On Tue, Dec 10, 2019 at 11:01 AM Alexander Perkins
wrote:
> Hi All. Does anybody know if Google/Android has an API I can sign up for
> that will allow us to query the caller ID and find out if it is spam or a
> robocaller?
I don't think that there is a public (free) API. All robocall
protectio
On Thu, Apr 12, 2018 at 11:43 AM, Antony Stone
wrote:
>> A few seconds after registration, the Digium phones will become
>> UNREACHABLE. Right after that, the entire VoIP network (where the
>> Digiums are located) will be also dropped - all other devices
>> (non-Digium) connected will be kicked fr
I'm trying to solve a mystery for the last couple of days.
I have a mix of D70, D50 and D40 behind NAT. Server is in a
colocation, not a VPS.
For several years, everything was working fine, no issues. A few days
ago I started having problems at one particular site. NO CHANGES have
been made to th
Jeng Yu wrote:
> I would like to hear if anyone out there in Asteriskland has used the
> Dock-N-Talk (DNT) box to connect cell phones to Asterisk box.
The only problem I noticed is that after a random amount of time the box
will lost contact/synch with the cell phone. I'm using DockNTalk for
about
Arun Kumar wrote:
> I've upgraded my server to asterisk-1.2.22 from 1.2.10 after that my
> DeadAGI scripts are not working properly. Like after hangup I used to do
> some more work now its not working.
Try, at your own risk, this:
http://svn.digium.com/view/asterisk/branches/1.2/res/res_agi.c?r1
Michelle Dupuis wrote:
> We're looking at a large wifi phone deployment, and we're looking for
> wifi phones that:
>
> 1. Are SIP compliant (Asterisk friendly)
> 2. Provision capable (ideally TFTP of own MAC address)
> 3. Industrial quality (no cheap plastic stuff).
> 4. Well documented (and non
Matt Brown wrote:
Does anyone have any experience with a GSM card, preferably Quad Span
(4 GSM modules or higher) for use in the UK. I have seen the
Junghanns* version but I am not keen on the limitation of having to
use a BriStuffed version of Asterisk.
I'm buying this one to test:
http://www.
Crazy Boy wrote:
If IPhone is released in India, Can you tell me any Apple authorized
showroom in Hyderabad (Andhrapradesh, India)?
Oh gosh... another troll... Google IS your friend:
http://www.google.com/search?q=apple+iphone
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Julian Lyndon-Smith wrote:
however, I get no errors, but still get the default Allison sounds
for the digits. Anyone got any clues on what I'm doing wrong ?
1) Create a directory named "your_country_iso_code" (AR|MX|ES|ETC) [1]
under the main sounds directory (/var/lib/asterisk/sounds/ ???);
Rizwan Hisham wrote:
is there anyway i can set SIP_HEADER(To) to the value i like?
If voip-info is correct, you can read, but you can't change.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
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Khaled Chehab wrote:
How to configure cisco 7905 with asterisk ,if you please can send me
step by step configuration steps .
This electronic message and its attachments are solely addressed to
the addressee(s), and contain confidential information protected from
disclosure belonging to Xplor
Farooq Ahmed wrote:
And any idea about the issue on card one... means why outgoing is not working.
Not quite sure if Traverse Technology Netjet ISDN-s will really work.
Last time I had to use a ISDN BRI I bought one with Cologne chipset and
used bristuff. Worked like a charm...
__
I'm having trouble to send calls to a Mera MVTS softswitch (with SIPHIT)
when the asterisk box has a dynamic IP address.
If the Asterisk box has a fixed IP, everything is OK.
Any ideas? I'm looking for a working sample of the sip.conf in this
case... user.cfg (for MVTS) is also appreciated if
Richard Klingler wrote:
Has any1 got their 7970 to work with * 1.4.x ?
Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2
without problems (Asterisk 1.2.16). Just remember that 7970 only will
register if your Asterisk is at the same network - no NAT between them -
check http://
Josu Lazkano Lete wrote:
I need to download the sources or just with apt-get install is
enought???
apt-get is the easiest way, but won't give you the latest release.
