Greeitngs !,
I am haivng asterisk 1.0.x verison and going to upgrade it to version 1.2.4.
with zaptel 1.2.8. i have PATA drives. asterisk 1.0.x verison didnt support
SATA (IRQ problems). so..this new relaeasr 1.2.x can support SATA drives on
dual core processor ??
iam using TDM04B card.
hope u
Sounds like a nice setup you have in mind.
All I can tell is that you might have trouble with clocking on your PRI's if you use multiple cards in one system.
I've read about it somewhere, but can't find the source. Have a look at the wiki.
Syncing clocks on one card happens on the card level. But
I use the latest version of zaptel, asterisk___
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Greeitngs !,
I am haivng asterisk 1.0.x verison and going to upgrade it to version
1.2.4.
with zaptel 1.2.8. i have PATA drives. asterisk 1.0.x verison didnt
support
SATA (IRQ problems). so..this new relaeasr 1.2.x can support SATA drives
on
dual core processor ??
iam using TDM04B card.
For multiple TE4XXP cards you strap them with http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?product_code=ACCTIM01
to prevent corruption due to timing slips on the second, third or fourth TE4XXP card.
The link above shows four cards linked.We tested up to three quad PRI cards in
Check sip.conf parameters:
rtptimeout
rtpholdtimeout
David Gagnon wrote:
I would recommend you to call Unlimitel as they have a very good support. Or
just send a copy of your post to : [EMAIL PROTECTED]
David
_
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
RR wrote:
I am currently running this with UnixODBC - FreeTDS - MSSQL Server
2K ( please don't hate me for using an 'evil empire' product amongst
the pure sanctity of open source :D). But the results are, well...So
far so good. But I can't say
Hi all,
I'm trying to find some info on how to create my own dialplan
applications. Like f.i. Echo (ast_echo.c in apps). The API used in there
is what I would like docs on.
TIA
/Rob
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-Ursprüngliche Nachricht-
Von: Tharanga [mailto:[EMAIL PROTECTED]
Gesendet: Freitag, 8. September 2006 07:35
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Asterisk 1.2 and SATA drives
Greeitngs !,
I am haivng asterisk 1.0.x verison and going to upgrade it to
Hello,
I recorded some files (gsm format) but i can not
hear these files without g729
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/86-08218198,
) in new stack
-- Executing [EMAIL PROTECTED]:2] Dial(SIP/86-08218198,
Sip/84|30|tTj)
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tharanga wrote:
Greeitngs !,
I am haivng asterisk 1.0.x verison and going to upgrade it to version 1.2.4.
with zaptel 1.2.8. i have PATA drives. asterisk 1.0.x verison didnt support
SATA (IRQ problems). so..this new relaeasr 1.2.x can support
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rob Lith wrote:
For multiple TE4XXP cards you strap them with
http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?product_code=ACCTIM01
http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?product_code=ACCTIM01
to prevent
--- [EMAIL PROTECTED] a écrit :
Hello,
I recorded some files (gsm format) but i can not
hear these files without g729
-- Executing [EMAIL PROTECTED]:1]
Answer(SIP/86-08218198,
) in new stack
-- Executing [EMAIL
Dan Austin wrote:
As far as the above is concerned I have the following:
I am using Asterisk 1.2.10, patched with this patch for 1.2.10.
I have 2 * boxes. They call each other over SIP, and I have in
sip.conf on both boxes
autoframing=yes
disallow=all
allow=g729:80
When A calls B,
Hi Dave , long time no ear :-)still progressing on * , I have now my own distribution on a compact flash including bristuff ...i m fighting with this nice 9133i, lastest firmware 1.4 , blf is turned on and working ( light flashing when calls comes in ) but I can not pick it up,
ps I followed the
Hello,
I look at the new sip firmware however i don't
undanstand the presence features.
I don't use LCS but SER as presence server this one is
able to provide a ressource list server and xcap
server
for sip buddies lists .
Does polycom phones can suscribe to a
sip:[EMAIL PROTECTED] for example
Dan Austin wrote:
2006-08-31 22:11:22 WARNING[1278]: frame.c:1072
ast_codec_pref_getsize: Framing not set for codec alaw, using
default 20
As far as the above is concerned I have the following:
I am using Asterisk 1.2.10, patched with this patch for 1.2.10.
