I confirm too, Sipura devices have flawless g729a codec. Tested
personally the Sipura-2100, 3000 and 841 hardphone models - all work
with Asterisk 100% straight out of the box, even with chan_sip's
not_so_100%_rfc3261 behaviour. I think the sipura-1001 model is the
stripped-down 1 fxs port
Hi,
For madplay, install it, then put this into your musiconhold.conf
(adjusting the paths, of course):
[classes]
default =
custom:/usr/local/share/asterisk/mohmp3/,/usr/local/bin/madplay -Q -z
--fade-in --mono -R 8000 --output=raw:-
Subjectively, the quality is a little worse than with
Hi all,
After googling around and searching both * and xten archives, I was
still unable to find a working pair of softphone/usb *handset* that work
with both keypad operating the softphones buttons *and* working incoming
call ringer on the handset. I'm hoping that, while being OT for *
Thanks a tons Kerry, I got spa-1001, 2100, 3000, 841 and other stuff...
I totally agree with you, but one of my customers still insists on
having a handset connected to his laptop as he doesn't want to have
additional devices.
Any other hints?
Kerry Garrison wrote:
Here is just my personal
The login and password are voip/voip
Miguel wrote:
Where I can find the firmware for the Wellgate 3804 ?
The files are:
- 2m4sipfxo.103
- 4fxosip.103
I don't have a password to pick up it at the welltech site.
Kind regards,
Miguel
___
Mendoza wrote:
Vahan,
Firmware 103 is working for you?, Not for us.
Pls advise.
Jorge Mendoza
Vahan Yerkanian wrote:
The login and password are voip/voip
Miguel wrote:
Where I can find the firmware for the Wellgate 3804 ?
The files are:
- 2m4sipfxo.103
- 4fxosip.103
I don't have a password
http://lists.digium.com/pipermail/asterisk-users/2004-August/059869.html
Paul Rodan wrote:
Using FireFly, all other codecs but G711 Ulaw is selected. But whenever
I place a call, I get:
Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband
DTMF is not supported on codec G.711
Was wondering if there are any news on the native MoH patch for
1.0.3/1.0.5.. or this still works on CVS HEAD only?
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Head, Research Development dept.
adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia
Sergey,
You should really revisit MySQL.com :) 4.0.x is way outdated...
Regarding the high load etc... how about this copy-pasted excerpt from
phpmyadmin?
---8
This MySQL server has been running for 19 days, 6 hours, 8 minutes and
28 seconds. It started up on Jan 07, 2005 at 02:21 PM.
Hi,
This is a confirmed bug with Welltech 38xx sip fxo and 35xx sip fxs.
Their SIP stack is not following SIP RFC and is using same CallID for
all ports. Welltech claims to have a new fixed firmware for 35xx SIP FXS
device, but they're still working on firmware for 38xx SIP FXO devices
(since
Greetings,
For the past 2 months I've been struggling with registration problems
with asterisk+external FXS/FXO gateways (www.addpac.com) that use
RFC3665 re-registration procedure.
This problem occured for devices with more than one FXS port with a set
non-empty password.
Those gateway
Wilson Pickett wrote:
I'd like to know what's most reliable
configuration for BudgeTone 101 in
snip
The .16 firmware is beta and it has been found to work poorly for
several people, including me. I went back to .5.11 I would try to
check that first
Exactly, .16 has several bugs like
[EMAIL PROTECTED] wrote:
On Sun, 14 Nov 2004, Vahan Yerkanian wrote:
Exactly, .16 has several bugs like message button not working, but .5.11
has a *nasty* bug with not-reregistering after the timeout period, which
leads to phone not ringing on incoming calls - you have to power cycle
Jean-Denis Girard wrote:
Well, 1.0.5.16 is the official version on the grandstream site:
http://www.grandstream.com/y-downloads.htm. I only installed it last
friday, so I'm not sure it is better or worse now.
