output on the Asterisk console.
-- Executing [19525553312@proxy-dial:2] PlayTones(SIP/testphone-0032,
Recently, I made a change to our dialplan and reloaded Asterisk. To my
surprise, the dialplan was reloaded for calls in progress. This caused a
problem because some of the dialplan changes affected some loops and this
caused an infinite loop.
Is there a way to change this so that reloading
1. Your softphone is not sending g729
[Jun 3 13:11:27] Capabilities: us - 0x10c (ulaw|alaw|g729), *peer -
audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing)*, combined - 0x4
(ulaw)
I think free version of eyebeam doesn't come with g729, try Microsip or
some other with g729 codec.
If it
Tried disabling qualify and changing frequency with qualify=yes already, no
luck :(
On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf
mehroz.ashra...@gmail.comwrote:
I believe qualify parameters does help in doing so. Asterisk forgets about
the peer info when qualify are not acknowledged. You
this is my secondary email
Regards
Zohair
On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry markhenry...@gmail.com wrote:
Tried disabling qualify and changing frequency with qualify=yes already,
no luck :(
On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf mehroz.ashra...@gmail.com
wrote:
I
Hi,
I have this for UAE,
dateTimeSetting
dateTemplateD/M/YA/dateTemplate
timeZoneArabian Standard Time/timeZone
ntps
ntp
name2.2.2.2/name
ntpModeUnicast/ntpMode
/ntp
Hello List,
I have some doubt about direct media settings.
I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone
on IP 10.100.210.51 and a gateway at 10.100.210.254
I have set both gateway and peer to directmedia=yes but still on gateway
I see RTP from asterisk's IP, have
Hello List,
I have some doubt about direct media settings.
I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone
on IP 10.100.210.51 and a gateway at 10.100.210.254
I have set both gateway and peer to directmedia=yes but still on gateway
I see RTP from asterisk's IP, have
Hi all,
We're testing TLS and SRTP on Asterisk 1.8.10.0 and have it working
with a commerical (not self-sign) AlphaSSL wildcard (GlobalSign) using
Blink Lite 1.6.2 as per
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
We've tested with Bria on an iPhone and that doesn't
My question is someone (Digium) must have this working against Polycom
(which is a requirement for this project) with commercial certs since
that's their partner of choice?
I don't believe we've done any interop testing with Polycom phones since TLS
and SRTP support were added to Asterisk.
Ah, this makes sense now. So as of today the status of TLS and SRTP in
anything
other than 1.4.X is unknown?
Umm... no :-)
OK, sorry :-)
Asterisk 1.4 did not have support for SRTP or SIP/TLS. Thus, neither of
these were tested with Polycom phones the last time we did interop testing
Perhaps some help on where to look myself?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: donderdag 3 november 2011 17:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: Monday, November 07, 2011 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?
Perhaps some help on where to look myself?
From
a phantom local call. Also, what
does this merriment look like in Master.csv?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: Monday, November 07, 2011 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial
Of Henry
Dogger
Sent: Monday, November 07, 2011 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?
Yes I do J here is the output http://pastebin.com/qpWqdA50
I don't put the cdr's in csv but in database, so not sure what you want
Anyone?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: dinsdag 1 november 2011 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?
Sorry it took
immediately without any wrap-up time.
Hope this logging helps...
Greetings,
Henry
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren
Selby
Sent: dinsdag 25 oktober 2011 18:07
To: Asterisk Users Mailing List - Non-Commercial
use the agi command record_file.
Which format setting do I need to be able to record in wideband?
Tried: wav, gsm, pcm. Nothing seems to give me the result I desire.
Hope someone can help me.
Kind regards,
Henry Dogger
Telecats BV
:24 AM, Henry Dogger wrote:
I have an aastra 6739i which supports the g722 codec.
Which format setting do I need to be able to record in wideband?
Tried: wav, gsm, pcm. Nothing seems to give me the result I desire.
Shouldn't you try g722 as the format?
Leif
It's replying so its up :)
On 23 Oct 2010 17:32, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello,
I'm trying to use SipSak to check if my Asterisk server is
available/running with the following :
sipsak -vv -s sip:usern...@sip.domain.tld -c sip:usern...@sip.domain.tld
--password
Hi all,
Can anyone help with the logic of which commands to use to say:
1. Extension is 600
2. See if has an ongoing call
3. Check if inbound or outbound to the extension
4. Find callerid of inbound call
Been reading http://www.voip-info.org/wiki/view/Asterisk+manager+API
Using latest 1.6.
Hi,
I look after this but have been very busy for months. Maybe you canhelp me test?
Thanks,
Gavin.