I have 2 * boxes. They call
Hello,
We currenty have an asterisk cluster running, with a quad PRI and a quad BRI.
This all works pretty well.
What i was wondering:
If i do a
show sip peers
I see all the ip addresses of the phones that registered, also, when restarting
the server.
Is there any way of copying this
Hello List
I am having a strange problem, that seems to have appeared from nothing.
Im running Asterisk 1.2.9.1, and we use the app_directed_pickup application.
But recently we get this result:
AGI Script Executing Application: (Pickup) Options:
Hello,
I set fr however app_voicemail play english digits
instead of french digits for voicemailmain !
Harry
___
Découvrez un nouveau moyen de poser toutes vos questions quelque soit
Hello,
Cups unfortunately don't support multi-page tiff printing.
http://www.cups.org/str.php?L1117
I have tried tiff2ps and tiff2pdf but both just embed original tiff
file and give the first page only.
Is there any solution for printing multi-page tiff easily? More likely
an alternative lp
Hello Harry,
I set fr however app_voicemail play english digits
instead of french digits for voicemailmain !
Did you check if the digits in French exist on the server?
If it cannot find a specific file in french, it will play the default (english)
ones.
Regards,
Roelof
Harry
Hi Harry,
digits sounds must be placed inside /var/lib/asterisk/sounds/digits/fr.
Check it.
Giorgio Incantalupo
[EMAIL PROTECTED] wrote:
Hello,
I set fr however app_voicemail play english digits
instead of french digits for voicemailmain !
Harry
On Fri, Sep 08, 2006 at 11:00:56AM +0200, [EMAIL PROTECTED] wrote:
Hello,
I recorded some files (gsm format) but i can not
hear these files without g729
Any chance that you try to play them to a channel that uses a g729
codec?
I believe that this requires a separate g729 codec instance.
Hello (o;
Ist there a way to remove the trailing @domain from
the displayed caller id on the Cisco 7970G?
No problem dialing a number from the missed call
directory with the domain attached...just looks
weird (o;
cheers
rick
___
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Artifex Maximus wrote:
Hello,
I have tried tiff2ps and tiff2pdf but both just embed original tiff
file and give the first page only.
I'm currently using tiff2pdf and using lpr to spool it to a CUPS
controlled, network attached printer (Minolta Dialta 850). Seems to be
printing multipage
On Fri, Sep 08, 2006 at 11:59:40AM +0200, [EMAIL PROTECTED] wrote:
Hello,
I set fr however app_voicemail play english digits
instead of french digits for voicemailmain !
That's an interesting fact that you stated. Did you want to ask
anything? Is there anything we can help you with?
It is
Fax2ps is what we use, works fine.
Yum tells me it comes from libtiff
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Artifex
Maximus
Sent: 08 September 2006 11:05
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Trouble with rxfax
I used codec_g729.so in stable realease so i set g729
with th highest priority .
With Asterisk SVN-trunk-r41990 i don't allow g729
Harry
--- Tzafrir Cohen [EMAIL PROTECTED] a écrit :
On Fri, Sep 08, 2006 at 11:00:56AM +0200,
[EMAIL PROTECTED] wrote:
Hello,
I recorded some files (gsm
Hi
today, i have a big problems with my asterisk ...
when i want call i have this error :
Sep 8 12:38:07 WARNING[28369]: chan_sip.c:1226 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 102 (Critical
Request)
Sep 8 12:38:07 WARNING[28369]: chan_sip.c:1243
Does the Linksys know it should be using port 5070? It would seem to me
that port forwarding would be required as the phones are behind a NAT'd
firewall. How would asterisk know how to get there since it's not on
the same subnet (outside the firewall).
If the asterisk box has physical access to
In extensions.conf I want to implement a dial plan that distinguishes
the users that wish to dial a PSTN number by their own domain, so that
[EMAIL PROTECTED] goes out to PSTN by a different DID than [EMAIL PROTECTED]
I tried the following line, but that doesn't distinguish between
domains,
Hi Mike,
I have a IP 500 and do similar mute for
calls etc. No probs here using Trixbox.
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Gagnon
Sent: Thursday, 7 September 2006
9:28 PM
To: 'Asterisk
Users Mailing List -
My q too!!