I was using 1.0.5.11 before, and was not aware of the non-reregistering
problem, which
As an additional info, G723 is like 1700 bytes per second, GSM is like
4600-4800 bytes per second as viewed by netstat - those numbers are with
ip overhead. I've been able to use a dialup as a link to get 2
simultaneous G723-based sip hardphones to * server.
kido noagbodji wrote:
Hi Hammoud,
further digging in the firmware reveals fm.grandstream.com/gs which has
some more files including .17
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Head, Research Development dept.
adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia
As easy as having something like this in your extensions.conf
exten = 555,1,Answer
exten = 555,2,MP3Player(http://www.yourfavradio.com:port/)
exten = 555,3,Hangup
Roy Sigurd Karlsbakk wrote:
Anyone been able to integrate say ICECast or Shoutcast broadcasts into
their MOH... I guess if you used
Was showing only the use of mp3player stuff to get a shoutcast stream.
if you want this in moh,
do the following:
1. create a sep directory inside /var/lib/asterisk or whatever you have
configured for that, f.e. /var/lib/asterisk/mohmp3-radio, then
2. touch
R A wrote:
the problem is this:
i plugin the phone but it never wake up.
there is something to do
Yes, search archives, I've previously given recovery instructions.
thanks
wert
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Head, Research Development
g723.1/gsm codecs?
P.S. 3502A is also affected by the registration bug, if you connect a
phone to TEL2 jack and call someone, everything goes by the TEL1's
account/password...
regards,
Vahan Yerkanian
___
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[EMAIL PROTECTED]
http
I've setup the voicemail that auths against the mysql db. Now,
everything works ok, except voicemail() calls fail with
Sep 26 18:09:34 WARNING[157070336]: app_voicemail.c:1517
leave_voicemail: No entry in voicemail config file for ''
all my users are in 'sip' voicemail context, but adding
Henry,
exten = 451,1,Answer
exten = 451,2,Wait(1)
exten = 451,3,VoiceMailMain([EMAIL PROTECTED])
exten = 451,4,Wait(1)
exten = 451,5,Hangup
is what you're looking for.
VoiceMailMain(${CALLERIDNUM}) might be enough if you're not storing your
voicemailboxes in mysql
regards,
Vahan
Henry Devito
time.
All that I could do is collecting the info from the client side and checking
out the latest version could resolve it or not.
If the bug is about the SIP stack such like the communication between Proxy,
that will take a long time.
no comments.
Vahan
Dinesh Nair wrote:
On 06/10/2004 17:29 Vahan
the number of people affected with the firmware bugs.
If you are one of those unfortunate to buy their buggy products, pls
send an email to [EMAIL PROTECTED], May Lin, and voice your concerns!
regards,
Vahan Yerkanian
___
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[EMAIL
Try connecting the phone with crossover utp cable to a computer, set the
ip number of that computer to 67.153.142.69 and 192.168.0.159, setup a
tftp server on it, and on the next reboot the phone should get the
firmware or you'll be able to sniff more info on what it wants.
regards,
Vahan
dean
One at a time, as X100P is to be connected to a single PSTN phone line
with a RJ-11.
christophe de coninck wrote:
Hey,
I knew that info already but the question i ment to ask was: how many
calls will I be able to make to the outside from my asterisk server with
one X100P card, only one at a
Greetings,
Is there a way to add artificial delay to the rtp stream? Due to
regulations in our country, it is required to add 400ms delay to *some*
VoIP calls.
Is this possible with any module?
regards,
Vahan
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global
1.0.5.16's Message button is bugged, it only sends 'INVITE:' instead of
the full INVITE message. 1.0.5.11 is the latest fully usable firmware.
Gary White (Network Administrator) wrote:
Can anyone tell me how to get the Granstream Message
Button back workig after upgrading to Firmware 1.0.5.16.
Todd Reese wrote:
Hi all,
I have just install and licensed Cepstral's Allison08kHz on my Asterisk
1.4.11 system.
