On Sun, May 15, 2011 at 10:16 AM, sean darcy seandar...@gmail.com wrote:
anyone actually used this on Android to connect to an asterisk server?
Yes. I purchased it a while ago from the Marketplace, and had some
issues with sound quality as my specific phone (Motorola Atrix) isn't
officially
On Fri, Apr 29, 2011 at 7:29 PM, Kaushal Shriyan
kaushalshri...@gmail.com wrote:
I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf,
but its not yet production ready. Can someone please pitch in about HA
feature in Asterisk ? (HA - High Availability.)
The current
On Mon, May 2, 2011 at 1:10 AM, A E [Gmail] all.efor...@gmail.com wrote:
Now, I wonder what're the alternatives that people have been using for
Asterisk HA other than commercially available solutions like HAAST and
Astribanks assuming that kaushal is right and SCF isn't production ready
yet.
On Sun, May 1, 2011 at 3:03 AM, Terry Brummell te...@brummell.net wrote:
8 PRI’s? I’d be using something like an AudioCodes Mediant 1000. No
messing around with switches and cables an crap.
I agree, use a SIP Gateway. The AudioCodes Mediant 1000 supports up
to 4 T1/E1/J1, so use two of them.
On 11-04-06 03:53 PM, Hans Witvliet wrote:
I'm going to have a go with realtime mysql.
Just wondering, most examples i came across while googling, was with 1.6
systems.
So any drastic changes with 1.8.3, table-layout? other pitfalls?
The tables migrate just fine, but you can update them to
On Wed, Apr 6, 2011 at 2:59 PM, Hans Witvliet h...@a-domani.nl wrote:
[snip]
I think i have to stick with mysql, as info is coming from other
applications, but perhaps some of the other code can be tweaked.
mysql is nice (lots of tiny programs writen for it), but i'm not
religious attached
On Fri, Mar 25, 2011 at 7:36 AM, Douglas Mortensen
d...@impalanetworks.com wrote:
But I would like specific reasons why I shouldn't use 1.8 in a production
environment if anyone has some?
That is a loaded question, in that no two environments are likely to
be the same. Some bugs are major
On Mon, Feb 14, 2011 at 10:31 AM, James Miller paramedi...@gmail.com wrote:
I did that and this is what I got when I tried to play the 24 ringtone:
13:29:16.573318 IP 192.168.1.103.50849 192.168.1.60.69: 39 RRQ Emergency
ring_emergency.pcm octet
That line should read something like:
blah..
On Tue, Feb 15, 2011 at 10:15 AM, Nic Colledge n...@njcolledge.net wrote:
I have been having a problem with asterisk crashing when using local
channels and realtime on asterisk 1.8.3-rc2.
Nic,
I can reproduce this using the latest SVN for the 1.8 branch. I
don't get the console locking, but
On Mon, Feb 14, 2011 at 5:40 AM, James Miller paramedi...@gmail.com wrote:
Good Day everyone,
Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by
Cisco, however now the phone does not and will not read the RINGLIST.dat
file. I’ve tried rebooting the phone, tried
On Mon, Feb 14, 2011 at 9:04 AM, James Miller paramedi...@gmail.com wrote:
I did the command listed, and its actually requesting RINGLIST.DAT, so I
changed the filename to match its request but now its showing in the ring
type setting:
Chirp 1
Chirp 2
24 24-ring-tone-1.raw
Att1
On Fri, Feb 11, 2011 at 7:59 AM, satish patel satish...@hotmail.com wrote:
I thought it has been resolved in 1.8.2 version
Issue 18403 was not resolved in 1.8.2, but in 1.8.3-rc1. Release
1.8.3-rc2 was cut on 1/20/2011, so hopefully the full release will be
out soon.
You can see where the
On Wed, Feb 9, 2011 at 6:55 AM, Vieri rentor...@yahoo.com wrote:
[snip]
Since all of the SIP devices in my LAN have static IP addresses, I can keep
track of
everyone on my own. For instance, could I do fake SIP registrations from
localhost
(the * server) and specify a LAN IP address?
On Tue, Feb 8, 2011 at 8:07 AM, Vieri rentor...@yahoo.com wrote:
Suppose you have 2 identical Asterisk servers and 1 alias IP address that you
assign to either one, according to system failures, etc.
