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
- allow=vp8,g722,h263,h265,opus
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 2016-02
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 unders
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.
interval - Interval, in seconds. Default is 15.
https://wiki.asteris
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 i
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",
"1400
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 As
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 is
this is my secondary email
Regards
Zohair
On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry wrote:
> Tried disabling qualify and changing frequency with qualify=yes already,
> no luck :(
>
>
> On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf > wrote:
>
>> I believe qual
Tried disabling qualify and changing frequency with qualify=yes already, no
luck :(
On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf
wrote:
> I believe qualify parameters does help in doing so. Asterisk forgets about
> the peer info when "qualify" are not acknowledged. You can also check
> "quali
Hi,
I have this for UAE,
D/M/YA
Arabian Standard Time
2.2.2.2
Unicast
and this for Kenya
http://wwp.greenwichmeantime.com/time-zone/gmt-plus-3/ my Kenya(Saudia
Ar
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
>>
>> 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
>>
>> 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 t
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 recogn
Behalf 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 no
?
My suspicion is that you are spawning 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
asterisk-users-boun...@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 l
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
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
901 with Customer
200, answers the calls etc. After disconnect a new call arrivers
immediately from Queue 901, without any wrap-up time. This should be
considered as a bug IMO.
Any ideas on how to fix, workaround this problem?
Kind regards,
Henry Dogger
Telecats BV
0-11-23 08:24 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
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
It's replying so its up :)
On 23 Oct 2010 17:32, "Jonas Kellens" 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 guessthis --hostname XX
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.
Tha
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 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 LDAP with
> Asterisk
Any probs with the circuits?
Try and upgrade?
On 17/03/2010, Russell Brown 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 zaptel 1.4.12.1
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 wrote:
>>
>> Hey Philipp,
>>
>
> You can check out
> http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk for
> sett
Why not pay for missing feature and contribute them to the project.
It's a very good product.
On 06/02/2010, bilal ghayyad 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 repeated
> that let t
What are the LDAP searches like?
On 05/01/2010, Jorge Salamero Sanz 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 and dialplan are
Which version of the LDAP schema? I look after the one in 1.6.
Thanks.
On 29/09/2009, John A. Sullivan III 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 integrate a ldap server ( wit
Aastra phones need reboots too :-(
On 20/09/2009, Alex Balashov 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 Polycoms you will
>>
2009/8/24 David Klaverstyn :
> 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.
--
http://www.suretecsystems.com/services/openldap/
http://www.s
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.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - Octobe
2009/7/31 Gordon Henderson :
> 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 did check to s
2009/7/31 Steve Howes :
>
> 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.
2009/7/31 Gordon Henderson :
> 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 tr
2009/7/31 Gordon Henderson :
> 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 tr
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/
http:/
Yeah, and the fxs port too.
On 18/07/2009, Alan Lord (News) 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
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 wrote:
> On Sat, Jul 18, 2009 at 4:36 AM, Alan Lord (News)
> wrote:
>
>> On 18/07/09 00:35, Gavin Henry wrote:
>> > This has to be an Ast
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.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mai
That is correct. That is the first test we did.
On 07/06/2009, Moises Silva wrote:
> On Sat, Jun 6, 2009 at 3:18 PM, Gavin Henry wrote:
>> Every call as soon as the sangoma card is live.
>>
>> Speak to Konrad on your techdesk for more info.
>>
>> Thanks.
>
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 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 we
>> get crosstalk.
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 curren
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 curren
2009/6/2 John A. Sullivan III :
> 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 :
>> > OpenLDAP isn't an option. And thanks very much f
2009/6/2 John A. Sullivan III :
> 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 for the call tree, voice mail, c
2009/6/2 John A. Sullivan III :
> 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 is, I wonder how we pass i
One last thing ;-) use OpenLDAP!
On 02/06/2009, John A. Sullivan III 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 Asterisk, FreePBX,
> Kamail
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 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
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 wil
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 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
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 wrote:
> Miguel Molina writes:
>
>> Hi everyone,
>>
>> I was
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 wrote:
> Hi David,
>
>
>
> Thanks for the reply. That's pretty much what I've already tried, but
> with no luck on the production
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 wrote:
> I have connected my Asterisk-box directly to my internetconnection. I
> have disabled my firewall.
>
> Still I
2009/4/23 Matt Riddell :
> 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 ??
&g
2009/4/20 jonas kellens :
> 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 Asterisk ???
>
> Thank
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/
__
e="0033972112355", realm="sip.ovh.net",
algorithm=MD5, uri="sip:91.121.129.17",
nonce="0019c92d503f745637b43af4264a11db",
response="04e848af655d00e03d032d9a1c2fae09", opaque="001934772ef6ed5"
Expires: 120
Contact:
Event: registration
Content-Length: 0
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:
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 sip_poke_noanswe
2009/4/3 John Todd :
>
> 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
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.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.
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.
Yeah, I've experienced that. But what can you do other than stick woth
a fat codec.
On 17/03/2009, Gordon Henderson wrote:
> On Tue, 17 Mar 2009, Gavin Henry wrote:
>
>> 2009/3/17 Gordon Henderson :
>>> On Tue, 17 Mar 2009, Geraint Lee wrote:
>
>>> I kn
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 wrote:
> On Tue, 17 Mar 2009, Gavin Henry wrote:
>
>> 2009/3/17 Gordon Henderson :
>>>
2009/3/17 Gordon Henderson :
> 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 get two connectio
2009/3/17 Gordon Henderson :
> 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 o
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 i
2009/3/12 Giorgio Incantalupo :
> 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 (PTP or PMP).
Thanks. It
2009/3/12 Paulo Santos :
> 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 do
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 wrote:
> Hello,
>
> setting up Meetme was very easy. I jut added the MeetMe Application to
2009/2/27 John Todd :
>
> 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 mult
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 t
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 Us
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
Discussion
Subje
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 inst
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
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 is
> 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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Ast
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 just
We do as do Gradwell.com
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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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What did you do to setup a button for alerts?
Thanks.
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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
>
> _
improvement however it would be
>> amazing if the entire dialplan/queues/etc could be put into voicemail
>> as well. Right now one has to use LDAP for account and Mysql for
>> extensions/queues.
>>
>> Quoting Gavin Henry <[EMAIL PROTECTED]>:
>>
>>>
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.
--
http://www.suretecsystems.com/services/openldap/
he
> list and archives, it just might be good if solutions could be posted
> here
> too.
>
> Thanks,
> Mark.
>
> PS: Remember, many people get their answers from mailing list archives.
> So
> we'd rather get them solved than getting the same question on the li
going to try and figure it out right now, but for the benefit of the
> list and archives, it just might be good if solutions could be posted
> here
> too.
>
> Thanks,
> Mark.
>
> PS: Remember, many people get their answers from mailing list archives.
> So
> we'
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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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 Aste
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