(firewall, NAT).
But this behavior is quite odd to me ...
Alberto.
PS: the network is at customer's site, so I haven't chance to have a clear look
over it...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
see a row like this:
999/9991.1.1.1 5060 OK (3 ms)
Can someone explain me this kind of behaviour? Is it normal? Can I restrict
registration of 999 peer only to SIP UA from network 1.1.1.X?
Thanks for your help! Regards,
Alberto Aggio
,
cheers
Alberto.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Sent: lunedì 4 gennaio 2010 22.13
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] No reply to SIP
Hi,
maybe this link can be useful:
http://www.voip-info.org/wiki/view/IAX+encryption
In particular, in your configuration I can't see the authentication method,
which must be md5, and a username to authenticate with, in either server.
But have a further look at the article, maybe you'll be able
-Commercial Discussion'
Subject: Re: [asterisk-users] destroy zombie session
What does the zombie call look like in core show channels?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aggio Alberto
Sent
(i.e. third party called) is registered to Asterisk or not.
HTH
Alberto.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ignacio
Sent: martedì 17 novembre 2009 14.07
To: Asterisk Users Mailing List - Non
no other way to achieve this result :))
Thanks in advance for your replies.
Cheers,
Alberto Aggio.
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Hi,
it's quite straightforward: you can do your dialplan like this (default is the
default context answered when inbound calls happen) - remember the underscores!
-
[default]
exten = _1703,1,Goto(place-IVR,s,1)
exten = _1567 ,1,Goto(place-other,s,1)
[place-IVR]
exten =
asterisk?
Many thanks in advance and regards,
Alberto Aggio
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Hi group
I'm newbie on Asterisk so I followed the Linux Networking CookBook by Carla
Schroder to make my first call. My asterisk box is on a Debian box with an
public static IP. The clients (2) are with dynamic private IP's
I'm using SJphone on a PC and a Linksys PAP2-NA to make calls between
Hi group
I wrote 2 years ago to know if there is some workaround for PacketCable.
Since then I got no answer and now I hope there's something about.
Is there any chance to use Asterisk as softphone with cable modem technology
using Packetcable?
Thanks in advanced
Carlos Bernat
, as many phones are powered by the AC adaptor.
I think I will able to put my hands on an UCM6.1 box very soon
to try that out and eventually grab the xml profiles.
As soon as I get the info I'll surely post it on this ML and on
voip-info too.
Alberto
that handles about 1000
incoming calls per day.
So far the feedback is really positive.
Alberto.
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fashioned rotary phones
into VoIP ones (I'd like to disassemble the ATAs to remove the
boards from the plastic case and to fit them into the phones
after making the appropriate changes to the phones' exterior
to add holes for rj-45 socks and dc power input)
Thanks.
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The Sangoma A500BRX is a 2 to 6 bri pci-x interface,
although I've never tested it
(A500BRECX comes with hw echo cancellation).
Regards,
Alberto
templates)
I have the win32 version for firmware 5.1.15
(spa942-5-1-15-spc-win32-i386.exe)
Below is the sample xml generated by this tool.
If you need additional info (such as the pdf guide or other stuff)
email me privately, as I can't post this on the mailing list
Bye,
Alberto.
Sample XML profile
that on AAstra 5xi phones it is
possible to push this settings on the fly without
rebooting the phones.
Can the same be done with Cisco Unified IP Phones?
Is there a URL inside the phone to which I can
make a GET or POST request from asterisk?
Thanks,
Alberto.
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Alberto Pastore
B-Press Srl - Gruppo
to bridge them internally on the asterisk server:
E1 == TE207 == Asterisk == (some hardware with FXS) == modems
TIA for your replies.
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B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
(of course no mention for that fact anywhere).
Alberto.
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network with 40 cellphones).
Alberto.
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.
...
It works like a charm.
Thanks a lot for the precious hint.
Alberto.
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as human, Customer A must
be bridged to Operator B
Everything is ok from 1 to 5, but I cannot really figure out
how to accomplish task #6
I've tried with MeetMe or call parking but with no success.
Can anyone point me in the right direction?