Also suppose that all SIP clients register requests go to the alias IP
address.
This is a
On Mon, Dec 20, 2010 at 9:46 AM, Dovey Forman dovey.for...@idt.net wrote:
I am trying to initiate a call FROM a softphone client to asterisk (either
an internal 4 digit extension call) or an outside line via a SIP trunk.
In both cases, asterisk rejects the call with a 420.
In this case,
On Sun, Dec 19, 2010 at 2:57 PM, Stephen Reese rsre...@gmail.com wrote:
I believe I have made a little headway. I have two outgoing DID
contexts and have changed the GotoIf statement to the extension name.
User One acts as expected and User two now displays unknown when
calling so I believe it
On Sat, Dec 18, 2010 at 4:03 PM, Stephen Reese rsre...@gmail.com wrote:
The host I am working with has two accounts from the same DID
provider. Incoming calls work correctly and dial the appropriate
extensions. This also allows incoming calls to be billed appropriately to
the
individual DID
On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com
wrote:
I am having issues with Blind Transfer on asterisk 1.8
What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS?
Verison 1.8.0, Suse 11.1
Try the latest SVN branch for 1.8 and see if that resolves your issue:
On Thu, Dec 2, 2010 at 6:56 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Thu, 2 Dec 2010, Jonas Kellens wrote:
I have Snom, Cisco, Grandstream YeaLink phones.
Is there a way to push a centralized phone book to these phones ??
Grandstreams support an XML format phone book
On Thu, Dec 2, 2010 at 4:56 PM, Mike l...@net-wall.com wrote:
Hi,
I know I am using SVN, but I was wondering if anybody ever came across this
error.
There is nothing wrong with using SVN.
Well, there isn’t a msg.txt file, I can see that. There is a
msg0003.txt and msg0005.txt (along
On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com wrote:
I am having issues with Blind Transfer on asterisk 1.8
What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS?
If I call from one Grandstream phone to another and us the transfer key
to do a blind transfer
On Sat, Nov 27, 2010 at 11:40 AM, Fabiano Carlos Heringer
b...@grupoheringer.com.br wrote:
Hi, it´s possible to mantain the original CallerId when making transfers?
(atx or blind)
Example: A calls to B, A transfer to C, C see the CallerID of B, and not A...
It´s possible?
Asterisk 1.8 added
On Mon, Nov 22, 2010 at 11:24 AM, Peter Kowalski kowalla...@gmail.com wrote:
Below is my xml button 1 and button 2 portion. Any help will be appreciated.
line button=1
name130/name
authName130/authName
authPasswordpass/authPassword
contact7b452e87-4496-4762-e11f-b26751a1884b/contact
/line
On Fri, Nov 12, 2010 at 10:17 AM, Ernie Dunbar maill...@lightspeed.ca wrote:
that goes from port 4 on the live server to port 1 on the backup server.
In /etc/asterisk/chan_dahdi.conf:
group=4
context=local
switchtype = national
signalling = pri_cpe
channel = 73-95
context = default
On Mon, Sep 20, 2010 at 8:58 AM, Paul Belanger paul.belan...@polybeacon.com
wrote:
On Mon, Sep 20, 2010 at 11:48 AM, Olivier CALVANO o.calv...@gmail.com
wrote:
Anyone have a AudioCodes with Asterisk ???
I use many AudioCode devices with Asterisk. Mostly Mediant 1000s and
MP-114s,
On Tue, Aug 24, 2010 at 5:48 AM, Dan Journo
d...@keshercommunications.com wrote:
Hi,
I think I already know the answer to this question, but is there any way to
do the following using realtime? Or do I have to create a full dialplan for
each client without using includes?
One way that I know
On Tue, Aug 24, 2010 at 7:03 PM, Bruce Ferrell bferr...@baywinds.org wrote:
Hi,
I've gone through the source tree and I don't see a MIB description file
for the SNMP agent in asterisk. can someone point me to it.