I can call the Allison's extension from my Grandstream IP Phone and she's
clear as a bell, but when a call to her extension traverses through one of
the Linksys/Sipura 3102 or
Andrew Kohlsmith wrote:
On Saturday 29 September 2007 18:43:59 Andrew Joakimsen wrote:
That's horrible. I don't buy too many IP phones these days, but can
anyone suggest a place better than the scumbags at VoIP supply?
I don't know about you, but I've had nothing but very good results with
Martin Joseph wrote:
Any feedback on this brand and in particular on doing business with
WelltechUSA?
Don't worry they're OK.
I am looking to the Wellgate 3701A which is a 1FXS-1FXO arrangement. I
am hoping to replace the near worthless Grandstream HT-488.
I'd personally recommend to get
Greetings,
What is the recommended settings for using SPA-3000's FXO port for
dialing out to PSTN in regard of the DTMF?
The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports
registered to the Asterisk box with unique username/passwords.
The inbound PSTN DTMF works
Actually,
I believe that something is wrong with the way asterisk implements the
whole rfc2833 in rtp.c , moreover, the default value of 100ms in
dtmf_tones[] in do_senddigit() inchannel.c is to short to be detected
for lots of commercially available fxo gateways.
This was reported several
Adding to what I already said, just tested that Asterisk indeed doesn't
translate between inband and rfc2833, here is the setup:
Analog phone-SPA-3000fxs(g729,rfc2833 friend in sip.conf)-Asterisk-
-anotherSPA-3000fxo(g711alaw,inband friend in sip.conf)-PSTN
Calling from PSTN, and into the FXS
Alan Ferrency wrote:
We are mass provisioning Sipura 841's as well.
After a normal reboot, our phone retains its previous configuration, so
as long as that configuration has not changed, we're fine.
However, if we do change the configuration, we have the same issue
that you have: it does not
Tomislav Parčina wrote:
Does anybody use UTStarcom F1000G Wi-FI VoIP phone?
http://www.utstar.com/Solutions/Handsets/WiFi/
I'm planning to buy one and I need to know did you have any problems with
phone. What is the sound quality? How close you need to be to the access point?
Please, any
Vladyslav wrote:
Just my couple notes on spa3000 and PSTN DTMFs.
Such schema:
PSTN - SPA3000 - Asterisk
Have problems with DTMF detection on incoming calls
when call comes from cell phone. Once per 4 times it
misdetect some ditigs (whether first digit will be
doubled or unrecognized at all).
wendell hamilton wrote:
Ditto on the volume issue, it's EXTREMELY low, but the latest firmware does allow login to web-portal protected billed wifi systems.
You don't want web-portal based auth schemes for your metropolitan wifi
network, as someone can use your network as a transport with
C F wrote:
On 3/2/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Can you please explain this?
In my case, how would I do this:
|9,:1xx|
|lt;9,:gt;xx|
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Hi,
I've been trying to setup hinting recently on 1.4.20.1, and was
wondering if there is a more elegant way to do the following
piece of dialplan without repeating the hints for every existing
extension/user?
context Main {
hint(SIP/10301) 10301 = call(${EXTEN});
Vincent wrote:
Hello
I'm running Asterisk 1.4.20.1 on a FreeBSD 6.3 host, and unless I'm
mistaken, it seems like /usr/local/etc/rc.d/asterisk script doesn't
make use of /usr/local/sbin/safe_asterisk to restart Asterisk in case
it crashes.
Is this correct, and if yes, why not use it?
Ricardo Melendez wrote:
Hi to all, I actually have an asterisk server configured to write CDR
mysql records in the same machine (localhost), but I want to write
this records to another machine also in mysql at the same time, It is
possible? It means that I want save the records in both
Hi,
We are processing lots of calls and I want to filter these that have
incomplete numbers sent
with a proper SIP response. These numbers are not in the local dialplan
by themselves, so
I'm trying to find a way to generate 484 Address Incomplete SIP
response based on the
length of the
Abel Monzon wrote:
and then in my softphone I call to 1 the asterisk log say this:
-- Launched AGI Script /usr/local/share/asterisk/agi-bin/a2billing.php
== a2billing.php: Failed to execute
'/usr/local/share/asterisk/agi-bin/a2billing.php': No such file or directory
--
Ronaldo Z. Afonso wrote:
Hi all,
What does (T) mean on the output of iax2 show peers?