On 23/04/2010, Sean Brady sbr...@gtfservices.com wrote:
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
Any probs with the circuits?
Try and upgrade?
On 17/03/2010, Russell Brown russ...@lls.lls.com wrote:
I'm seeing both inbound and outgoing call failures on our ISDN-30 lines
that only seem to go away when I do a zap restart or in extremis
restart Asterisk (1.4.25 with a Digium TE205P and
Has anyone done this with OpenSIPS? For example where it fronts an
Asterisk cluster with the load balancer module?
Thanks,
Gavin.
On 19/03/2010, Ryan Bullock rrb3...@gmail.com wrote:
Hey Philipp,
You can check out
http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk
Why not pay for missing feature and contribute them to the project.
It's a very good product.
On 06/02/2010, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
I used A2Billing, basically it is nice and fine, but management
possibilities is not that rich, so a lot of staff are need to be
What are the LDAP searches like?
On 05/01/2010, Jorge Salamero Sanz ben...@cauterized.net wrote:
Hi all,
I've updated Asterisk trunk LDAP schema [0] [1] to include queues and other
attributes needed for a working LDAP backend (I'll open a bug to include
these
changes on svn).
SIP users
Which version of the LDAP schema? I look after the one in 1.6.
Thanks.
On 29/09/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote:
On Tue, 2009-09-29 at 11:01 -0300, Rafael Seste wrote:
Hi all,
I looked on the Internet but I didn't find any good how-to.
I would like to
Aastra phones need reboots too :-(
On 20/09/2009, Alex Balashov abalas...@evaristesys.com wrote:
Philipp Kempgen wrote:
IMHO the Polycoms are a bad choice for the test because they
reboot for every modification of the SIP account parameters so
unless you have previous experience with the
2009/8/24 David Klaverstyn d...@klaverstyn.com.au:
I’d appreciate it if someone could give me an answer to using LDAP in
Asterisk 1.6.x
You can use res_config_ldap for storing Asterisk data in a directory
server for the realtime framework.
Thanks.
--
Hi,
Would it be sane to run ntop on the same box as Asterisk or better to
mirror a LAN port etc?
http://www.ntop.org/OpenSourceVoipMonitoring.pdf
Thanks.
___
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AstriCon 2009 -
Hi All,
Has anyone passed the tests using Asterisk:
http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html
I presume the same rules apply for scaling and possibly have
OpenSIPS/Kamailio on the front?
Thanks.
--
http://www.suretecsystems.com/services/openldap/
2009/7/31 Gordon Henderson gordon+aster...@drogon.net:
On Fri, 31 Jul 2009, Gavin Henry wrote:
Hi All,
Has anyone passed the tests using Asterisk:
http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html
Intersting. Looks like BT trying to become an ITSP
2009/7/31 Gordon Henderson gordon+aster...@drogon.net:
On Fri, 31 Jul 2009, Gavin Henry wrote:
Hi All,
Has anyone passed the tests using Asterisk:
http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html
Intersting. Looks like BT trying to become an ITSP
2009/7/31 Steve Howes st...@geekinter.net:
On 31 Jul 2009, at 08:22, Gavin Henry wrote:
Has anyone passed the tests using Asterisk:
BT guy we spoke to said yes : )
Good to know!
--
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com
2009/7/31 Gordon Henderson gordon+aster...@drogon.net:
On Fri, 31 Jul 2009, --[ UxBoD ]-- wrote:
Gordon,
Cast your mind back as I had a similar issue ... changing the cable sorted
it for me!
Cursiously enough, I thought about that - but these were 2 brand new
cables out of packets and I
Exactly. I was thinking that a similar service would be a good addon
as an option to an ITSP.
Gavin.
On 18/07/2009, Steve Totaro stot...@totarotechnologies.com wrote:
On Sat, Jul 18, 2009 at 4:36 AM, Alan Lord (News)
alansli...@gmail.comwrote:
On 18/07/09 00:35, Gavin Henry wrote:
This has
Yeah, and the fxs port too.
On 18/07/2009, Alan Lord (News) alansli...@gmail.com wrote:
On 18/07/09 00:35, Gavin Henry wrote:
This has to be an Asterisk based appliance no?
http://www.truecall.co.uk/acatalog/trueCall_Features.html
I saw this on the TV the other night. Couldn't believe how
This has to be an Asterisk based appliance no?
http://www.truecall.co.uk/acatalog/trueCall_Features.html
Looks pretty easy to setup using AstLinux or similar.
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asterisk-users
That is correct. That is the first test we did.
