I would be curious to know where, in this classification, fall various
telemarketing schemes that are technically not cold-calls, but are
generated from leads that come from customer-provided information, but
where the customer does not know explicitly that they are signing up to
receive
Hi,
I am requesting for a E1 connection from my telco. They are asking if I
want DSS1 or SS7, and I am stuck here. Could someone tell me the difference
between the two? How should I decide which one to use?
Thanks in advance for your help.
Mark
___
Use DSS1. It's European ISDN and would give you the equivalent of a
North American PRI.
You don't want SS7.
mark morreny wrote:
Hi,
I am requesting for a E1 connection from my telco. They are asking if I
want DSS1 or SS7, and I am stuck here. Could someone tell me the
difference
Hi,
There is a question about the fxo of the zaptel card which is set a
number to use as common analog phone. When I use ${CALLERID(num)}to get it's
number, it could'n be done. But ${CALLERID(num)} could get the other number
of the SIP or IAX . Could you tell me the reason, and how I could
On 20 Aug 2008, at 18:00, Eric Chamberlain wrote:
We are exploring using Asterisk for a project and we are looking for a
way to encrypt/decrypt the peer passwords stored in the realtime
database (postrges).
Ideally, we want to use a public key to encrypt the passwords before
they go into
Hi,
I have recently been having difficulty with cmd record where calls
are not being recorded. I would like to know whether it is possible
that my fastagi script is the root cause of the problem.
I am using a fastagi script written in python to answer the calls,
and the dialogue interaction
Hi,
thanks a lot for your answer!
If you just need Astrerisk for building Zaptel, you don't need the
kernel modules installed.
I don't need Asterisk for buildung zaptel, I need zaptel running to be
able to compile Asterisk WITH meetme-module (and some others) to build a
RPM that can be
Hi!
I'm setting up my IVR system, how can I register in a mysql database the
IVR menus accessed by the clients ?
Thanks a lot,
Szasz Szabolcs
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Thanks
Shaun ___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
You could use func_odbc in your dialplan, check here :
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+func_odbc
Yves.
On Thu, 2008-08-21 at 14:57 +0300, Szasz Szabolcs wrote:
Hi!
I'm setting up my IVR system, how can I register in a mysql database the
IVR menus accessed by
I'm setting up my IVR system, how can I register in a mysql database the
IVR menus accessed by the clients ?
Just use the MYSQL-Functions in the dialplan to write the menues name
(and datetime maybe) in a table.
To access MYSQL from the dialplan you need to have the asterisk-addons.
Sorry, maybe I misunderstood your question.
If you want the dialplan to be in a MySQL dabtase, check here :
http://www.voip-info.org/wiki/view/Asterisk+configuration+from+database
Works great, but the documentation is sometimes a bit outdated.
Good luck.
Yves.
On Thu, 2008-08-21 at 14:57
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} = 670]]|Set|CALLERID(num)=581560)
exten = _9.,3,ExecIf($[$[${REALCALLERIDNUM} = 361] | $[$
Chris Hastie wrote:
Is it possible to have two peers register to Asterisk from the same
IP/port combination?
I have a Zoom 5821 two port ATA that can support up to 4 VOIP accounts.
I want to use it to provide two different extensions on an Asterisk
system. In the past I have configured two
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
Billsec instead of Duration ?
Regards
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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} = 670]]|Set|CALLERID(num)=581560)
exten =
Shaun Wingrin wrote:
Thanks
There will be a (T) after the iax entry:
asterisk.cw 192.168.200.2 (D) 255.255.255.255 4569 (T) OK
(76 ms)
asterisk.liv 192.168.102.15 (D) 255.255.255.255 4569 (T) OK
(77 ms)
asterisk.bc 192.168.104.10 (D) 255.255.255.255 4569 (T)
I am using centos 4.6 i586.
I have compiles zaptel 1.4.11 ztdummy.
When I load ztdummy the /proc/interupts rtc does not increment.
centos runs 2.6.9 kernel.