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Yuan LIU wrote:
Just noticed that no matter what the line condition is, zttool always
reports "OK", so it's pretty useless. (In contrast, I'd get "Red alert"
if I unplug the line connecting to an X100P.)
This is the normal behavior. Only X100P will report the real status.
_
Tom Lynn wrote:
Do they appear to have failed as a result of Daylight Savings time?
DST for 7905/7912 are set inside the lddefault/gkdefault - or the
individual config file (ldMAC / gkMAC), but can't be set in advance like
7940/7960. DST is not the reason here...
Matt Putnam wrote:
anything useful any sugestions?
Are they requesting anything via TFTP? Do you have the full tftp files
ready?
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Jim Freeze wrote:
I suppose that is my alternative - remove the 4FXO card and add an
8FXO card. But I'm not seeing the prices you list. The Digium
TDM2402B is listed at $837.00. Am I missing something?
Digium is releasing a new 8 FXO/FXS card TDM800P, based on the same
expansion cards used for
Carlos Rojas wrote:
Anyone know a good carrier of voip for international calls?
Please use asterisk-biz list
http://lists.digium.com/mailman/listinfo/asterisk-biz
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Noc Phibee wrote:
after 2 mounth of search, i don't have see a billing solution
for my small business..
Not quite sure as I didn't research very much their product, but did you
check Aradial?
http://www.aradial.com/voip-billing-radius.html
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programming dept wrote:
What happens is that if we terminate calls to carriers who accept
only the g729 codec we get a 503 service unavailable.
are you sure that your carrier will accept g.729? Sometimes they don't
accept under iax2 and do accept under sip... check your debug for more
informa
On Sun, Oct 08, 2006 at 10:39:26PM -0600, Joseph wrote:
> I have bind-address = 127.0.0.1 in my.cnf
> the cdr was working find with asterisk 1.0.1 just after upgrade
> something is not connecting.
I don't know if asterisk will use the localhost or the "network" IP to
connect. Just try to comment
On Sun, Oct 08, 2006 at 10:04:51PM -0600, Joseph wrote:
> What am I missing?
Maybe your /etc/mysql/my.cnf ?
# Instead of skip-networking you can listen only on
# localhost which is more compatible and is not less secure.
# bind-address = 127.0.0.1
#skip-networking
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Technical Support wrote:
Can someone point me to call center reports available from Asterisk?
http://queuemetrics.loway.it/
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Noc Phibee wrote:
anyone know where i can solve this problems ? :
1) By doing a quick google search;
2) By reading previous posts regarding the same issue;
3) By disabling VAD (Voice activity detection) in your device.
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PLEASE DON'T CROSS POST!
Kannaiyan Natesan wrote:
I heard of a news, that there is a replacement codec available for
g729 and accept the g729 codec data for decoding. [...] If there is
any royalty need to pay, is that cheaper than the existing g729
cost?.
G729 is not royalty free.
http://lis
Michael Strelnikov wrote:
1. I want all incoming calls are redirected from SPA3000 to my
asterisk server. 2. Asterisk then should direct this call to my SIP
phones (including Sipura) 3. In case asterisk server is down I want
that call be directed straight to the handset connected to the Sipura
Cisco released last Aug 23 the latest SIP firmware for Cisco 7960G. Any
info?
SIP Flash Image for 7940/7960 IP Phone v8.4(0) - Non CallManager
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Ricardo Carvalho wrote:
I've been searching for sound files in Portuguese language to use in
Asterisk for example for voicemail, but I couldn't find anything...
Does anyone know where I could find them for download, if there is such
thing already?
Brazilian Portuguese only...
http://www.google
Andre Courchesne - Consultant wrote:
[EMAIL PROTECTED] tmp]# date
Mon Aug 14 16:44:15 EDT 2006
The Linux command line time is connect, but not Asterisk...
just guessing... not sure:
date -u
is showing what?
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Julian Lyndon-Smith wrote:
Is there any way of specifying a directory to load tftp files from
instead of from the root tftp directory when booting a cisco 7960 phone ?