On 07/06/2009, Moises Silva moises.si...@gmail.com wrote:
On Sat, Jun 6, 2009 at 3:18 PM, Gavin Henrygavin.he...@gmail.com wrote:
Every call as soon as the sangoma card is live.
Speak to Konrad on your techdesk for more info.
Thanks.
I'll
Hi,
Has anyone ever experienced crosstalk with sangoma analogue cards on 64bit?
We have exhausted every test to try and replicate this and find a
solution with Sangoma tech support, but we can not fix it.
We are about to try the card and four *seperate* UK BT lines in a 32bit system.
The
Hi,
Has anyone ever experienced crosstalk with sangoma analogue cards on 64bit?
We have exhausted every test to try and replicate this and find a
solution with Sangoma tech support, but we can not fix it.
We are about to try the card and four *seperate* UK BT lines in a 32bit system.
The
Every call as soon as the sangoma card is live.
Speak to Konrad on your techdesk for more info.
Thanks.
On 06/06/2009, Moises Silva moises.si...@gmail.com wrote:
Currently we have put in a temp OpenVOX tdm400 card and it works
perfectly. As soon as we swap that and use Sangoma via wanrouter
Where do they currently change their password? If it's somewhere you
control, why not add some to create the realmed password?
Gavin.
On 02/06/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote:
Hello, all. I'm afraid I've been dropped into the deep end even though
I am an
It also depends where you are registering your users. If merely using
Asterisk for a media server, do the auth via LDAP in Kamailio, which
will just use the userPassword attribute (or however the Kamailio LDAP
module binds to check auth or what you script it to do) then a normal
password change
Sorry, lastly I defined it as auxilary to do exactly that; add it to
any existing entry.
Thanks.
On 02/06/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote:
Hello, all. I'm afraid I've been dropped into the deep end even though
I am an Asterisk novice. I've set up a few tiny,
One last thing ;-) use OpenLDAP!
On 02/06/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote:
Hello, all. I'm afraid I've been dropped into the deep end even though
I am an Asterisk novice. I've set up a few tiny, tiny systems in the
past and have now been asked to pull together
2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com:
Most of the desktops are KDE and they use the KDE change password
facility. It works via pam I believe. Is there an Asterisk interface
with pam that would cause it to simultaneously change the Asterisk SIP
realm password? If there
2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com:
grin OpenLDAP isn't an option. And thanks very much for all the
responses. I've not had a chance to mock it up yet and see how it works
hands on. I am planning that the users ultimately interface SIP to
Kamailio and use Asterisk
2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com:
Thanks. I do appreciate the input as I am jumping into the deep end as
I said :)
On Tue, 2009-06-02 at 21:43 +0100, Gavin Henry wrote:
2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com:
grin OpenLDAP isn't an option
Is there any document on the reasons for the 1.6.0 and 1.6.1 branches?
I remember reading something but can't find it again.
Was it stability versus new features?
I'm currently playing with 1.6.1
Gavin.
On 19/05/2009, Benny Amorsen benny+use...@amorsen.dk wrote:
Miguel Molina
Why not use OpenSIPS or Kamailio in stateful mode?
You will need to look at how media is handled though, but a SIP proxy
will work easily.
On 13/05/2009, Adrian Marsh adrian.ma...@ubiquisys.com wrote:
Hi David,
Thanks for the reply. That's pretty much what I've already tried, but
with no
Is your box on a public ip or via nat? If eth0 isn't the ip you set it
to bind on it will ignore it.
I mean, is your * box on an internal address?
On 02/05/2009, jonas kellens jonas.kell...@telenet.be wrote:
I have connected my Asterisk-box directly to my internetconnection. I
have disabled my
2009/4/23 Matt Riddell li...@venturevoip.com:
On 18/04/2009 2:28 a.m., Gavin Henry wrote:
Hi all,
What other open source tools are people using for this? I was looking
at Openfire and their asterisk plugin.
Is it easy to roll your own with res_jabber.so ??
I used openfire in the past
2009/4/20 jonas kellens jonas.kell...@telenet.be:
Please, is there anyone who can help me with this zaptel -- Dahdi -problem
??
Will chan_dahdi.conf work with zaptel.conf ? Will Asterisk be able to
communicate with the Digium TDM pci-card ?
Or do I need to compile dahdi and recompile
Hi all,
What other open source tools are people using for this? I was looking
at Openfire and their asterisk plugin.
Is it easy to roll your own with res_jabber.so ??
Thanks.
--
Sent from my mobile device
http://www.suretecsystems.com/services/openldap/
a écrit :
And sip set debug peer ovh?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Sent: Tuesday, April 07, 2009 4:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
hello every body
my connexion on ovh to pass in UNREACHABLE and not reidentified were not
reboot the server.
