HI
Here is a question about the fxs of the zaptel card which is set a
number to use in the inter as common analog phone. When I also use
${CALLERID(num)}to get it's number, it also could not be done. At this time
,the fxs phone does not get any relation with the outbound which is like
PSTN
On 08/21/08 21:10, Joseph wrote:
I'm trying to configure Linksys 3102 for a short splash ring when someone
leaves a message.
in my sip.conf I have
mailbox=number
I have can see a visual indicator (light blinking on the phone) but there is
no short splash ring)
Linksys setting:
Regional - tab
On Fri, 22 Aug 2008, larry wrote:
HI
Here is a question about the fxs of the zaptel card which is set a
number to use in the inter as common analog phone. When I also use
${CALLERID(num)}to get it's number, it also could not be done. At this
time ,the fxs phone does not get any
Hi,
Is it possible to assign a plus sign on the callerid(num) ?
currently this is what i do CALLERID(num)=+6523450017
but telco is denying calls, coz they said they are seeing bs523450017
instead of +6523450017.
i tried putting it inside double quotes CALLERID(num)=+6523450017
telco says the
Just change your dial command and add the plus sign there.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 22, 2008, at 1:28 AM, ronald wrote:
Hi,
Is it possible to assign a plus sign on the
Hi
I am a premium voice service provider giving some services on IVR to a Telco X
. As my premises is some 10 kms away from that telco , i have taken a PRI
connection (30 DID with 1 hunting/pilot number) from telco Y When a customer
of Telco X dials my short code @Rs.6/- per minute his call
Hi Sir,
I actually have a plus sign on my dial plan
exten = _+.,1,Dial (
that is ok, dialed number (telco refers to it as B-number) is correct.
the prob is the originating number(they call this A-Number), i want to
set it to +65 so that it shows it is an international call.
so on my
Another example of the North American frequency allocations being just a
little bit different from everywhere else in the world... So does that
mean you've stopped using your S685 IP, Michael? ;)
Siemens USA does offer a few of the Gigaset DECT models over here (e.g.
the E450, S450 and S455
On Fri, Aug 22, 2008 at 03:12:41PM +1000, Col Ferguson wrote:
Hello all,
I have a system at a motel that is mostly analog phones with 2 32 port
astribanks.
What exactly is the trunk? FXO ports in the astribank?
I am having problems getting a modem data call to connect.
There are many
Dear Darren;
You might be right because one day it happened with me and the situation was
same like this as following:
The status that the ping result is very good for all partied (Asterisk machine,
IP Phones on the Internet), and no problem in the processor utilization or RAM
or hard disk
Hello Tzafrir,
Yes the trunk is an FXO port in the astribank
One astribank is 32 FXS ports, and one is 24 FXS and 8 FXO ports.
Just in case it makes a difference, the testing I am doing is with the modem
plugged in to the same astribank as the FXO ports.
Zap/69 is an FXO port and Zap/67 is an
Actually both calls have to be originated to the outside world. Thats why im
using @TRUNK-OUT, when the first call is answered only then the call goes to
a context. Thats where the problem is, the first call does not originate so
i cant throw it to any context.
On Thu, Aug 21, 2008 at 8:47 PM,
Check on dial plan rules, remember if you need dial to +number your
rule must be +|number this submit the number on your dialout plan
without +.
Regards,
Luis Morales
On Fri, Aug 22, 2008 at 3:40 AM, ronald [EMAIL PROTECTED] wrote:
Hi Sir,
I actually have a plus sign on my dial plan
exten
On 21 Aug 2008, at 14:40, Philipp Kempgen wrote:
Andy Dixon schrieb:
I am trying to alter the outbound callerID for extensions within a
context I have created.
I wrote the following:
exten = _9.,2,ExecIf($[$[${REALCALLERIDNUM} = 360] | $[$
{REALCALLERIDNUM} =
21 aug 2008 kl. 16.47 skrev [EMAIL PROTECTED] [EMAIL PROTECTED]:
Yesterday I blogged a post about some ideas that I think will help
Asterisk appliances further penetrate SMB/SOHO sites in ways that are
not presently being addressed.
I would prefer if you mailed the content too. After all
Andy Dixon wrote:
On 21 Aug 2008, at 14:40, Philipp Kempgen wrote:
This *should* change the callerID for (for example) 700 and 701 to be
581557, and any extensions not listed above, it should leave them
alone.
Andy,
If you're not bound and determined to do it this way, you can
Philippe Sultan [EMAIL PROTECTED] writes:
Well, if someone steals the md5secret (HA1) for a given username and
realm, he can use it to authenticate to the SIP proxy or B2BUA that
serves the target user.
This is unavoidable with password-based systems.