I mean, simple extension numbers, 3000, 3001, etc, can be present in
multiple pbxes(in any hosted pbx service).
I guess 'context' is a way, but it seems, in *, the dialplan(and hence
the context) is decided by the callee digits. Any work arnd over this one?
cheerz
Ben.
Ricardo
HiTry inkeys instead of inkey and you should be fine.Regards Andrew
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Thanks for the words of hope. I actually think I
found something, I set LineSeize=`30` in the Polycom config file (I still dont
know what that means...) but it seemed to work once (out of one
try)
"Once" doesn`t mean "always", but it is definitely better
than "never"
BTW, Someone
Hello all!,
Ive install all package of Vicidial and
astguiclient as I read on a scratch install notes in a CentOs 4 with trixbox.
But when I use some AGISadded in a dialplanof the install
documentation i get some sintax error on this scripts
likeagi-VDAcloser_inboundCIDlookup.agi.
I have a problem with a really weird piece of hardware, the KG1000 voip
gateway. Information about this device is nonexistant, and I have been
able to get it to work through trial-and-error except for one issue: DTMF.
Basically, a call is initiated okay, but Asterisk ignores any DTMF signals
sent,
Hello,
Many thanks for your help Doug and Steve!
I tried tiff2pdf again and this time prints all pages. Hm. I have
nothing to say. :-)
I tried fax2ps as well and looks faster than tiff2pdf so I am using this.
bye,
Zsolt
On 9/8/06, Steve Hanselman [EMAIL PROTECTED] wrote:
Fax2ps is what we
First call is answered by LINE1, but if this line is still busy and a second call comes in, it doesn't go to LINE2, instead called listens asterisk message, all lines are busy, please leave your message after the tone. I tried resetting phone to factory default setting too, but still it does the
Hello,
Yet another
questionI've been attempting to set CDR variables during my
dialplan.
ex:
exten =
test,1,Set(CDR(ACCOUNTCODE)=${EXTEN}) ; works fine!
exten =
test,2,Set(CDR(src)=${EXTEN}) ; gives me the following
error:
Sep 8
09:23:53 ERROR[10269]: cdr.c:289 ast_cdr_setvar:
whatever the did is needs to be put in the extensions.conf told to dial your cellphone.
Example:
exten = _011123445566,1,Dial,SIP/[EMAIL PROTECTED]
assuming that your using a SIP carrier, replace 1234567890 with your cellphone 1.2.3.4 with the carrier's IP or carriers context name in
The configuration tool downloaded from grandstream website creates an output file, but how do I load configuration file to 20 phones? Is there some separate software for that?-- Zeeshan A Zakaria
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You need to enable call waiting on the phone's config.- DanielOn Sep 8, 2006, at 9:35 AM, Zeeshan Zakaria wrote:First call is answered by LINE1, but if this line is still busy and a second call comes in, it doesn't go to LINE2, instead called listens asterisk message, "all lines are busy, please
I have an asterisk box configured to perform media translation (TDM -
SIP). With this configuration, calls are essentially only passing
through the asterisk box. Thus, I would think that a disconnect request
should be received from one end of the call (SIP or TDM) into the
asterisk box as an
Check out http://www.grandstream.com/GAPSLITE/
It's a bit of a pain to setup but it works. You will need to create a script that changes the info on the template for each ATA that you want to configure.
bp
On 9/8/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
The configuration tool downloaded from
Im sending this to the asterisk
list consciously knowing it will piss some people off but this is the most important business issue of the day
for anyone doing business in or with a US company.
Oh and before anyone on this list thinks
of calling this OT how about Mark Spencer being
Whenever I issue a reload command I never seem to get any console feedback and the command never finishes. Is there a lock file or
something I should check for? The system is still running and working just not reloading. Is there any way to find out where it's
hanging?