The following my output.
darkstar*CLI iax2 show peers
Name/UsernameHost Mask
Port Status
ronaldo (Unspecified) (D) 255.255.255.255
0
I find the idea very interesting and quite useful.
There is an ongoing effort on Wikipedia to gather as much information as
possible, and keep it current:
http://en.wikipedia.org/wiki/List_of_country_calling_codes
Here is the data for Armenia (phone numbers are 11 digits):
Wolfgang Pichler wrote:
Or can anyone here tell me where to get good (and not to expensive)
2.5mm plug connection binaural headsets ?
Ebay might be a source for these:
http://shop.ebay.com/items/?_nkw=2.5mm+to+3.5mm+adapter+headphone
___
--
Just got a reply from sipura support confirming the problem and
recommending to use this firmware:
http://www.sipura.com/download/temp/phone/spa841-03-01-03-a-vol-fix.zip
while they're fixing it and until they release the version 3.1.4
thumbs up for their fast reply.
Hugh L. Johnson wrote:
Anyone else has problems with CVS HEAD's from today with voicemail
hanging up silently without any debug/error messages when checked?
It also keeps insisting that the user's voice mailbox is full and can't
store more messages even if I clear/rebuild the
/var/spool/asterisk/voicemail stuff.
! ;-)
I guess someone posted a bugfix a few mins ago and I just picked it up! ;-)
cheers,
Mark
On 7/14/05, *Vahan Yerkanian* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Anyone else has problems with CVS HEAD's from today with voicemail
hanging up silently without any debug
Greetings,
Since the past week I've started receiving the following warnings on my
asterisk servers (FreeBSD / CVS-HEAD). This warning manifests itself
with x-lite/x-pro/eyebeam clients as well as sipura devices.
All of them have qualify=yes in their settings.
Jul 18 22:52:01
Hi all,
Today I've stumbled upon a very strange behaviour with an analog fxs/fxo
gateway (AddPac AP1002, http://www.addpac.com/english/AP1002.html)
connected to a CVS HEAD(from today) Asterisk server. This manifested
itself after enabling the CallerID on the pstn lines connected to the
FXO
Try reinstalling sox - it is responsible for mixing the caller and
callee channels. Also, if IAX2/4506:[EMAIL PROTECTED] is your
real username and password, change them asap, you just made it available
to 1+ people and the archives ;)
Regards,
Vahan
Eric Smith wrote:
We are using the
Yes, sipuras work well in Russia.
Actually, they're so configurable that I think they'll work everywhere.
You'll need to re-configure to make them detect/generate Russian tone
standard.
snacktime wrote:
Will sip/iax devices designed for European use also work in Russia?
I'm specifically looking
http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150tid=147tid=136
http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Head, Research Development dept.
adr:;;28, Isahakian ave., PO BOX
Greetings,
I'm experiencing the same problem. It manifests itself mostly in noisy
environments - as soon as there is some increase of the ambient noise
the volume in the headpiece or the speakerphone decreases immediately,
and starts to randomly increase/decrease for some time after the
Try with dtmfmode=auto and DTMF Tx Method: InBand+INFO, this was the
best configuration for me, although still not 100% guarantee. If the
dtmf tones are sent very fast without a 1 sec delay, in most of the
cases asterisk won't detect half of them. There are a couple of patches
for the trunk
Tofik Suleymanov wrote:
Thanks all for replying
tommorow i'll try to do like you suggested.
One more quick question: why Sipuras cant do more than 1 g.729 channel
at a time ?
Insufficient CPU power to process 2 g729 streams.