Thanks
--
Alberto Pastore
B-Press Srl - Gruppo
Folks, please, take a look at this asterisk log message:
[Oct 2 08:55:13] WARNING[10290] app_queue.c: Announcement file
'atcert' is unavailable, continuing anyway...
[Oct 2 08:55:13] WARNING[10290] app_queue.c: Agent on Agent/1001002
hungup on the customer.
but:
-bash-3.1$ whoami
asterisk
--
Alberto Sagredo
RD area
Peoplecall
Email : [EMAIL PROTECTED]
Blog: http://www.voipnovatos.es
Office phone : +34 91 120 5080
Direct phone : +34 91 120 50 39
Peoplecall Network : 700 757 139
Fax number : +34 91 661 9460
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/chris.html
This email is made from 100% recycled electrons
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Alberto
!
Alberto Sagredo (M) ha scritto:
You could try, N80, N95 devices. It cost arround 300 dollars and works
fine with SIP , Wifi and GSM.
I have been trying for several weeks with Truphone, Gizmo, Asterisk and
other providers my N80 IE, and it works perfectly
Regarsd
2007/5/23, Chris Bagnall [EMAIL
ACK
2007/4/12, Razza [EMAIL PROTECTED]:
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Alberto
to server
10.0.10.5:5060, handle = 8 local port= 0
sipTransportSendMessage:Sent SIP message to 10.0.10.5:5060,
handle=8, length=1224, message=
..many other retransmissions follow
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Doug Lytle ha scritto:
Alberto Pastore wrote:
Firmware on 7940 is 8.6 (the latest one).
I had the same issue. I ended up moving back to firmware P0S3-07-4-00
on the phone. I did a telnet into the phone, did a show register and
shaw some very weird info. Normally, I would see
,
Alberto
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is far from being usable...
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B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
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asterisk
Siemens Gigaset IP phones (C450-IP, S450-IP) are not that bad
(gigaset.siemens.com).
C450IP costs less than 100 USD (in Italy at least), S450 is slightly
more expensive.
Grandstream HT-286 works quite well with DECT handsets too.
(I've deployed both and both are working with Asterisk).
Alberto
Andrew Joakimsen ha scritto:
I know of the call pickup issues but what asterisk issue and what BLF
issue?
On 1/25/07, Alberto Pastore [EMAIL PROTECTED] wrote:
Andrew Joakimsen ha scritto:
Actually I noticed just three days ago there is a new release, and the
releae notes seem to address
Andrew Joakimsen ha scritto:
Actually I noticed just three days ago there is a new release, and the
releae notes seem to address
Disable TrMail and Pickup keys
Disable call progress indication
___
but it does not address poor guys' troubles with
only a little bit annoyed about not being able to take
advantage of the onboard DSPs to perform audio transcoding,
because
of the lack of a suitable asterisk driver
(the cards themselves support hardware gsm/g726 codecs,
for instance).
Alberto
to be
manually activated before I can see the price! That's *STUPID*. If I
have a choice, I'll buy it from somewhere else...
BTW, check out on eBay
(search for diva server)
you could make good bargains.
Alberto.
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.
Armin
Wow!
Well, maybe annoyed was the wrong term, as I don't
really-really miss the transcoding feature.
Let's just say it would be better to have it.
Surely, I will test any test release I can put my hands on.
Keep up with the good work!
Thanks,
Alberto
with two DIVA 4BRI-8M 1.0 in it.
Whe you run Eicon Config utility,
the software recognizes instantly all cards models, and numbers
each port sequentially, so that you can access them in asterisk
individually or as a group by properly configuring chan-capi
via capi.conf and Dial().
Alberto
the person that I am calling.
Honestly, I'm experiencing a good audio quality,
no humming noise or hiss. Well, I'm using g711a...
Alberto.
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. With some amount of luck I can try to
change the behavior of chan_sip code
Alberto.
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Olivier ha scritto:
Alberto,
Call pickup is not implemented yet within Thomson ST2030 (1.50 firmware).
More precisely, call pickup current implementation is not Asterisk
compliant.
A new release is scheduled for February (I've got this confirmed by
Thomson 10 minutes ago) but we don't know
() dialplan function...
Do you have a direct contact with Thomson guys?
I've tried to reach them on e-mail or phone
but with no success...
Anyway, thanks again for the NOTIFY call-id patch tip.