There is an asterisk-mib.txt and a diguim-mib.txt in the doc
directory, and
On Wed, Jul 21, 2010 at 3:09 AM, Murali Vasu vimurli@gmail.com wrote:
Hi All,
I am trying to configure asterisk realtime. But i am unable to get the
extensions listed successfully when i type sip show peers in the asterisk
CLI . i am unable to see any failure logs when i do a reload
On Thu, Jul 8, 2010 at 8:30 AM, Jared Terrell jared.terr...@mcc.edu wrote:
# Span 1
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
echocanceller=mg2,1-23
# Span 2
span=2,2,0,esf,b8zs
bchan=25-47
dchan=48
echocanceller=mg2,25-47
# Span 3
span=3,3,0,esf,b8zs
bchan=49-71
dchan=72
On Mon, Jun 21, 2010 at 10:19 AM, Warren Selby wcse...@selbytech.com wrote:
On Mon, Jun 21, 2010 at 12:10 PM, Ken D'Ambrosio k...@jots.org wrote:
Howdy, all. What's the difference between split and combined
firmware, which can be seen at the above link? I've googled to no avail,
I'm afraid.
On Sun, May 30, 2010 at 9:37 AM, bruce bruce bruceb...@gmail.com wrote:
Thanks for the tip. I have been checking those two options. Would you be
able to provide an example of how GROUP or GROUP_COUNT may check for a trunk
usuage?
Here is how I do it. It is based on Asterisk 1.6.1.x, and I
On Sat, May 29, 2010 at 2:02 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
I am looking to use System() function along with some bash scripting to
determine if a Trunk is being used during certain time of the day or not.
Here is what I have in mind. Please guide me if you know a better
On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime achera...@gmail.com wrote:
Hello Everyone,
I must deploy an asterisk system that can support
at least 500 pstn outbound calls.
It's a challenge as it's the first time i'm gonna build such a large
system.
I want to
On Wed, May 19, 2010 at 6:13 PM, Adolphe Cher-aime achera...@gmail.com wrote:
Jonathan for redundancy which software do you recomand?
Without knowing exactly what you are trying to do beside having at
least 500 outbound calls, that would be impossible to say. I mostly
use a home grown HA Linux
On Fri, Apr 30, 2010 at 5:26 AM, Peter peterp...@aboutsupport.com wrote:
I consider buying three GrandStream GXW4024 and connect 72 analogue
phones to asterisk
I recommend against that product. I have two that now sit on a shelf
due to bad call quality, echo issues, and random one way
On Wed, Apr 28, 2010 at 7:58 AM, Tim Nelson tnel...@rockbochs.com wrote:
- Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i want change my asterisk server. Actually, Asterisk work's on a IBM
Server with a internal digium E1 card.
For High availability, i don't want now use internal E1
On Wed, Apr 21, 2010 at 1:09 PM, Robert Grignon rgrig...@fleetone.com wrote:
I am investigating High Availability solutions for my front end servers.
Always good to hear.
I got into a discussion regarding replicated local databases versus
a single fiber connected shared database on an EMC.
On Sun, Apr 18, 2010 at 12:30 AM, Jonas Kellens
jonas.kell...@telenet.be wrote:
Jonathan,
'sip show peers' works just fine...
Sorry, I wasn't clear. It has been my experience in 1.6.1.x that 'sip
show peers' does not work without rtcachefriends=yes for realtime
implementations.
asterisk*CLI
On Sat, Apr 17, 2010 at 4:42 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
Do I need to 'sip prune realtime all' after every change ??
If you change a sip peer and you have caching enabled, then you need
to prune that peer for the change to take effect. On 1.6.1 I issue
the following:
sip
On Sat, Apr 17, 2010 at 11:14 AM, Jonas Kellens
jonas.kell...@telenet.be wrote:
Is rtcachefriends=yes a wrong setting ??
No, not if you want caching enabled. I enable sip realtime caching on all
of my systems.
What if I do not enable caching ? What would be the effect on my realtime
On Tue, Apr 13, 2010 at 11:17 AM, khalid touati khalidtou...@gmail.com wrote:
Hi Guys,
i wanted to share this with u and ask for little help at the same time:
i used iptables to secure my server, so i wnet ahead and blocked avery thing
except a couple of domain protocols and UDP ports of SIP,
On Sat, Apr 10, 2010 at 9:50 AM, James Lamanna jlama...@gmail.com wrote:
Hi,
Has anyone had any experience using DRBD to mirror an entire asterisk machine?