SIPDefault.cnf:
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./7960/" ; Example: ./sip_phone/
/
Abdul wrote:
Could any one tell me how i can change CDR variable value from
extentions.conf file.
for the example i would like to change the src field value different
that caller phone on the first attempt of call?
exten => blabla,1,Set(CDR(fieldname)=new_value) (for asterisk >= 1.2)
http:/
Mr. Jones wrote:
I have 20 DIDs, some I want to send to a menu, most directly to an
extension.
sip debug is (really) your friend. It should give you the [context]
where your DID is being send to and the 404 not found error also.
A particular line to look for: "Looking for ...".
___
Vic wrote:
> I am in immediate need of configuring an Asterix to act as wake up call
> system.
Amazing:
http://www.google.com/search?q=asterisk+wake+up+call+site%3Avoip-info.org
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Ron Wellsted wrote:
I have been trying all the major distributors but they are all out of
stock with no dates for new stock to be delivered.
As you are in the UK, why not talking directly to Billion? Maybe they
can help: http://www.billion.uk.com/contact.htm
I'm also trying to find a new sup
voiplist wrote:
Is there a command to check the call duration of an active call in
the CLI?
show channels verbose
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Andrea Spadaccini wrote:
Is there any hope to change the caller-id on a BRI line?
I guess you can do it within the range assigned to you. If you have 2
numbers, you can choose between these two numbers. Not tested, as I have
only 1 number here (and still fighting with the "zaphfc: empty HDLC
Nhadie wrote:
Does anyone here have Japanese version of the asterisk sound files?
http://www.google.com/search?q=japanese+sound+files+site%3Avoip-info.org
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Mimmus wrote:
Could some goodwill man summarize this topic for me before I engage
myself in the rediscovery of warm water?
Read a topic posted a few days ago: "ISDN BRI NetJet"
You will find good advice there.
If you need to buy a Cologne chipset card, check here:
http://www.solwise.co.uk/isdn.
Grady Neely wrote:
Has anyone gotten Packet8 setup as a sip trunk for Asterisk?
I have it here. With a TDM400.
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I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1
Anyone was able to use this card with asterisk? I couldn't find much
information about it. Any help?
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samuel wrote:
I am of Argentina, and I do not speak very well English, I cannot
install asterisk in red hat 9.
Don't send HTML messages to the list.
Install [EMAIL PROTECTED] Please remember that [EMAIL PROTECTED] will erase all data on
your HD.
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Kim Culhan wrote:
Was running the Digium FreeBSD g.729 codec until recently when the latest
Asterisk bits were obtained via svn:
svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
MAYBE it is the same problem:
http://lists.digium.com/pipermail/asterisk-users/2006-April/147577.html
Steve Totaro wrote:
In what way is their email server configured "badly"?
Wrong DNS entries.
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Carlos Chavez wrote:
Now that Nufone is dead, what are other providers of 800 numbers that
work with Asterisk?
Nufone is NOT dead. It is working and I just added more funds into my
account.
You may also consider Asterlink. I'm a new client there, their support
is a little slow, sometimes irre
Andy Jefferson wrote:
Went to their site today. Site claims they are still in biz. What is
the story? What really happened to Nufone anyway?
I'm using them for almost 3 years now. They are having some problems
with OLD DIDs and toll free numbers, but newly assigned are working
fine. I ordered
Matt wrote:
Is there more to this story then we know?
No secrets, but at least some information may be found here:
http://www.nufone.net/press/
Latest update April 28.
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Jefferson Carvalho wrote:
I always used a compiled version for a x86 system
From [...]
Someone could help me on this?
Yes, the folks at Digium will be more than happy to help you.
Visit http://www.digium.com/en/products/voice/g729codec.php and get a
licensed codec.
___
Alexander Burke wrote:
Just in case anyone here hadn't noticed, Cisco is apparently making
7940/7960 SIP 8.2 firmware freely downloadable by anyone:
8.2 isn't broken? Any comments?
http://lists.digium.com/pipermail/asterisk-users/2006-March/143501.html
_
Shaun wrote:
Well looks like the phone is sending some data... I was unable to debug the
problem however..
> Looking for 9011905326471222 in default (domain 204.10.xxx.xxx)
Do you have a pattern in the default context that will match
9011905326471222 ?