That's a new toy to play with for a couple of day before giving up.
Alberto
chan_sip
as indicated on this forum:
http://www.ip-phone-forum.de/showthread.php?p=590842#post590842
No success.
Any help would really be appreciated.
Thanks,
Alberto.
--
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B-Press Srl - Gruppo MSoft
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P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
, to force the device state
for a dialplan hint, e.g. on a fake or local channel,
so that I can map a BLF key on
the phone to that hint?
I have not find anything suitable so far (except for
a PC system tray's icon, which is not applicable to
my situation).
Thanks,
Alberto
On Dom, Dicembre 17, 2006 15:26, Doug Lytle wrote:
[EMAIL PROTECTED] wrote:
Is there a way, for instance, to force the device state
for a dialplan hint, e.g. on a fake or local channel, so that I can map
a BLF key on the phone to that hint?
[turn on mwi]
touch
it'll
compile outside bristuff,
without the need to patch the whole source).
Alberto.
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asterisk boxes.
I'll try to play with app_devstate.c alone (maybe it'll
compile outside bristuff, without the need to patch the whole source).
Alberto.
I'm happy to report that with a very litte change to app_devstate.c
(just in the way ast_device_state_changed_literal() is called)
that module just
Joao Pereira ha scritto:
Do you know if it has 802.1x authentication as it is defined in
EDUroam ( http://www.eduroam.org/ ) ?
I never found a WiFi phone working with 802.1x I tested ZyXel
Prestige 2000 but the sound was bad and it doesnt support 802.1x :(
Thanks
Joao Pereira
Well, I
to set that up, I believe it's highly unreliable.
Alberto.
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:4569]
Thanks in advance for any lead
Alberto.
--
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Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
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,Playback(sorry-dude-youre-not-allowed)
exten = _X.,n,Hangup()
Alberto.
--
--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
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there are 10 phones to check, and the process is not atomic,
it can (very rarely) occur that a phone is included in the dialstring
but has just become busy, and the user gets the annoying call waiting
tone.
Any clue?
--
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B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100
Sipura Profile Compiler is only for ITSPs and agreements does not permit
that
Regards
Andrew Joakimsen escribió:
Does anyone have a copy of spc.exe they could send me?
___
with many phone calls.
My greatest concern is about its poor
exterior robustness... I hope I'll never let it
fall to the floor, it looks like it'll break into
a thousand pieces.
I can say I'm overall satisfied, although I'm
hopefully waiting
for a more stable firmware.
Alberto.
--
--
Alberto Pastore
B
);
//sanity check
if (transfer_timeout = 0) transfer_timeout = 15;
}
newchan = ast_feature_request_and_dial(transferer,
Local,
ast_best_codec(transferer-nativeformats),
dialstr,
transfer_timeout * 1000, // ---
outstate,
cid_num,
cid_name);
Bye,
Alberto
/4xx and gxp2000) with
this method.
My template still lacks the last additions (xml phonebook/screensaver,
daylight savings, etc.) as I could not find the time for updating it yet.
--
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B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321
Despite of monitor-join being equal yes, I get individual -in and
-out files for queue calls. My box runs Asterisk 1.2.10 and I've set
up real-time queues. Does anybody have any idea of what is going on?
Thanks in advance.
Carlos.
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Asterisk Development Team ha scritto:
The Asterisk Development Team is pleased to announce the release of
version 1.2.11 of Zaptel.
Where is it??? The link on asterisk.org is broken...
Also, no Changelog anywhere.
--
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P.le
the call is *not* picked up, and the status line key
gets fast-blinking, and remains in that status,
being unusable, until I reboot the phone.
Any hint?
Thanks.
Alberto.
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Alban ha scritto:
Yes, same channel and same ESSID for all AP's.
Are you connecting each AP to the LAN? Or only one connected, and the others
as relay?
With WDS, you have to keep same channel and ESSID for a good roaming.
If connected to the lan, doing it worked really good for me, roaming
.
--
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B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
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--
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Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
console.
That helped me a lot.
Alberto.
--
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Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
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and
no problems.
Marty
Can I ask you guys which phones are you using?
Alberto.
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Marco Mouta ha scritto:
pls post your misdn.conf as well as extensions.conf
May be i can help.