Is this somehow related to g.729 licensing ? Is there any other SIP
Tofik Suleymanov wrote:
1. assume 1-st line is in use
2. after dialing 2-nd line from outside i immediately go to the
voicemail announcement (also i immediately go to voicemail if i dial
from extension to extension both of which are on the same sipura device)
Check what response code
This is a known issue with Asterisk's implementation of DTMF detection.
There are two bug reports open up on bug tracker. Currently the best
combination is to set DTMF TX method on Spa3k to INFO and auto on
asterisk side. Works 95%, skips digits if you press buttons on the FXO
end too fast.
Scheda wrote:
I'm sure this has been asked a million times. Therefore, I must ask
again. Generally speaking, what do you guys think of it. It looks pretty
good, but for my uses, I'm not sure that a calling card method is the
*best* way to go. But, either way, what is the general concensus?
Andrew Kohlsmith wrote:
On Thursday 04 May 2006 11:31, Louis-David Mitterrand wrote:
I've got this low-ping 100%-up dsl connection between two asterisk
1.2.7.1 servers. And oftentimes one of them would declare its opposite
UNREACHABLE.
Same, here, two asterisk 1.2.7.1 boxes connected to the
On Thu, 4 May 2006 23:29:37 +0200, Louis-David Mitterrand [EMAIL PROTECTED]
wrote:
On Thu, May 04, 2006 at 10:31:17PM +0500, Vahan Yerkanian wrote:
Same, here, two asterisk 1.2.7.1 boxes connected to the same switch...
Over a week I see at least one case of one of the boxes becoming
Julio Arruda wrote:
Anyone tried the new PSTN/FXO port in the new SPA 3102 FXO/FXS adapter ?
From a quick test (got mine yesterday), seems like it is not
recognizing Caller ID from PSTN/FXO port..
It's a known bug in the current firmware.
HTH,
Vahan
Erick Perez wrote:
should I use mpg123 with asterisk 1.2.7 or should i use the native
player asterisk has?
the target machine will receive heavy load.
mpg123 was used back when asterisk didn't have native format support. If
you are expecting heavy load, the native format is the way to go. You
exten = _2,1,Answer
exten = _2,2,Wait,2
exten = _2,3,DeadAGI, a2billing.php
exten = _2,4,Wait,2
exten = _2,5,Hangup
I tried it and the call is answered bu Asterisk and never dials the
destination. :(
Yes that's the correct way
Doug Crompton wrote:
I am using an SPA-3000 3.1.10d
When I have transfer enabled - 'T' in the dial string I cannot reliably
send DTMF keys to a bank, voicemail, or other service requiring tones. If
I disable (remove transfer option) from the dial string all is fine. I
would like to be able to
Doug Crompton wrote:
Two other Digium bug reports on this issue. It sure looks like it is an *
issue and rather complex??? Any hope for a solution??
http://bugs.digium.com/view.php?id=5970
http://bugs.digium.com/view.php?id=6027
It is an * issue, tested and confirmed. Makes any PSTN side IVR
Hi all,
Does anyone on the list have any experience with this piece of hardware?
It looks to be another way of bridging * with existing wirings / pstn lines.
here is the url:
http://www.soundwin.com/s2400.php
P.S. it claims to have t38 support too.
regards,
Vahan
I'd recommend using native mp3 support that is available in CVS HEAD, as
madplayer mp3 decoder gives a lower quality sound (audibly more
cranky/noisy).
Vahan
Jason Becker wrote:
Steve Totaro wrote:
Anyone know how to get around this? I am stumped.