Either you transfer the password
Hi,
Has anybody managed to get this configuration work with PCI passthrough or
should I look to buying a separate server ?
Regards,
--
--[ UxBoD ]--
// PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import
// Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84
//
On Thu, Aug 21, 2008 at 08:36:44PM -0500, Dwayne Hubbard wrote:
I also want to reiterate that the libpri and Asterisk branches above
are development branches, so be careful in a production environment.
This functionality will be available in Asterisk 1.6.2. To disable a
channel via the CLI
Thanks for your answer and doing some test I have this SIP debug:
From Huawei SIDE we have:
12:53:41.358166 IP (tos 0xb8, ttl 127, id 0, offset 0, flags [none], proto:
UDP (17), length: 856) 189.8.113.170.5060 189.8.126.177.5060: SIP, length:
828
INVITE sip:[EMAIL
On Thu, Aug 21, 2008 at 3:11 PM, Andy Dixon [EMAIL PROTECTED] wrote:
Hello,
I am trying to alter the outbound callerID for extensions within a
context I have created.
I wrote the following:
exten = _9.,2,ExecIf($[$[${REALCALLERIDNUM} = 360] | $[$
{REALCALLERIDNUM} =
ronald [EMAIL PROTECTED] writes:
Is it possible to assign a plus sign on the callerid(num) ?
Yes.
currently this is what i do CALLERID(num)=+6523450017
but telco is denying calls, coz they said they are seeing bs523450017
instead of +6523450017.
Which techology? SIP? PRI? POTS? ...?
On Thursday 21 August 2008 08:26:47 am Olivier wrote:
Hi,
To check telco billing, I'm usinfg Asterisk-Stats from
http://www.areski.net/asterisk-stat-v2/about.php .
How can you tweak this application to display graphics and data that use
i started working from that software to come up that
Alex Balashov wrote:
Some carriers now do offer private SS7 instead of ISDN. But there is
absolutely no reason why you should be doing this with Asterisk.
Asterisk-SS7 is quite tenuous at best. Unless you have some specific
reason to be using it, don't.
Actually, SS7 support in Asterisk
On Fri, 22 Aug 2008 01:13:25 -0700, Paul Chambers wrote:
Another example of the North American frequency allocations being just a
little bit different from everywhere else in the world... So does that
mean you've stopped using your S685 IP, Michael? ;)
Yes, between the power problems that I
+ is not a valid Caller*ID character. Asterisk allows you to use + in
Caller*ID, but many carriers will reject the call if you do that.
Benny Amorsen wrote:
ronald [EMAIL PROTECTED] writes:
Is it possible to assign a plus sign on the callerid(num) ?
Yes.
currently this is what i do
Atis Lezdins wrote:
[clid-mangle]
exten = 70[01],1,Set(CALLERID(num)=581557)
exten = 70[01],2,Return()
exten = 10[01],1,Set(CALLERID(num)=581500)
exten = 10[01],2,Return()
; and so on, just better reorganize your extensions so that this can
match patterns better.
[dial-out]
exten =
On Friday 22 August 2008 07:54:43 am Anthony Messina wrote:
On Thursday 21 August 2008 08:26:47 am Olivier wrote:
Hi,
To check telco billing, I'm usinfg Asterisk-Stats from
http://www.areski.net/asterisk-stat-v2/about.php .
How can you tweak this application to display graphics and
On Fri, Aug 22, 2008 at 5:14 AM, bilal ghayyad [EMAIL PROTECTED] wrote:
Dear Darren;
You might be right because one day it happened with me and the situation was
same like this as following:
The status that the ping result is very good for all partied (Asterisk
machine, IP Phones on the
I think you could minimize the incidence of the problem by having a PRI
with like 100 numbers associated, with the CO doing the routing stripping
off the last two forwarded digits. You also have a premium service
provider that forwards premium calls to one of those numbers (I think from
On Fri, Aug 22, 2008 at 4:23 AM, Johansson Olle E [EMAIL PROTECTED] wrote:
Just some friendly advice if you really want a discussion. Of course,
I clicked, read and commented ;-)
If this is a way we can get you to say something, Olle, I'm for it! :)
This said, I think Michael was trying to
Requesting help.
Thaks
On Tue, Aug 19, 2008 at 4:40 PM, Jim Boykin [EMAIL PROTECTED] wrote:
We run asterisk to handle incoming DIDs and we have observed
inefficient Codec Translation.
Here is the scenario
[DID Vendor] --- [Asterisk ]
I have been told before on this list that a modem through a zaptel card
will not work. And mine doesn't, at least not for data calls (it works
fine for fax). Apparently the modem requires the full bandwidth of the
POTS line, which you do not get through the zaptel card.