[Apr 7 20:17:21] NOTICE[19947]: chan_sip.c:15605
handle_response_peerpoke: Peer 'ovh' is now Lagged. (2067ms / 2000ms)
[Apr 7 20:17:35] NOTICE[19947]: chan_sip.c:19829
nat=yes
qualify=yes
insecure=port,invite
context=entrant-ovh
thank you.
Danny Nicholas a écrit :
Show us your sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Sent: Tuesday, April 07, 2009 2
Hi all,
Has anyone put * in between an Avaya and Cisco system to connect two
offices together?
I was thinking about adding a SIP trunk on each side and getting
Asterisk to pass calls between them. There is a leased line for
bandwidth.
Any tips/ideas on whether this is possible or dumb?
Thanks.
BTW, what's the recommended production version of Asterisk source
you'd recommend, the latest 1.4 or 1.6?
In fact, nevermind. This is asked so many times I'll hit the archives.
Cheers.
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2009/4/3 John Todd jt...@digium.com:
On Apr 3, 2009, at 7:40 AM, Gavin Henry wrote:
Hi all,
Has anyone put * in between an Avaya and Cisco system to connect two
offices together?
I was thinking about adding a SIP trunk on each side and getting
Asterisk to pass calls between them
2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
On Mon, 16 Mar 2009, Gavin Henry wrote:
Dear all,
I'm currently researching options for a MT asterisk gui/system for a
small business centre that will have 12 units in it. Each unit will be
configured for one extension.
The system
2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
On Tue, 17 Mar 2009, Geraint Lee wrote:
We can put about 9/10 calls using SIP/gsm through our BT Business Network
ADSL package connection (832kbit upstream, £65/month) before you notice
the
quality starting to drop, but you could always
A2billing is a good fit for that then. Yeah, voipon. Thanks for the
input Gordon. Maybe worth hooking up offline if we're doing similar
stuff.
Gavin.
On 17/03/2009, Gordon Henderson gordon+aster...@drogon.net wrote:
On Tue, 17 Mar 2009, Gavin Henry wrote:
2009/3/17 Gordon Henderson gordon
Yeah, I've experienced that. But what can you do other than stick woth
a fat codec.
On 17/03/2009, Gordon Henderson gordon+aster...@drogon.net wrote:
On Tue, 17 Mar 2009, Gavin Henry wrote:
2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
On Tue, 17 Mar 2009, Geraint Lee wrote:
I know
Dear all,
I'm currently researching options for a MT asterisk gui/system for a
small business centre that will have 12 units in it. Each unit will be
configured for one extension.
The system there will have a max of 12 concurrent calls to PSTN
provided via an ADSL/SDSL link to our VoIP provider
2009/3/12 Paulo Santos paulo.r.san...@sapo.pt:
Gavin Henry wrote:
Hi All,
We've got msidn configured:
Port 1: TE-mode BRI S/T interface line (for phone lines)
- Protocol: DSS1 (Euro ISDN)
- childcnt: 2
I don't know if it depends on the card, but in my case I need to set
2009/3/12 Giorgio Incantalupo gincantal...@fgasoftware.com:
Hi Gavin,
if you can make and receive calls it works...do not worry if your line
is shown as DOWN, some telco turns it off but it works without problem.
Remember to ask your telco for the right signalling and set it the right
way
Hi All,
We've got msidn configured:
Port 1: TE-mode BRI S/T interface line (for phone lines)
- Protocol: DSS1 (Euro ISDN)
- childcnt: 2
mISDN_close: fid(3) isize(131072) inbuf(0x8fd5060) irp(0x8fd5060)
iend(0x8fd5060)
and running on Asterisk 1.4.21.2:
pbx*CLI misdn show stacks
Just transfer them to your meetme extension after you've called them.
Just like you would transfer someone who has called you.
* will then put them into that conference.
Thanks.
On 08/03/2009, Sven Geggus use...@fuchsschwanzdomain.de wrote:
Hello,
setting up Meetme was very easy. I jut added
2009/2/27 John Todd jt...@digium.com:
On Feb 26, 2009, at 6:01 PM, Senad Jordanovic wrote:
Gavin Henry wrote:
Hi all,
In a pure VoIP env, what is the current state of do's and don't s of
virtualizing * in order to provide multiple separate instances, say
for hosting lots of Asterisk-gui
Hi all,
In a pure VoIP env, what is the current state of do's and don't s of
virtualizing * in order to provide multiple separate instances, say
for hosting lots of Asterisk-gui/FreePBX/a-n-other gui?
I've read lots of threads going back to 2007 and I'm in the general
option that kvm is the way
Try first just asterisk and after asterisk -r
If still doesn't start try asterisk -c to verbose...