___
Shaun wrote:
I'm having a problem with my Cisco 7960 phones with the SIP image. When i
try to dial a international number i keep getting a busy signal but i dont
see anything on the asterisk console (-vc) like i do when i dial
local or long distance numbers.
sip debug peer your-phone
Corne Labuschagne wrote:
How do I setup faxing in asterisk
http://tinyurl.com/qddpf
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Corey S. McFadden wrote:
PHP/MySQL based content manager for the Cisco 79XX series IP Phones
Any mailing list available for this project? I have some
questions/updates about this project...
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[EMAIL PROTECTED] wrote:
Well yeah, I had no intention of buying one, I was just wondering what
the hell it actually was that the seller was trying to hide.
Their supplier?
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Waldo Rubinstein wrote:
I'm looking for a reliable 2 FXO-port gateway to connect a Panasonic
PBX to Asterisk. Can anyone recommend a stable and reliable one?
Use 2x Sipura SPA-3000 - and you will also get 2x FXS...
Or use a Digium TDM02B (2x FXO).
__
Diego Mariano Velo wrote:
Hi, i have a cisco 7912G with SIP firmware, its connect to the asterisk
through nat. The only problems is in the voice mailasterisk not
detect the tones, therefore i cant access to my voice mail extension.
Check the DTMF settings...
http://www.voip-info.org/wiki/v
Hermann Wecke wrote:
I'm having a little problem to update the database after a call was
placed. I have several PSTN lines and I need to split the calls between
them.
[...]
Any idea?
Solution: write to the database BEFORE the dial command. Worked very
I'm having a little problem to update the database after a call was
placed. I have several PSTN lines and I need to split the calls between
them.
The approach I used didn't work:
[sipphone]
include => trunktest
; other rules here blah blah blah
[trunktest]
exten => _1800NXX,1,DBget(LAST
Innocent Evil wrote:
I am trying to download a list of international dialing codes.
Would anybody please post a link to get it
Google IS your friend. Did you try?
Google: international country code
Wikipedia: http://en.wikipedia.org/wiki/List_of_country_calling_codes
__
[EMAIL PROTECTED] wrote:
I need to reboot every day an asterisk box, but I would like to do that
only when asterisk is not doing anything.
I have no idea *why* do you need to reboot the machine every day.
What I do is a full asterisk restart - removing the modules and
reinstalling them. My box
Does anyone know if Clipcomm CG-410 [1] is able to handle caller-id
information from PSTN and send it to Asterisk? Any "trick" on asterisk
side to handle it? I tried several configurations but none worked. TIA
[1] http://tinyurl.com/c6k4f
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Edwin Lam wrote:
is there anyway to make the dial command return and execute
the next line in the dial plan after the channel hangs up?
Try "g":
exten => 1234,1,dial(SIP/1234,,g)
exten => 1234,2,
g: When the called party hangs up, exit to execute more commands in the
current context.
http:
I need to place a call using a "pin code". To access an external line,
the host PBX (a Ericsson MD-110) will require that I dial
*72*pincode#phone_number to complete any (trunk) call.
When I send the number, my Sipura 3000 will reject the call with
"Forbidden - wrong password on authentication
altus wrote:
We installed a box a long time ago and they bought g729a licenses
Now we want to upgrade and reinstall,whats going to happen with the
codec,if I give the box the same ip as always will it work?
The Digium g729 license is bonded to the MAC address of all the
interfaces you have. I
Kumara Jayaweera wrote:
I want to run VoIP in the same LAN (15 windows clients) which we use for
surfing the Internet.
Some magic words: QoS Asterisk HTB TC. Not easy to find good material
over the internet, but Google may give you some ideas - how to use them
is another problem, which you have t
I'm trying to use TFTP to update the firmware on a Cisco ATA188 but I'm
receiving this error:
May 1 06:51:50 mail2 in.tftpd[11499]: connect from 192.168.2.2
May 1 06:51:50 mail2 tftpd[11500]: tftpd: trying to get file:
ata01234567890a
May 1 06:51:50 mail2 tftpd[11500]: tftpd: serving file fro
William Suffill wrote:
According to the small print in the bottom graphic:
http://www.sipura.com/products/spa2100.htm
The SPA 2100 would give u 2 ports + 2 RJ45 as well as 2 G729
When I was placing an online order, I found this:
"support for two concurrent calls using the G.729 codec (in a firmwar
Chris Lee wrote:
Has anyone else upgraded to 7.4 and found that the date & time no
longer appears on the phone?