Entire, no. Specific/Important mounts yes.
If so, is there a performance issue at all when people are recording
voicemails and the
On Wed, Feb 17, 2010 at 8:50 AM, Dave Poirier dpoir...@mesd.k12.or.us wrote:
Hello,
We recently upgraded our Asterisk box from 1.4 to 1.6.1. In both versions of
voicemail you can press 3 for advanced options, 5 to leave a message and
enter an extension to leave a voicemail. This feature worked
On Thu, Jan 21, 2010 at 4:56 PM, Matt Darnell mattdarn...@gmail.com wrote:
Most manufacturers charge in excess of $80 to upgrade from a 10/100
switch to a 10/100/1000 switch built into the phone.
The cost might have been in the chipset 5 years ago but I can get a 5
port gigabit switch for $30.
You need to set: host=dynamic Otherwise only .112 is allowed.
-Jonathan
On Tue, Jan 19, 2010 at 1:17 PM, Bruce Ferrell bferr...@baywinds.org wrote:
I'm using realtime sip peers and I need to enable a range of IP
addresses for a peer.
I have:
deny = 0.0.0.0/0.0.0.0
permit =
On Sun, Jan 10, 2010 at 1:17 PM, C F shma...@gmail.com wrote:
Anyone using the above mentioned SIP Gateway made by grandstream?
I would like to hear some feedback on real life experience using this gateway.
I have a few that I used for about 2 days before I replaced them with
AudioCodes
On Sat, Jan 2, 2010 at 4:27 PM, hin lee hi...@yahoo.com wrote:
yes, fxs for my fax machines.
I don't have any experience with the 4004, but I do with the GXW-4024.
I purchased one for a Fax gateway, tested fine, had it in production
for two days and ordered an AudioCodes MP-124 to replace it
The web interface is a bit confusing at first. Here are some of the
steps that I remember off hand. Change as little as possible, makes
it easier to troubleshoot later.
Get the latest code from your vendor (5.6 is what I run)
Configure the proxy to register with
Configuration - Protocol
/27/09 07:56, Jonathan Thurman wrote:
The web interface is a bit confusing at first. Here are some of the
steps that I remember off hand. Change as little as possible, makes
it easier to troubleshoot later.
I did not change much and trying to register just one line first,
but is not easy
On Fri, Dec 11, 2009 at 12:44 PM, Connor Spiess cspi...@idea-ma.com wrote:
Joseph
You could also check out the Audio Codes gateways if the Grandstream doesn't
work out for you. They make FXO/FXS
gateways. They were reliable boxes for us but this was to a non-asterisk PBX
over MGCP. I
On Fri, Dec 11, 2009 at 7:52 PM, Joseph syscon...@gmail.com wrote:
[snip]
Thank for suggestion.
Well, it is not that cheap but the problem with their equipment is luck
support and decent manual.
I actually find the Quick-start guide that comes in the box the most
useful, if you aren't doing
On Fri, Nov 27, 2009 at 11:17 PM, Michael Munger
mich...@highpoweredhelp.com wrote:
In 2007, I released a Polycom Provisioning Tool. I retired the package
earlier this year, and have had so many requests for it, I have revived the
concept, new, improved, and still FREE.
Any chance of you
On Sat, Nov 28, 2009 at 5:22 PM, matthieu Nicaise
techni...@thinkrosystem.com wrote:
Hello everybody,
I'm using Asterisk ( 1.6.1.9 ) Voicemail.
For example, if i call extension *11 which is not registered, from extension
*12, i have no greetings at all, i only have the please leave a message
On Sat, Nov 28, 2009 at 7:34 PM, matthieu Nicaise
techni...@thinkrosystem.com wrote:
I made an error in my first mail, i'm calling voicemail in extensions.conf
this way :
exten = _*.,1,Dial(SIP/${EXTEN:0},60)
exten = _*.,n,VoiceMail(${EXTEN:0},u)
exten = _*.,n,Playback(ss-noservice)
You
On Tue, Nov 24, 2009 at 12:49 AM, Olivier oza-4...@myamail.com wrote:
Hello,
LLDP is more and more available on various network elements (endpoint,
switches, ...).