Connected to Asterisk
On Fri, 2006-09-08 at 10:31 -0400, Mailing List wrote:
Whenever I issue a reload command I never seem to get any console feedback
and the command never finishes. Is there a lock file or
something I should check for? The system is still running and working just
not reloading. Is there any
In article [EMAIL PROTECTED],
Jamin W. Collins [EMAIL PROTECTED] wrote:
I have an asterisk box configured to perform media translation (TDM -
SIP). With this configuration, calls are essentially only passing
through the asterisk box. Thus, I would think that a disconnect request
should be
Hi All, I try to transcode Speex to G711 through Asterisk, however, the voice quality is crappy. Does anyone has a clue to fix this? The current version i running now is 1.2.4, and the Speex codec was sent from eyeBeam.The following warning below:Sep 7 13:48:55 WARNING[5297]: codec_speex.c:278
It has nothing to do with Asterisk as far as I know. The kernel needs to
support it I believe. DO NOT even attempt SATA RAID without a hardware RAID
card that is supported in the kernel or your asking for headaches. For
non-RAID, many BIOS's emulate IDE so if that is the case you should have no
Hi!
I try to make my Asterisk contact a SIP user thanks to a redirect
server. In fact Asterisk try to reach a SIP address that is redirected
to the good one.
The error response is:
*CLI -- Executing Dial(OSS/dsp, sip/[EMAIL PROTECTED]|30|
H|g) in new stack
-- Called [EMAIL PROTECTED]
I highly recommend the 3ware line of SATA RAID cards for doing SATA RAID. I've installed them in upwards of 400-500 servers, and they're rock solid cards and affordable.Disclaimer: I do not work for 3ware or AMCC, but am a very satisfied customer.
-brandonOn 9/8/06, shadowym [EMAIL PROTECTED]
It's all in the Asterisk database, which is a Berkeley DB format as far as I
know. If it's just for migration, you could probably just move the database.
If you want to do this while the system is up, maybe using an external
database for this information would work better.
-Tim
On September
Tony Mountifield wrote:
It looks like the PRI connection is going down first, and when that channel
exits, it causes the SIP channel to be hung up. So concentrate on the PRI.
Yep, that's what I've seen so far. Been trying to concentrate on the
PRI, but not seeing any indication of what is
Did this email go through the first time
or is the email down?
Ive been reading about this online
everywhere this morning and I havent seen it come back on the asterisk server.
Cheers,
Dean
From:
Dean Collins
Sent: Friday, 8 September 2006
10:22 AM
To: 'Asterisk
8 sep 2006 kl. 17.18 skrev Michel Zenone:
Hi!
I try to make my Asterisk contact a SIP user thanks to a redirect
server. In fact Asterisk try to reach a SIP address that is redirected
to the good one.
The error response is:
*CLI -- Executing Dial(OSS/dsp, sip/[EMAIL PROTECTED]|
30|
The guy was arrested because he was conciously allowing US citizens to gamble online, something illegal in the US at this point. He ran a business who's sole profit source came from this illegal activity. Spencer is an entirely different situation. He's not running a business in which he provides
Exactly so why arent they trying to
arrest the 50 million people in the USA who have gambled online?
Mark (as far as I know) isnt actively checking
with asterisk users for what country they are in so therefore in the reciprocal
eyes of the indian government he is similarly breaking the
It sounds like a good idea, I tried it and get this error
Sep 8 09:52:17 WARNING[27193]: chan_sip.c:1968 create_addr: No such host: gafachi-o
Sep 8 09:52:17 NOTICE[27193]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
In
Is autoframing set to yes in the [general] section? A current
limitation in the code is that a global autoframing will
override a user/peer setting.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of yusuf
Sent: Friday, September 08, 2006 2:26 AM
To:
IS running a gambling site illegal in the
UK?
If so, perhaps they had a warrant out for him, and
we (US) are just going to extradite him.
-- -- Steven
http://www.glimasoutheast.org
"Dean Collins" [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]...
Exactly so why arent
Why don't you keep political diatribe to your blog? This is OT, and
quite frankly it displays that you have less than perfect grasp on reality.
Mark Spencer makes a software product that is perfectly legal to use
anywhere in the world, even in India (as long as it stays within a
building and
No its totally legal. Sportingbet
PLC is a legal and registered company in the USA.
He was arrested in NY for a charge that is
going to be extradited and charged in Louisiana.
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
anyone know this error ??
Noc Phibee a écrit :
Hi
today, i have a big problems with my asterisk ...
when i want call i have this error :
Sep 8 12:38:07 WARNING[28369]: chan_sip.c:1226 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 102 (Critical
Yes I am aware he is the second executive to be arrested..the first
is still yet to be charged and is still awaiting trial and has fallen
off the face of the general media which is why I'm 'motivated' to draw
attention and outrage to this second case.