This problem was pointed at the SIPPhoneReleaseNotes7_4.pdf file.
What I noticed is that when the phone lost the internet connection the
date/time will no longer be present on the phone.
I found a thread [1] last month about the poor/crappy g729 quality on
Sipura units. Anyone noticed an improvement or the quality is still poor?
If the Sipura firmware/g729 offers no quality yet, who else is offering
a dual channel g729 ATA? I heard about Uniden, but I have no "reports"
about th
Brian Dingman wrote:
The FWD -> Vonage interconnect has been down for some time now. Vonage
claimed there was a secuity issue and pulled the plug. No word when/if
it will ever be working again.
So I'm guessing that FWD <-> Packet8 falls into the same problem? Not
working here for a couple of weeks
Sys Admin wrote:
couldnt agree with u more !!
And, please, add another one to the list: PLEASE TRIM THE &^*&[EMAIL PROTECTED]
MESSAGE. TIA.
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Vicky Shrestha wrote:
I have tried a lot of things to make broadvoice work with asterisk , but I
failed each time.
I had some problems here, mainly because I was trying to use g729 and
broadvoice will only accept g711. Other than that, configuration itself
took about 10~15 minutes with some goog
Tom wrote:
What times are others seeing for the load when you reboot a phone?
About the same here, but I don't care as I never reboot my phone (about
once every month or two).
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C F wrote:
how are you telling the cisco what the password is? TFTP?
TFTP (SIPmacaddress.cnf)
you will not see anything on * CLI unelss you do sip debug
And after "sip debug" I saw (among other lines):
[...]
Retransmitting #5 (NAT):
SIP/2.0 407 Proxy Authentication Required
[...]
SIP/2.0 401 Unauth
Rich Adamson wrote:
Looks like a couple of problems here. I don't believe the Cisco phone
handles md5, so remove that line.
As I told before, tried 3 different approaches:
1) password; md5;
2) password, no md5;
3) no password, no md5.
Only the third one worked. Trying to give SOME security, I added
After fighting with a "Unable to create/find channel" [1] [2], I gave up
on my previous installation and rebuild my asterisk from CVS-Head. I
guess the Debian package available today is broken somewhere (after a
previous broken release made with an old libpri package), but now I'm
having anothe
Christian faucher wrote:
I read that, using a modem,I can use a standard phone line, and
"convert" that as input for Asterisk PBX, right?
Not that simple, not every "modem", but yes.
Also, where can I get VOIP phones?
eBay
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Guy Decarpentrie wrote:
Try to configure your Cisco type=friend in your sip.conf
It is already type=friend
[1234]
type=friend
username=1234
auth=md5
secret=supersecret
deny=0.0.0.0/0.0.0.0
permit=my_ip/255.255.255.255
canreinvite=no
reinvite=no
host=dynamic
dtmfmode=rfc2833
qualify=1800
mailbox=123
I'm trying to place a call from my Cisco 7960 and I'm receiving this error:
Mar 1 06:19:44 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to
create/find channel
Mar 1 06:19:58 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to
create/find channel
I can't place calls, but I can recei
Paul A Brown wrote:
Anyone had a Cisco 7970 working with Asterisk?
As 7970 uses SCCP, you can do it with asterisk. I did it with 7960.
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Matthew Boehm wrote:
Is there a way for asterisk to notify you of this? Send an email? Send a
page? Call you?
Nagios (I believe now is called NetSaint) can do this and much more.
But you must have "the power" to configure it... after that, Nagios can
send you an email, a pager, even call you and t
Max wrote:
Pessoal estou querendo montar um servidor SIP para fazer testes [...]
wrong list. For Portuguese mailing list please subscribe to
http://groups.yahoo.com/group/asteriskbr/
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http:/
Olaf Klein wrote:
Why not just kill yourself, fucking wannabe spammer? DIE DIE DIE
This is *REALLY* offtopic, but Isamar is the founder of Brazilian
AntiSPAM - http://antispam.org.br/ and later http://spambr.org/
Does it matter here? I don't think so, but calling he (or even me) a
"spammer" is re
dean collins wrote:
Guess I'll just have to stick with running connections to the ATA's via
X100P
That's what I do here. As I have a very old plan (US$ 6/month), I only
use pkt8 to place international calls - and keep it as an emergency
backup for the US.