It seems to ease network configuration.
Makes Voice VLAN assignment much easier for sure.
Do you have any experience with it ?
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian
We have several types of phones, Cisco 79xx, Aastra 9133i etc. We have a
hotdesk type system where anyone can log on to an extension - however what
I would love to do is relabel the phone with the current owner when this
On Mon, Nov 16, 2009 at 7:29 AM, Peder pe...@networkoblivion.com wrote:
I'm pretty sure it only pulls the background image during a reboot.
On a 79x0, yes. On the 79x1 phones the user can change the background
to a list of custom images that you provide. It downloads the image
on the fly, and
On Wed, Nov 11, 2009 at 2:04 PM, Kevin P. Fleming kpflem...@digium.com wrote:
Scott L. Lykens wrote:
Any progress on new Fax for Asterisk modules? Last update here was
October 19 as quoted above; Original discussion is now over six weeks
old. FAA Download Selector still shows modules for
On Mon, Oct 19, 2009 at 3:42 PM, Joseph syscon...@gmail.com wrote:
How hard is to setup Cisco 1751 w/2x FXO with asterisk?
I was googling but couldn't find much information; how to access unit
interface for programming?
I haven't personally used a 1751, but I have used the 1760 series and
On Thu, Oct 15, 2009 at 12:24 PM, Olivier oza-4...@myamail.com wrote:
Hi,
I've downloaded for a demo, a P0S3-08-12.zip file which is suppose to work
with 7960.
Is it supposed to be the same file that the one needed to 7942 model ?
No. The SIP firmware for each model are different except for
destination-pattern .T
What does destination-pattern .T mean? I'm not familiar with what
.T would match. I would suggest using a more specific pattern that
you expect to be coming down the line.
One or more characters (up to 31 characters), waiting timeouts
inter-digit before sending.
I don't have any experience with E1, but here are some comments from
the T1 perspective (on a 2800 series Cisco). Here is also a link to
my collection of Cisco voice debugging commands:
http://thurmantech.com/node/5
On Thu, Oct 15, 2009 at 3:27 AM, Phibee Network Operation Center
n...@phibee.net
On Wed, Oct 14, 2009 at 12:27 AM, jonas kellens
jonas.kell...@telenet.be wrote:
Hello list !
I don't have the money to test out all the products and reading the manuals
is not always that enlightening...
Does someone here know a good gateway-product that lets analogue telephones
communicate
Depends on what the router is. If you get a 2800 series router (we
use 2801s and 2811s for T1s in production with no major issues). You
need the T1/E1 module, DSPs, and an IOS that supports voice.
For a 2800 series you would need something like:
- VWIC2-MFT-T1/E1 ( or VWIC2-2MFT-T1/E1 if you
On Wed, Oct 14, 2009 at 12:57 PM, Julian Lyndon-Smith aster...@dotr.com wrote:
Thanks for the info. I didn't have any model in mind, just wondering
what was required.
If you haven't purchased anything yet, or don't have anything, it
might serve you better to look at other products. While the
On Wed, Oct 14, 2009 at 1:52 PM, Dan Journo
d...@keshercommunications.com wrote:
Hi,
Can anyone recommend a cheap SIP doorphone?
Please only respond if you’ve had personal experience of a doorphone.
I searched around for a while and couldn't find a hardened SIP
external phone. We ended up
On Sun, Oct 11, 2009 at 8:03 AM, James Stocks stoc...@stocksy.co.uk wrote:
OK. For anyone finding this thread, the problem exists in Asterisk
1.4, but upgrading to Asterisk 1.6.1.6 appears to eliminate the problem.
Sorry, I lost your last response in my inbox... Your phone configs
look fine.