Yes you are right it does belong off
After using PauseQueueMember in my dialplan.
I used 'show agents' cli to show the agent status. It is still show that agent
available. Here is the output from asterisk console:
5156597 (Agent1 ) available at '[EMAIL PROTECTED]'
(musiconhold is 'default')5156598 (Agent2 ) not logged in
Do you have gafachi-o in your sip.conf?
Since it's not a valid host name, you need to have an entry in sip.conf to
tell asterisk how to make a call to gafachi-o.
That's why it is telling you No such host.
On September 8, 2006 12:57, [EMAIL PROTECTED] wrote:
It sounds like a good idea, I
gc wrote:
After using PauseQueueMember in my dialplan. I used 'show agents' cli
to show the agent status. It is still show that agent available. Here is
the output from asterisk console:
5156597 (Agent1 ) available at '[EMAIL PROTECTED]'
mailto:'[EMAIL PROTECTED]' (musiconhold is
The biggest problem with your argument is that VoIP is not illegal anywhere. Voice over _internet_ is, but voice over the Internet protocol is not. Anyone in China is free to setup Asterisk to use within their offices. There's nothing illegal about it. It's using VoIP, but it's not transmitting it
So... does anybody know how can I do this?
Maybe using a way to distinguish users not by their username, but by
other fields of SIP INVITE messages?
Regards,
Ricardo.
Ricardo Carvalho wrote:
In extensions.conf I want to implement a dial plan that distinguishes
the users that wish to
Friends,
Two times I've taken Asterisk to the SIPit interoperability tests,
both times has led to
much improved functionality and a lot of new ideas. It is important
to meet other
software developers struggling with the SIP standards, wanting to
make sure that
equipment and servers work
On 9/8/06, gc [EMAIL PROTECTED] wrote:
After using PauseQueueMember in my dialplan. I used 'show agents' cli to
show the agent status. It is still show that agent available. Here is the
output from asterisk console:
5156597 (Agent1 ) available at '[EMAIL PROTECTED]' (musiconhold is
The calling extension is 5156598.
After I dial into 881112 from this phone. It no longer accept call from
queue but the 'show agents' still show it is available.
- Original Message -
From: Julian Lyndon-Smith [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi all,
I have a box with an ISDN HFC card (1 BRI) connected to an Italian
ISDN, the card is using zaphfc driver and it's receiving and
originating calls quite regularly.
There are some numbers (mostly toll free numbers) that I cannot
connect to, here is what I get from the CLI:
--
Tim, this is the way I have Gafachi set up in sip.conf and works well with channels that have anATA attached to it but not the virtual one. I have changed the host in extensions.conf to the .sip.gafachi.com. But I have calls on the server and cannot restart it yet. I'll keep you
I've been running
into an issue with my Polycom 501 and Asterisk.
I realized, after
much mucking around, that when I dial a number (and press the send key) that is
invalid , but could still match an Asterisk pattern (example: I dial 567, which
is not a valid extension, but my diaplan
hithanks for reply.I'm using vicidial to make calls at 2.0 dial level it is able to make calls but when I see the asterisk -r most of the time it shows Outgoing Spool Failed. Which Spool File ?thanks
arunOn 9/8/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-Hash:
This looks like a networking issue - asterisk isn't receiving any
replies to signaling packets and assumes that the UA is no longer
reachable.
CP
On 8-Sep-06, at 10:33 AM, Noc Phibee wrote:
anyone know this error ??
Noc Phibee a écrit :
Hi
today, i have a big problems with my
Hi,
I'm using Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s.
I've got 3 ISDN phones attached.
When I want to dial out I can do it in 2 ways..
1) Type in number with handle still on.. Lift handle and we dial the
number
2) Lift handle and then press the number
Both methods should work, but only
Check your Dial() string to make sure that you haven't mistyped and put
gafachi-o instead of gafachi-out. Specifiying the full host name will also
work.
As a hint, you can refresh these changes with out restarting your server (and
therefore without disrupting any calls in progress)
With SIP, asterisk processes the digits it receives in the invite from the
Polycom.
There is no communication of dialplan information in SIP. The polycom should
send the digits as soon as you press dial. You can program the polycom with
a dialplan that will tell it when to send the digits,
Isn't there a way to specify a context based on the incoming domain in
sip.conf?