Sou Português:)
On 10/29/06, *Pedro Silva* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Thanks Alberto!
I tested with telsampl like you said (with various configurations for
de
Alban ha scritto:
I've made some tests with Hitachi WIP3000 and 5000, works really good with
roaming (without authentification). Some parts of the AP in the mesh are
wired (no WDS), some others are not (using WDS), but all use the same SSID
and channel. In all cases roaming was fast, quite not
= 7-8
--
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B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
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Andrew Joakimsen ha scritto:
Are you using WDS? While it won't totally fix every issue, I've found
in my trials that turning off WDS and making sure all the AP were
connected to the same wired network was way more reliable, no more
random unregistartion and issue with registering (still seems
jobs.
Bye,
Alberto.
Thomas Winter ha scritto:
Am Thursday 26 October 2006 23:35 schrieben Sie:
On Thu, 26 Oct 2006, Thomas Winter wrote:
I would recommend the Eicon DIVA Server 4BRI cards. They have a
capi interface which is used by chan-capi (chan-capi.org) and
onboards DSPs for the faxing
bristuff manages layer 1 (not sure if that's a driver
problem, hardware problem or both).
Alberto.
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perfectly working since november 2005, when
we first started this new pbx as a replacement of the old Samsung DCS,
we almost forgot about its
existence, as we did never have to put hands on it to fix
problems (except for some ordinary
maintenance and diva server software
upgrades).
Alberto
live with that.
The major problem is... roaming between cells.
Is that a dream or something that can actually work?
Unfortunately I have to replace a good old
DECT network (I know it'll never compare to DECT)...
Alberto
--
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Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le
, the misdnportinfo seems not able to find any card.
Has any one successfully managed to run Junghanns cards with
mISDN? (there are a couple of serious issues using bristuff
and we've been looking for alternate drivers).
Thanks,
Alberto.
--
--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
=0xB552 to the list of supported pci devices.
I'm not very familiar with kernel drivers, so... good luck to me.
Alberto.
Alberto Pastore ha scritto:
Hi.
I'm trying to run a Junghanns quadBRI card with mISDN drivers.
I'm able to compile kernel mode user mode mISDN components as well
as chan_misdn
Hi Klaus.
I'm not sure about the timer expiry meaning,
but you could use the xlog command (usually found
in /usr/lib/eicon/divas)
Just run it as root indicating which span (1..4)
you want to trace:
./xlog -c 2
that shoud show you layer 1 layer 2
dump
Alberto.
Klaus Darilion ha scritto:
Hi
/listinfo/asterisk-users
--
--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
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whether it's some sort of kernel
related problem (i.e. irq sharing settings etc.) by which
the card loses packets.
Henrik Woffinden ha scritto:
I have the exact same problem on a normal ISDN2 BRI line.
I solved it by having my Telco put layer 1 to permanent.
Best regards,
Henrik Woffinden
Alberto
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--
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Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4
dial out
for 5 to 15 minutes then everything gets back to normal
(no idea about what triggers the return to working state).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alberto
Pastore
Sent: 16 October 2006 17:26
To: asterisk-users@lists.digium.com
with no difference; my
current
settings are like this:
span=1,1,0,ccs,ami
bchan=1-2
dchan=3
span=2,1,0,ccs,ami
bchan=4-5
dchan=6
etc
Any clue?
Thanks,
Alberto
--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321
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Alberto Sagredo
I+D Area (Asterisk // Cisco
Check it!
http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk
Robert LaPoint escribió:
Hello All
Does anybody know where I can find information on configuring Asterisk
1.4 to work with Google talk.
already tried to follow this document but it did not work under 1.4,
so I am just wondering if Google talk is even supported under asterisk 1.4
yet.
Thanks Alberto
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alberto
Sagredo
Sent: Saturday, September 30
Maybe you could try an asterisk forum in spanish in order to get better
results using your native language.
DiegoF escribió:
hola a todos, tengo una duda, ye he resuelto algunas pero otras
llegan, bueno como habia dicho quiero conectar una pbx a una te110p,
la pbx me ofrece señalizacion r2
It has a proxy inside (asterisk), you could register to it as a regular
sip proxy, so you could use it.