# make mpg123
[ -f mpg123-0.59r.tar.gz ]
Jason Becker wrote:
Had to do some digging to find out what you were talking about - I guess
you are referring to the section Using native Asterisk format_mp3 for
Music on Hold* found here:
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf
Some of the comments suggest that
Rafael R. GV wrote:
Hi
I was using a2billing with mysql-4.1.12 and php-5.0.4 very successfully
(thanks to areski for this great project and its invaluable assistance
to solve some issues in my last installation...) now I´ve upgraded mysql
to last release 5.0.15 and, without changes in
Rafael R. GV wrote:
thanks Vahan
you are right, I have changed 'call t1' for 'calls t1' in balance.php
and invoices.php files and then tried to create a new table named
'calls' but mysql 5 has changed syntax for 'TIMESTAMP DEFAULT' and this
is the error:
- starttime TIMESTAMP
Matt Riddell wrote:
Patrick wrote:
Hi all,
Can anyone please tell me which format music needs to be in for native
MoH if my local phones use alaw/ulaw and some gsm g729 connections
that come in through the Net.
You can have all the codec versions of the moh file. Asterisk shall pick
the
) default NULL,
PRIMARY KEY (`id`)
)
hth,
Vahan Yerkanian
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John Fraser wrote:
does anybody know what i am doing wrong? help please
gzip: stdin: unexpected end of file
tar: Read 8572 bytes from Open_A2Billing_version_Raccoon.tar.gz
tar: Unexpected EOF in archive
tar: Unexpected EOF in archive
tar: Error is not recoverable: exiting now
Rafael R. GV wrote:
Hello
1.- I am testing a2billing in a SER-Asterisk implementation but using
Mysql versions 4.1.15 and 5.0.15 because I want to integrate its cdr
with some tables from Ser database, the a2billing-mysql-schema does not
work properly in mysql-5 and in 4.1.15 works well but
Rafael R. GV wrote:
Hi
I have this 3 warnings running a2billling with asterisk new version:
a2billing.php|2:
-- AGI Script Executing Application: (SetLanguage) Options: (en)
Nov 18 12:06:19 WARNING[17440]: pbx.c:5435 pbx_builtin_setlanguage:
SetLanguage is deprecated, please use
Also try executing /var/lib/asterisk/agi-bin/a2billing.php from the
shell, most probably the path to php-cli is wrong or you don't have it
installed at all.
Jose M. Ramirez wrote:
Hi list, all. Please, I need help. Although already I installed
a2billing, simply I cannot initiate its
What is the proper way of adding hints to multiple extensions?
In my case extensions are the same as the sip usernames, while as per
http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence
exten = 1234,hint,SIP/1234 works,
exten = _1,hint,SIP/${EXTEN} doesn't. Not sure if I can
Done. Not sure if picked categories under SIP Mantis correct but here it
is: http://bugs.digium.com/view.php?id=5149
VY
Olle E. Johansson wrote:
File a bug report if it does not work. I think it would be a good idea
if it works, even though I usually don't recommend using the extension
as
Dear Matt,
Thanks for your great work and the effort documenting the whole process.
I'm sure the whole Asterisk community benefits from this kind of work
and it's really something to end up in the wiki.
Thumbs up!
Best regards,
Vahan
Matt Roth wrote:
List members,
My previous post
Andy Hamilton wrote:
Anyone have good words to say about any of the WiFi handsets currently
available?
The UTStarCom F1000 (an 802.11b device) works pretty well. It's about
half the $$$ of a Cisco 7920 (which are also pretty nice), but it
seems like most of the config is done from the keypad.
Massimo De Nadal wrote:
Works perfectly out of the box, almost for my customers :-)
The only note is to disable echo training.
Could you please elaborate which exact model you're using and what are
your opinion about the echo can/training quality? Have you tried spandsp
faxing?
Thanks in
Stay away from Grandstream and AddPac. These are some of the companies
with undereducated software developers that have problems with
understanding written english, mainly the SIP RFC documents. I learned
this the hard way, wasting half a year with helping them fix problems
which shouldn't be
Go with SPA-3000. While it's much more awkward to maintain, they're rock
stable and provide the features they advertise for. I'd also add AddPac
VoiceFinder series as being not 100% asterisk compatible, expensive and
not worth your time (learned this the hard way). It took me 6 months to
welltech... last time i tested their fxo 4 port gateway like year ago
all ports were trying to communicate using same Call-ID.