I'm not sure ztdummy.c uses RTC by default in this case.
Anyone using centos 4.X successfully with console/dsp and not internal
cards.
I have LinkSYS PAP2t and it worked the way you discribed it.. Asterisk simply
assigns a different port for the peer automaticaly.
Date: Wed, 20 Aug 2008 20:09:32 +0100 From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Subject: [asterisk-users] Two peers, same IP
and port Is it
Hello
I have a SendTEK UNIK-22 USB ISDN TA unit attached to my Asterisk
i it possible to use it to make and receive calls with asterisk? and if so can
anyone help me? or at least give me some hints? i tried but couldn't manage it
_
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.
http://blog.mgraves.org/2008/08/20/a-suggestion-to-asterisk-appliance-developers/
Michael Graves
mgraves at mstvp.com
o(713)
Please google VoIP2.0 apps... this is old old news... even Cisco has
marketed this going back to 2001.
-E
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Hi,
Hoping someone can help with this most frustrating situation.
I have a Linksys PAP2T registering with ADSL to my asterisk server which also
sits behind a Mikrotik router.
Thanks
Shaun ___
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On Thu, Aug 21, 2008 at 02:38:49PM +, Tariq .. wrote:
Hello
I have a SendTEK UNIK-22 USB ISDN TA unit attached to my Asterisk
i it possible to use it to make and receive calls with asterisk? and if so
can anyone help me? or at least give me some hints? i tried but couldn't
manage it
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. The only way to stop the flood of queries
coming from
On Thu, Aug 21, 2008 at 10:29:18AM -0400, Jerry Geis wrote:
I am using centos 4.6 i586.
I have compiles zaptel 1.4.11 ztdummy.
When I load ztdummy the /proc/interupts rtc does not increment.
does ztdummy itself tick?
try zttest
If it does not stay hung there, it's working.
--
On Thu, Aug 21, 2008 at 02:15:58AM -0400, Alex Balashov wrote:
I would be curious to know where, in this classification, fall various
telemarketing schemes that are technically not cold-calls, but are
generated from leads that come from customer-provided information, but
where the customer
On Thu, Aug 21, 2008 at 02:38:19AM -0400, Alex Balashov wrote:
Use DSS1. It's European ISDN and would give you the equivalent of a
North American PRI.
You don't want SS7.
I would assume that means SS7 protocol over a link not routed directly
to the SS7 backbone.
At least I hope it means
On Thu, Aug 21, 2008 at 10:29:18AM -0400, Jerry Geis wrote:
/ I am using centos 4.6 i586.
//
// I have compiles zaptel 1.4.11 ztdummy.
// When I load ztdummy the /proc/interupts rtc does not increment.
/
does ztdummy itself tick?
try zttest
If it does not stay hung there, it's working.
Hi all,
asterisk is giving me tough time. its been 3 days I am trying to originate
outgoing call using manager api/callfiles. both seem to work fine when i
originate a call for a local peer, but if i try originating a call outside
using a trunk thats when everything goes wrong. It does originate
On Thu, Aug 21, 2008 at 11:19:22AM -0400, Jerry Geis wrote:
On Thu, Aug 21, 2008 at 10:29:18AM -0400, Jerry Geis wrote:
/ I am using centos 4.6 i586.
//
// I have compiles zaptel 1.4.11 ztdummy.
// When I load ztdummy the /proc/interupts rtc does not increment.
/
does ztdummy itself
I Googled as you suggest and nothing even vaugely related is returned.
In fact, VOIP 2.0 as a term doesn't seem to relate.
What I'm suggesting is that smaller PBX systems should embrace a larger
role in the end users operation.
I don't see CCM is small companies or home offices. This is all
Hi,
I noticed that when dial terminates it does not return to the dialplan,
and therefore can not execute any entry after Dial().
Is there any trick to overcome this limitation ?
How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if
I can not execute
Hi,
I would like to arrange that when an IAX client logs in / registers with
my * AND there are unread voicemails, this IAX client will be
automatically called and connected to the respective voicemail box.