Carlos Chavez escribió:
Does anyone know if the Linksys SPA400 is compatible with Asterisk or
is it only for the SPA9000 system? It is interesting because it is a 4
FXO ATA at a
Not True!
You could register against it any spa product, and also asterisk.
Cory Andrews escribió:
It's only designed for use with the SPA-9000 (LVS-9000) product ecosystem.
Cory Andrews
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez
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Alberto Sagredo
I+D Area (Asterisk // Cisco-Linksys)
Peoplecall
Email : [EMAIL PROTECTED]
Blog: http://www.voipnovatos.es
Tel./Ph. : +34 91 120 5080
Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39
Fax./Fax.: +34 91 661 9460
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VAD maybe was caussing this.
Regards
Zeeshan Zakaria escribió:
Actually the problem was somewhere in the Cisco equipment, as the service
provider has confirmed. Some option in their device to conserve
bandwidth by
compressing voice data was causing this choppyness. As they've turned this
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Alberto Sagredo
I+D Area (Asterisk // Cisco-Linksys)
Peoplecall
Email : [EMAIL PROTECTED]
Blog: http://www.voipnovatos.es
Tel./Ph. : +34 91 120 5080
Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39
Fax./Fax.: +34 91 661 9460
I prefer Linksys ones. Spa 9xx series, are great, and provisioning from
Sipura/Linksys is much better than PA1628 (Unencrypted).
Supports https,tftp and http. With Encryption. Vonage use it.
Regards
Thomas Kenyon escribió:
Michael Graves wrote:
Polycom Aastra are both great in this
If you want to answer directly to him, try Reply to all, and delete
[EMAIL PROTECTED] email address.
It is not so had to do.
FRANCISCO PEREZ-LANDAETA escribió:
Hi, this message is for Steve.
Sorry for replying to the digest. It wasn't my intention.
I would appreciate if you can guide as to
Yes you could script a dialplan putting ... and S0 (zero) at the end.
An example :
(xxS0) It will dial 6 digits directly when you enter the 6th.
You could learn how to adapt your Linksys dialplan looking this wiki.
http://voip.wikispaces.com/
[EMAIL PROTECTED] escribió:
Yes that
I think remember there is a readme on /docs that talks about
chan_h323.Check it !
Anyway you could try too at voip.info dot org.
Regards
Wasif escribió:
Hello,
Could anyone tell me how to install/configure H323 with Asterisk 1.2.11 .
Thanks
Wazb
I have 30 SPA 941 phones with Asterisk 1.2.7.1 and 1.2.9 and it works
fine. Are you canreinvite=yes ?.
I have not been notice any problem related to transferring calls (blind
and attended)
Regards
Dan Serban escribió:
I have a system running Asterisk 1.2.10 (Debian packaging) and about 50
I use canreinvite=yes in my config files, and it does work, so maybe its
a spa 941 misconfiguration.
I think if nat=no sometime it has problems if you are behind NAT, but
under same network it must not fail.
Which firmware are you running on spas?
Dan Serban escribió:
Alberto Sagredo wrote
This Guide is offered as i know only to ITSP and large distributors not
to end-users.
You could find a User Guide for SPA 3102 at Linksys Website.
Regards
Marcos Rubino escribió:
Anybody have a recent copy of the Admin Guide (not the
user guide) for the SPA3000/3102? The only one I was
able
Nunes
[EMAIL PROTECTED]:
Hi Carlos!!!Let me ask one thing... ... r u brazilian???Becouse I work with * projects and if u r in brazil maybe i can help u. But about your problem, What are u using to call thru *? IP Phone, softphone? What is your
sip.conf settings? Carlos Alberto Bernat
Hi Group!Still I'm concern about my problem with echo on the voice and I want to ask some advice to developing VoIP. Maybe I'm very ambiciuos or maybe not because I want to give VoIP to near 500 users.We got an small ISP and we have the project to give telephony (for now) to our users between
Hi Group!Thanks Joshua Colp for your answer!!. I thinking (aprox. ) 50 simultaneous users up at the beginning. Some of our users, most, have private adresses, automatically assigned by dhcp. Some other, very few, has the public ones.
I was thinking on hardphones because the idea for the user to
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