Best regards,
Chris Hariga
--Original Message--
From: Scott Berry
Sender: asterisk-users-boun...@lists.digium.com
To: Asterisk Users
ReplyTo: n7...@northlc.com
ReplyTo: Asterisk Users
X-lite from CounterPath work with Asterisk. No g729 support on the free
version. If u plan to use ulaw will work perfectly.
Best regards,
Chris Hariga
--Original Message--
From: Georgecooldude
Sender: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
ReplyTo: Asterisk Users Mailing
On my eeePC I install windows, for the same reasons, sound video drivers...
Chris
Sent from my BlackBerry® smartphone with SprintSpeed
-Original Message-
From: Joseph [EMAIL PROTECTED]
Date: Sat, 15 Nov 2008 22:39:40
To: Asterisk Users Mailing List - Non-Commercial
That looks cool. Will have a play.
On 10/18/08, Ming Yong [EMAIL PROTECTED] wrote:
Anael,
You should take a look at Druid (Open Source Unified Communications)
Project based on Asterisk that has complete LDAP backend and Zimbra
connector.
It's an open source project we are looking for
The LDIF needs updating as it's not a working example. I'll have one
next week. I'll release an updated schema too.
Gavin.
On 10/18/08, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Saturday 18 October 2008 02:30:16 Anael DIAZ wrote:
I need help in implementing Asterisk with LDAP. I' ve
I use Amazon EC2 when my capacity reach the max limit. Because I don't have
control on witch datacenter or Internet connection my new virtual machine will
start I got some problems, not very often, with low bandwith and now I'm
working on a new AMI with watchdogs for voice quality and latency
Or provide both solutions - let the offices call each other via VoIP, but
if too laggy, fall-back to VoIP - PSTN... (- VoIP)
How can you test for this precall?
Cheers.
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Thanks all for your suggestions.
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Dear All,
What setup would you recommend for making VoIP calls whilst bringing
latency down between offices at:
* Edinburgh
* Kuala Lumpur
* Singapore
* Tokyo
* Seoul
* Beijing
* San Francisco
Some of the Asia offices are 300ms some 200ms.
Any advice greatly apreciated.
Thanks.
I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm using
CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk. When I
try to build Asterisk this is the error I'm getting.
src/add.c:1: error: CPU you selected does not support x86-64 instruction set
I
2008/7/2 Loic Didelot [EMAIL PROTECTED]:
Depends on the phone.
On many devices you can setup buttons to call a url. Thats what I did.
Ah, yes. Would be a good thing to implement here. Then you can do
anything, like a support ticket etc.
Cheers.
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asterisk-users mailing
What did you do to setup a button for alerts?
Thanks.
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To
Google Apps version might.
2008/6/25 Marc Smith [EMAIL PROTECTED]:
Hi,
Anyone using Asterisk IMAP voicemail storage with Google Apps / GMail
IMAP? If so, does their IMAP implementation support any kind of
master user (Dovecot) abililty? Good? Bad?
--Marc
LDAP for account and Mysql for
extensions/queues.
Quoting Gavin Henry [EMAIL PROTECTED]:
On 17/03/2008, Faraz Khan [EMAIL PROTECTED] wrote:
Good Idea and done. It is now available here:
http://www.voip-info.org/wiki/view/LDAP
The correct LDAP Schema is included:
/asterisk-1.6.0-beta4
2008/6/16 Syed Nasruddin [EMAIL PROTECTED]:
Thanks for the link. I think I will be using this product.
It's very, very good. You can hook it up to MySQL instead of sqlite if
needed, just e-mail support.
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http://www.suretecsystems.com/services/openldap/
months later. :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gavin
Henry
Sent: June 13, 2008 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] cdr-custom/Master.csv rotation
2008/6/13 Mark
the same question on the list 3
months later. :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gavin
Henry
Sent: June 13, 2008 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] cdr-custom/Master.csv
2008/6/12 Syed Nasruddin [EMAIL PROTECTED]:
HI,
I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair
command over Asterisk up till now and have run it in different scenarios
such as Call Center Solution, PBX solution.
There is a requirement to use Asterisk only as
2008/6/13 Mark Hamilton [EMAIL PROTECTED]:
Hi,
How can I rotate /var/log/asterisk/cdr-custom/Master.csv nightly by date?
Logrotate on a *nix box.
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When I mv a file to /var/spool/asterisk/outgoing in order to place a
call from a user extension that will always be recorded, what
parameter do I set in the call file in order to specify an exact name
for the wav file?
This is on Trixbox and at the moment I'm considering setting an extra
variable
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