You might at least check
On 22 Aug 2008, at 14:55, randulo wrote:
On Fri, Aug 22, 2008 at 4:23 AM, Johansson Olle E [EMAIL PROTECTED]
wrote:
Just some friendly advice if you really want a discussion. Of course,
I clicked, read and commented ;-)
If this is a way we can get you to say something, Olle, I'm for it!
It's tough to say why a voice would start sounding like a robot. There
are so many variables that could effect your Asterisk server.
I always go for process of elimination when I have a problem similar
to this with call quality.
What I would do is install an end point on the same local
Not sure what you've heard before, but I have successfully used a
modem at 9600 baud (forced via AT commands) through a zaptel card on
several occasions.
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug
On Fri, Aug 22, 2008 at 7:18 AM, Tim Panton [EMAIL PROTECTED] wrote:
I often read this list offline (during my commute) and articles which
reference a web page without at least summarizing the content
are frustrating :-)
Not to argue, but to add that for what you describe I use Google
Reader.
Then after many hours of chasing ghosts, you decide, hmmm, let me
eliminate IAX2 as a possible cause, and boom!, everything works and
voice quality is perfect.
Then you are happy you got it fixed but mad you you wasted so much
time an the highly touted IAX2 that should just work but doesn't in
On Fri, 22 Aug 2008 13:23:09 +0200, Johansson Olle E wrote:
21 aug 2008 kl. 16.47 skrev [EMAIL PROTECTED] [EMAIL PROTECTED]:
Yesterday I blogged a post about some ideas that I think will help
Asterisk appliances further penetrate SMB/SOHO sites in ways that are
not presently being addressed.
On Tue, Aug 19, 2008 at 7:10 AM, Jim Boykin [EMAIL PROTECTED] wrote:
We run asterisk to handle incoming DIDs and we have observed
inefficient Codec Translation.
Here is the scenario
[DID Vendor] --- [Asterisk ]
External GW [G729]
On Tue, Aug 19, 2008 at 7:10 AM, Jim Boykin [EMAIL PROTECTED] wrote:
We run asterisk to handle incoming DIDs and we have observed
inefficient Codec Translation.
Here is the scenario
[DID Vendor] --- [Asterisk ]
External GW [G729]
Kevin P. Fleming wrote:
Alex Balashov wrote:
Some carriers now do offer private SS7 instead of ISDN. But there is
absolutely no reason why you should be doing this with Asterisk.
Asterisk-SS7 is quite tenuous at best. Unless you have some specific
reason to be using it, don't.
Just read this on Alec Saunders fantastic blog
http://saunderslog.com/2008/08/21/diamondware-acquired/
Interesting concept, makes me wonder if it is possible in Asterisk to
'adjust' the left/right mix for audio conference participants?
Yeh I know there is only one channel in a telephone call
We will be gathering at 9AM PDT, 12 Noon EDT, 4PM GMT (i think?) for
our weekly conference of asterisk and VoIP users. Any and all
discussion related to telephony is welcome. Please join us any Friday.
More info using the links below.
PSTN (724) 444-7444 and enter 22622# 1#
SIP [EMAIL PROTECTED]
On Fri, Aug 22, 2008 at 7:55 AM, Dean Collins [EMAIL PROTECTED] wrote:
It's probably covered under patents etc but has anyone tried to 'spatially
separate' audio mixes for each participant in an Asterisk conference call?
I've never tried it, but as soon as I heard about the idea a few
months
Hello All,
I have an Asterisk Box currently running 1.6.0, dahdi drivers and libss7
with a TE120P (01 E1) card for the past 02 months.
I have it connected to a Cellular Operator switch (MSC), and it is working
perfectly. Traffic is still quite low, but increasing as we start to use it
for new
This is a poor example but basically if you have 'stereo' I was thinking
an equation like this.
Number of speakers in a conference room = 'N' deviations
Range = 80%
(obviously you couldn't do 100% otherwise would be silent in outer
channels once you get over 10 participants.
(50/50 + / -
On 08/20/08 14:46, Paul Hales wrote:
Joseph wrote:
Does anybody know if the process of upgrading firmware on Linksys
SPA3102-NA in Linux is the same as on Sipura 3K as described on
voip-info.org
http://www.voip-info.org/wiki/view/Sipura
I'm pretty sure it works - I used it to upgrade a
- Jay R. Ashworth [EMAIL PROTECTED] wrote:
1) can you do it gracefully (both that and immediate are sometimes
useful)?
Right now you can only disable an idle channel.
2) can you take down either an entire span, or a channel range on the
command line?