On September 8, 2006 14:03, Ricardo Carvalho wrote:
So... does anybody know how can I do this?
Maybe using a way to distinguish users not by their username, but by
other fields of SIP INVITE messages?
Tony Mountifield wrote:
Try enabling intense PRI debugging pri intense debug span N. You may want
to direct the PRI debugging to a file with pri set debug file filename.
It's not clear from the log you posted whether q931_hangup() was called
because of a Q.931 message Asterisk received, or
Could you send us some CLI output?
Look for something like this
Invalid extension s in context whatever your dial context is
It could be that lifting the handset without dialing is opening a channel to
the s extension, since there are no digits being dialed. There is a
workaround for this,
Thanks Tim.
I've been trying to find out what's happening. Basically, somehow, it seems
that my Polycom 501 knows what extensions are valid and which aren't in my
dialplan. Obviously, the 501 doesn't really know that, but Asterisk seems
to return it this info (sort of :valid, invalid or could
Mike wrote:
I've been running into an issue with my Polycom 501 and Asterisk.
I realized, after much mucking around, that when I dial a number (and press
the send key) that is invalid , but could still match an Asterisk pattern
(example: I dial 567, which is not a valid extension, but my
This isn't working for me either. I was about to ask this same question, but discovered this recent thread.
I have the following set up in my extensions.conf file, as per Granstream instructions:
[macro-page-grandstream]
exten = s,1,ChanIsAvail(${ARG1}|js); j is for jump, s is for ANY call
Now that is really odd.
Try sip debug peer (peername of the polycom)
This will let you see the sip packets go by when you do this, so you can see
the responses it is, or isn't getting.
I'll have to look up the SIP response codes, but I do know that there is one
for not found which should
Lol, ok - last post on the topic...I promise.
I'm going to let Ze Frank have the last word about the state of
government policy under the current administration and the role it plays
in the international stewardship here in the USA
Even in India, you can use VOIP for overseas calls
coming from your own company.
You just can't sell services that allow people to
call a PSTN number and then have their call sent over VOIP to another
location.
-- -- Steven
http://www.glimasoutheast.org
"Alex Robar" [EMAIL PROTECTED]
Steve,Forgive my ignorance, but why does India institute that policy?-brandonOn 9/8/06, Steven
[EMAIL PROTECTED] wrote:
Even in India, you can use VOIP for overseas calls
coming from your own company.
You just can't sell services that allow people to
call a PSTN number and then have their
That's exactly what happens:
When I pick up the handle, this is what I get:
-- Extension 's' in context 'from-inside' from '11' does not
exist. Rejecting call on channel 0/2, span 2
Do you know what to do in the dialplan?
Best regards,
Henrik Woffinden
Tim St. Pierre wrote:
Could you
Brandon Galbraith wrote:
Steve,
Forgive my ignorance, but why does India institute that policy?
Why does France blow up bombs in the south pacific? Each country can do
as it pleases - unfortunately - but that is also good for us VoIP
carriers because it creates and protects high retail
Protectionism; its not that
uncommon. Any number of countries around the world still have similar laws.
(even Australia until about 1998 I think).
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brandon Galbraith
Sent: Friday, 8 September
Mike wrote:
Let's just take 1) and 2). Why is Asterisk not going into the i extension
like it should?
Because the i extension is for IVRs and things like that.
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asterisk-users mailing list
Hi,I'm trying to connect my cell phone (motorola V3) to asterisk, using this guide: http://www.thetechguide.com/howto/asterisk/chanbluetooth.html
Everything has worked ok, but when I actually want to start asterisk, my phone doesn't connect all the way. All I'm getting in the asterisk CLI is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Mike wrote:
Thanks Tim.
I've been trying to find out what's happening. Basically, somehow, it seems
that my Polycom 501 knows what extensions are valid and which aren't in my
dialplan. Obviously, the 501 doesn't really know that, but Asterisk
Remove immediate=yes from /etc/asterisk/zapata.conf
Henrik Woffinden wrote:
That's exactly what happens:
When I pick up the handle, this is what I get:
-- Extension 's' in context 'from-inside' from '11' does not
exist. Rejecting call on channel 0/2, span 2
Do you know what to do in the
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