Don't use them for Fax... I spent too much time trying to use one for
a faxing ATA. (We went with the AudioCodes MP-124 instead, which
rocks). We to have some analog phones and an analog IVR system hooked
up to one with no issues. They are easy to configure if you just need
to hook up some
On Sat, Oct 3, 2009 at 6:17 AM, James Stocks stoc...@stocksy.co.uk wrote:
Hi everyone,
I hope someone can help me with a problem I'm having with Cisco 7940
phones on the SIP 8.12 image. When I place a call from one of the
handsets, the call proceeds as normal for 20 seconds and is then
I have been working on a HA procedure for Asterisk on CentOS 5.3, but
haven't had time to publish it. It is a little complex, but here are
the components used:
- CentOS 5.3
- Asterisk 1.6 (version doesn't matter)
- MySQL
- Cluster services
- GFS2
- DRBD
A basic run-down is:
* Two servers
On Fri, Oct 2, 2009 at 11:41 AM, Fred Posner f...@teamforrest.com wrote:
* Two servers configured with DRBD in Master-Master mode. All data is
replicated between the two so in case of a failure there should be
very limited data loss (voicemail) if any at all.
If you put the asterisk spool,
I am working on updating to 1.6.1.6 and if I have res_snmp.so
auto-loading on CentOS 5.3 Asterisk Seg faults every time. I can load
the module manually after the initial startup. I am starting to dig
into it further and will open a ticket, just wanted to see if anyone
else knew of any issues off
I have with CentOS 5.3 and custom 1.6.1.6 RPMs. If you use RPMs for
the installation of Asterisk then it's really easy. As for the
Kickstart, if you haven't used it much here I did a quick write-up
with example script here: http://thurmantech.com/node/3
Either use RPMs and add them to the
I have a SIP trunk between CCM 6.1.2 and Asterisk 1.6.1.1 working
without any issues. What does your peer section of the sip.conf look
like? When do you get the error (call direction)?
-Jonathan
On Fri, Sep 4, 2009 at 12:00 PM, Jerry Geisge...@pagestation.com wrote:
Hi all
I have asterisk
You could put something into the Asterisk Database with DBput/DBget.
I don't have an example off hand, but create a stickypark family and
store which channels go back into which parking slot. Or something to
that effect, and it would exist until you remove it from the database.
-Jonathan
On
When I reload chan_sip.so, it seems that connected terminals are no longer
detected by Asterisk because when I tape CLI command sip show peers,
there is no results displayed. Any reflexions about that ?
They won't be found in the CLI command until Asterisk receives another packet
from that
Ideally, the way realtime works, it shouldn't matter at all whether the record
exists in memory or in the database. In reality, there's a few cases where
the data needs to exist in memory for a particular event to occur correctly
(such as device state notifications). I think a better goal
I try to off-load specific tasks like PRI-to-SIP to dedicated hardware for
the task. It is also easier to have centralized call processing and easy to
configure/manage devices in our remote locations. I have colleagues that
use Digium PRI cards just fine. Just depends on your budget and
I am also using them quite extensively, but with English menus. I know that
the Locale files from Cisco do not come with the firmware, but usually as an
update for CallManager. There are a ton of languages that work with the
latest firmware, but I have no idea how to actually get the files from
On Wed, Aug 12, 2009 at 12:39 PM, Olivier oza-4...@myamail.com wrote:
2009/8/12 Jonathan Thurman jthurma...@gmail.com
I am also using them quite extensively, but with English menus. I know
that the Locale files from Cisco do not come with the firmware, but usually
as an update
On Tue, Aug 11, 2009 at 5:12 PM, Jimmy Ezell jez...@hmhca.com wrote:
Sorry for not being real clear.
What I have is 1 front desk phone only with 6 lines
Front Desk Phone line 1 - incoming extension 1
Front Desk Phone line 2 - incoming extension 2
Front Desk Phone line 3 - incoming
Are there any other phones registered, or is it just this phone that is
having issues? The first thing that I see is the qualify=200 line, and I
have not had good experience with Cisco devices and any qualify setting. I
would try leaving that out. I also have double quotes around the line1_*
Huh?
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0-current.tar.gz
is not the same as
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0.10.tar.gz?
Their sha1 files are identical.
sean
I believe he means that:
This was fixed in the 1.6.1 SVN, and I would guess that it was also fixed in
the 1.6.0.
SVN log:
r189951 | russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines
Fix call parking callback. Pipes - Commas.