Roger Schreiter wrote:
But when dialing a number, I get:
Feb 2 09:44:45 NOTICE[20380]: chan_sip.c:7296 handle_request: Unable to
create/find channel
After I installed my Digium g729 license, I'm trying to place a call
from my Cisco 7960 and I'm receiving the same error:
Feb 19 09:47:06 NOTICE[2
Matthew Boehm wrote:
[...] In the meantime, get a Sipura 2100, supports 2 729 calls and
has both WAN/LAN ports.
I was told that the Uniden DTA200 also supports 2 g729 calls. I'm buying
one to test. Street price around US$ 90.
Another one with dual g729 channels is MTA V102. Street price US$ 100.
Tzafrir Cohen wrote:
BTW: did I mention that we have binary packages for standard Debian
Sarge kernels in our apt source?
zaptel is the only package that never worked for me from apt-get. I need
to download, compile and install the kernel (specially because the
original debian install is pre 2.4.
I'm trying to find some "live" examples on how to use the h, H and g
parameters on the dial command
(http://www.voip-info.org/wiki-Asterisk+cmd+dial)
Any ideas? I was testing with the code below but after pressing *
nothing happens (only after a long pause the "goodye" file was played)
[testse
Dave Green wrote:
Following a top posted thread is a pain.
not trimming the useless part of a reply is another pain...
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Brian West wrote:
exten => _8001133[12345789]XX.,1,Dial(SIP/france-gateway,60,tr)
or
exten => _8001133[1-57-9]XX.,1,Dial(SIP/france-gateway,60,tr)
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Is there any winblows softphone available offering g729 *and* IAX?
I couldn't find any http://www.voip-info.org/wiki-VOIP+Phones
The best choice should be dIAX, but it is only GSM.
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Eric Wieling aka ManxPower wrote:
What company are you using for your service?
Intelsat. But I'm not using it "point-to-point" as I'm not the primary
contractor of this channel - I'm buying internet access.
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Michael Graves wrote:
[...] Although there have
been a few (very few) times when I've notcied a brief pause after
dialing and found that it had in fact dialed out on the last possible
option.
[...]
The problem of your approach is that if you are out of credit with the
first provider, your call wil
Federico Gonzalez wrote:
I have an Asterisk with one local Cisco ATA and one remote Cisco ATA
connected to the Asterisk, the remore connection is a satellite link
with an 900ms delay.
This is the same delay I have here. Never less than 900, sometimes over
1500 ms.
Check
http://lists.digium.com/pi
I noticed that I'm no longer able to receive calls from PSTN to my
SipGate DID number.
I changed the sip.conf and extension.conf as per
http://www.voip-info.org/tiki-index.php?page=Sipgate but the problem
remains...
However, I can receive calls from another sipgate user. The problem is
only a
Joseph wrote:
After somebody records a message asterisk notifies me and encloses the
WAV file. Though I'm not sure if this is a WAV format. I can not play
it.
How to play received message?
Did you try to use Windows Merdia Player?
In other hand, if you are receiving a .GSM file, you can use the j2
nkb wrote:
So, do I still need to have an Asterisk server connected to my IAXy even
after I've made provision for it?
You can only connect IAXy to an asterisk server. Yours or from a VoIP
provider.
Like, can I just carry this IAXy
around(after provision) and just plug into any broadband connect
nkb wrote:
I was wondering if I could use IXAy to forward my call via the internet
to my destination, something of similar function to SIPURA 3000?
The IAXy is similar to the Sipura 1000 or 2000, or the Cisco ATA 18x...
You can use it to connect to a VoIP server with the IAX2 protocol
(instead o
Damon Estep wrote:
[...] Contains a link you need for firmware.
Correct URL is http://www.voip-info.org/wiki-Asterisk+phone+cisco+ATA18x
http://www.voip-info.org/wiki-Asterisk+phone+cisco+ATA18x>
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