This functionality will be added
Hello,
I want to ask, how to detect queue timeout? If queue members are busy or
not answering to the call, and after queue timeout caller would hear :
Sorry all operators are busy, please leave a record:
This example:
[ivr]
exten = start,1,Ringing
exten = start,n,Wait(2)
exten =
Are there parameters for em wink?
1) timing parameters
2) dial delay or pre dial.
Thanks
Jerry
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now:
Giedrius Augys wrote:
Hello,
I want to ask, how to detect queue timeout? If queue members are
busy or not answering to the call, and after queue timeout caller would
hear : Sorry all operators are busy, please leave a record:
This example:
[ivr]
exten = start,1,Ringing
exten =
On Fri, 2008-08-22 at 14:40 -0400, Jerry Geis wrote:
Are there parameters for em wink?
A quick glance at the sample zapata.conf that comes with Asterisk shows
prewink, wink, and rxwink timing parameters.
--
Jared Smith
Training Manager
Digium, Inc.
On Fri, 2008-08-22 at 21:26 +0300, Giedrius Augys wrote:
I want to ask, how to detect queue timeout? If queue members are
busy or not answering to the call, and after queue timeout caller
would hear : Sorry all operators are busy, please leave a record:
The Queue() application sets a channel
On Aug 22, 2008, at 2:26 PM, Giedrius Augys wrote:
Hello,
I want to ask, how to detect queue timeout? If queue members are
busy or not answering to the call, and after queue timeout caller
would hear : Sorry all operators are busy, please leave a record:
This example:
[ivr]
exten =
larry schrieb:
Here is a question about the fxs of the zaptel card which is set a
Didn't you post almost the exact same question yesterday?
(twice already)
Grüße,
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566
Andy Dixon schrieb:
whats AEL?
Asterisk Extension Language. extensions.ael
Grüße,
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister:
Steve Totaro wrote:
On Tue, Aug 19, 2008 at 7:10 AM, Jim Boykin [EMAIL PROTECTED] wrote:
We run asterisk to handle incoming DIDs and we have observed
inefficient Codec Translation.
Here is the scenario
[DID Vendor] --- [Asterisk ]
External
Sorry to have posted to the wrong maillist. Repost here.
Regards.
--- On Fri, 8/22/08, Ming-Ching Tiew [EMAIL PROTECTED] wrote:
From: Ming-Ching Tiew [EMAIL PROTECTED]
Subject: [asterisk-dev] frequent channel reset problem
To: [EMAIL PROTECTED]
Date: Friday, August 22, 2008, 3:04 PM
Hi,
From: Ming-Ching Tiew [EMAIL PROTECTED]
Subject: [asterisk-dev] frequent channel reset problem
To: [EMAIL PROTECTED]
Date: Friday, August 22, 2008, 3:04 PM
Hi,
I am stucked with a nasty PRI problem for 2 weeks now and
will appreciate if I could get some help from here. The
problem is that
Sorry to have posted reply to your email directly.
Repost to maillist.
resetinterval already set to never to start with.
Thanks
--- On Fri, 8/22/08, Ming-Ching Tiew [EMAIL PROTECTED] wrote:
From: Ming-Ching Tiew [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Fw: [asterisk-dev] frequent
I expected to find th module ztd-ethmf[.c...] in support of the redfone
TDMoE product in my zaptel distro (I have 1.4.11). But it's not there.
I am awaiting a response to a trouble ticket from redfone. Can anyone
give me a jumpstart? I can't seem to google this up.
smime.p7s
Description:
On Thursday 21 August 2008 10:08:53 J.M. wrote:
I am running Asterisk 1.4.21.2 with Realtime. I have a phone setup in the
database and when I connect that phone to Asterisk there are suddenly an
endless number of SELECT * FROM sip WHERE name = '1001' AND host =
'dynamic' queries being run.
Hi Thanks for all your reply.
Just figured out that ISUP does not decode plus sign very well.
regards
nhadie
Eric ManxPower Wieling wrote:
+ is not a valid Caller*ID character. Asterisk allows you to use + in
Caller*ID, but many carriers will reject the call if you do that.
Benny
Darren Sessions wrote:
Not sure what you've heard before, but I have successfully used a
modem at 9600 baud
Well, OK, it won't work was a little strong. Faxes work because they
too are at slow speed. But for me at least, 9600 baud is pretty much
useless. Instead, I just patch the modem
--- On Fri, 8/22/08, Ming-Ching Tiew [EMAIL PROTECTED] wrote:
From: Ming-Ching Tiew [EMAIL PROTECTED]
Subject: Re: [asterisk-users] frequent channel reset problem
To: asterisk-users@lists.digium.com
Date: Friday, August 22, 2008, 11:00 PM
Sorry to have posted reply to your email directly.
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