You will have to create a patch against the 1.6.0 source, but you could
start
On Tue, Jul 14, 2009 at 2:19 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Dear;
I would like to ask: when Asterisk was registering on the gnugk, both
(asterisk and gnugk) were on the same hardware machine and same IP address?
Can they be on the same IP address?
If I understand your
On Sat, Jul 11, 2009 at 12:09 PM, Wayne wa...@planetwayne.com wrote:
Thanks for all for the feedback with this - I'd like to help where I can -
I'm building another 1.6 system for the office to try out the exchange tie
in so if the general consensus is SIP is ok - then that's good for me too
On Fri, Jul 10, 2009 at 4:33 PM, Wayne wa...@planetwayne.com wrote:
Hi Steve,
Thanks for the pointers. I must admit - I was leaning towards 1.6 as
this apparently has support for SIP over TCP (?). My end goal with this
was to try and get Asterisk talking to Exchange 2007 servers unified
snip
Audiocodes supports SRST on their mediapack analog gateways.
This might be a viable option. I haven't used any Audiocodes devices
before. Are people pleased with them?
snip
Deploy a lot of small asterisk based appliances...
This way you can completely decentralise your setup and give
We are currently moving away from a wide-spread Cisco CallManager deployment
to Asterisk. For many of our small sites we have the routers configured for
what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP
registrar. We are converting to SIP, and from what I can tell
On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III
jsulli...@opensourcedevel.com wrote:
On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote:
Hello, all. With the assistance of very helpful folks, our brand new
multi-tenant setup seems to be working smoothly from start to
This has been fixed in the 1.6.1 SVN, and you will have to back port a
patch until these changes are rolled into another release. I was
disappointed that more bug fixes were not included in 1.6.1.1.
-Jonathan
Asterisk 1.6.1.1 was released for a security issue, AST-2009-001. Why
would
On Jun 26, 2009, at 10:44 AM, Tim Nelson tnel...@rockbochs.com wrote:
- David Backeberg dbackeb...@gmail.com wrote:
On Fri, Jun 26, 2009 at 1:31 PM, James Lamannajlama...@gmail.com
wrote:
The use case is that a customer has a fax machine attached to an
ATA.
The ATA sends T38 to
David's directions will work on a 7941/7961, not the 7940/7960. You do have
to keep the line configuration for the 79x0 series phones in the
SIP${MAC}.cnf file.. I have not tested setting them to , but I know if
you telnet into the phone they will show UNPROVISIONED as the setting.
You can also
The phone caches the configuration... To remove it update the config like
so:
line2_name:UNPROVISIONED
line2_authname:UNPROVISIONED
line2_password:UNPROVISIONED
line2_shortname: UNPROVISIONED
line2_displayname: UNPROVISIONED
For each line that you don't want anymore. So on a
-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7941G Auth
Jonathan Thurman wrote:
What does your SEPMacAddress.cnf.xml file look like? In my
experience,
the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I
had
to specify the firmware version in each SEP file
What does your SEPMacAddress.cnf.xml file look like? In my experience,
the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I had
to specify the firmware version in each SEP file. I am using 8-4-4S, but
for you this would be something like this:
device
I believe that 'externpasscheck' was added in the 1.6 branch. Since we use
this, I wrote a quick perl script that checks for password length,
difficulty, repeated digits, etc. which are required for us. If you get it
back-ported to the version you are on you can have the script, just contact
me
On Mon, May 25, 2009 at 2:58 PM, John Novack
jnov...@stromberg-carlson.orgwrote:
sean darcy wrote:
The local telco is now going 10 digit dialing even for local (free)
calls which used to be 7 digit. For a while no problem, everyone will
continue to dial 7 digits, and I'll add the area
From the front page ( http://wiki.centos.org/FrontPage ):
*What is CentOS?*
CentOS is an Enterprise Linux distribution based on the freely available
sources from Red Hat Enterprise
Linuxftp://ftp.redhat.com/pub/redhat/linux/enterprise/.
Each CentOS version is supported for 7 years (by means of
When the phone is plugged back in to CallManager network, it should
get handed a TFTP server via DHCP, and should automatically download
the configuration from CallManager which includes what firmware to
load. It should then reload the SCCP firmware (if you are not using
SIP) and reboot back to
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