[EMAIL PROTECTED] wrote:
Sorry, but I don't remember the name of this chinese company.
I have meet it once time at a Cebit exhibition at Hannover in Germany few
Hi,
I'm looking for a full list of xml provisioning variables of the
SPA-2100/3000. Currently the Sipura website has example XMLs only for
the SPA-841 [1] and SPA-941 [2].
I'm mostly interested in the CallerID type selector variables and
whatever variables control the PSTN-VoIP settings.
http://www.sipura.com/support/spa941faq/ has the sample xml for
provisioning.
Mark Wiater wrote:
I asked the [EMAIL PROTECTED] for the documents and the tools that
are referenced in the admin guides and was told that I had to become
a registered user in the support section of the ww.sipura.com
Greetings all,
I'm running Asterisk 1.2.9.1 installed from /usr/ports/net/asterisk on
FreeBSD 6.1-RELEASE.
I'm experiencing a guaranteed asterisk core dump with any Sipura device
set to forward all calls to an extension that is mapped to a queue:
-- Executing Macro(SIP/10040-4c43,
Jeremiah Millay wrote:
I'm trying to provision some spa-942 phones via TFTP. The phones get
their address from a dhcp server which sends it option 66 (address of
the tftp server). After spending some time with the phones and even
breaking down to sniff traffic from the phones I see that they
On 2/25/10 6:50 AM, Tilghman Lesher wrote:
DHCP is designed in such a way that you can legitimately have multiple DHCP
servers on the same network. The first DHCP server which replies and meets
the DHCP client's requirements will be the server to which the client
registers. If the Linksys
On 4/15/10 1:26 AM, Tonty T wrote:
That's is all the overhead I am trying to avoid. What I need is a DID
with unlimited channel, but they do not offer DIDs in that country. I
wanted to know for example when I get a DID from lets say Vitelity,
with unlimited channel, what are they using to
On 4/16/10 3:15 PM, mosbah abdelkader wrote:
Hello,
Is it possible to transfer ss7 signaling over an FXO interface.
I need to setup an ss7 test system composed by two Asterisk based
IP-PBX systems with anlog interfaces only (FXO and FXS). I want to
know if it is possible to connect the
On 6/17/10 12:49 AM, Steve Edwards wrote:
On Wed, 16 Jun 2010, Landy Landy wrote:
I'm unable to place any calls through a2billing. I followed instructions
here: http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/F.A.Q to
DISABLE PIN number request Prompt for some users but, I'm not
On 9/27/10 8:57 PM, Michelle Dupuis wrote:
HAAST runs a sync script a regular intervals (time to sync data prior to a
failover check etc)
HAAST includes a sample script which syncs voicemail (and config, etc) files
using rsync from master to slave. After a master/slave reversal the
On 4/14/11 1:04 AM, Shaun Ruffell wrote:
On Wed, Apr 13, 2011 at 01:00:36PM -0700, Jian Gao wrote:
Centos 5.6 came out. Any one tried to update to the 5.6 yet?
I am running Asterisk 1.8 and is there any risk to upgrade to Centos 5.6?
I'm not sure about Asterisk in general, but if you use
On 4/14/11 5:03 PM, m...@tdiehl.org wrote:
On Thu, 14 Apr 2011, Vahan Yerkanian wrote:
A word of notice: asterisk/digium yum repos xmls haven't been updated
yet (properly):
Yes, I noticed that also. For some reason the latest Dahdi rpms are
sitting in
the top level dir at http
On 4/14/11 5:03 PM, m...@tdiehl.org wrote:
Yes, I noticed that also. For some reason the latest Dahdi rpms are
sitting in
the top level dir at http://packages.asterisk.org/centos/5/current/
but they are
not signed. They need to be signed and moved into the approiate arch
directory
and the
Has anyone used this board as an Asterisk server?
http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=HIPMI=Y
I'm mostly interested about the possible compatibility issues this board
may have with the AEX800 card.
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