One possibility is to have a cronjob that creates a callfile - let's say
- every five
We recently discussed DeadAGI on the list - I'd check the archives
first.
I just finished doing a write up on DeadAGI and Perl on my website if
you're interested.
DeadAGI *can* be very reliable if done properly.
- Darren
_
[EMAIL PROTECTED]
Jay R. Ashworth wrote:
On Thu, Aug 21, 2008 at 02:15:58AM -0400, Alex Balashov wrote:
I would be curious to know where, in this classification, fall various
telemarketing schemes that are technically not cold-calls, but are
generated from leads that come from customer-provided
Rizwan Hisham wrote:
Hi all,
asterisk is giving me tough time. its been 3 days I am trying to
originate outgoing call using manager api/callfiles.
I would say remove the @TRUNK-OUT part and make sure that the context
you send the call to knows about sending calls to the outside world.
--
Hi Stefan,
I'd expect there's a Manager event that is fired when an IAX client
login happens. You could watch for that and initiate your call if
there's voicemail at that time.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352)
Hi folks! I have a problem with our Sip provider that have a Softswitch
Huawei SoftX3000 to send us SIP calls to our Asterisk PBX, we are working
with G711 with them. They start sending calls to our pbx, some time after
they start to receive 408 messages from asterisk and some time after this
they
First, if you want to use that, you may want pass the call tracknum to
the myagi.agi,
so you will know which call the dialedtime and answeredtime belongs to.
But you can use the Dial 'g' option that doesn't hangup up both legs of
the call when the called party hangs up.
selmak se wrote:
Thank you for your answer,
Is the call tracknum stored in some variable?
Could you let me know how to pass a call tracknum to an AGI.
Se.-
- Original Message -
From: Ruddy Gbaguidi
First, if you want to use that, you may want pass the call tracknum to
the
On Thu, Aug 21, 2008 at 09:44:50AM -0600, Anthony Francis wrote:
Jay R. Ashworth wrote:
On Thu, Aug 21, 2008 at 02:15:58AM -0400, Alex Balashov wrote:
I would be curious to know where, in this classification, fall various
telemarketing schemes that are technically not cold-calls, but are
Jay R. Ashworth wrote:
On Thu, Aug 21, 2008 at 02:38:19AM -0400, Alex Balashov wrote:
Use DSS1. It's European ISDN and would give you the equivalent of a
North American PRI.
You don't want SS7.
I would assume that means SS7 protocol over a link not routed directly
to the SS7 backbone.
Anthony Francis wrote:
Actually in the US all you have to do is provide some proof of a
business relationship with them. Companes get away with calling you if
you have ever bought even one item from them.
So, what if you never bought anything, but ended up as a lead in their
system through
Gustavo A Gonzalez wrote:
Hi folks! I have a problem with our Sip provider that have a Softswitch
Huawei SoftX3000 to send us SIP calls to our Asterisk PBX, we are
working with G711 with them. They start sending calls to our pbx, some
time after they start to receive 408 messages from
On Thu, Aug 21, 2008 at 12:21:36PM -0400, Alex Balashov wrote:
I would assume that means SS7 protocol over a link not routed directly
to the SS7 backbone.
At least I hope it means that. shudder
Indeed, it is certainly private SS7. :-) But that does not mean it is
any more
In article [EMAIL PROTECTED], selmak se [EMAIL PROTECTED] wrote:
I noticed that when dial terminates it does not return to the dialplan,
and therefore can not execute any entry after Dial().
Is there any trick to overcome this limitation ?
You can give the 'g' option to Dial, but that
Not to sound arrogant but the law is one thing and enforcement is another,
these types of calls have been illegal for a long time and 9 times out of 10
the only penalty one receives is a civil suit by some back yard attorney
looking for a couple thousand bucks.
Unless that is you are a serious
To those running call centers I have a question: what kinds of soft
phones, if any, do you use? I'm wondering what is out there that has
some hooks for custom applications or host system integration, etc.
OTOH, do you prefer a desk phone for any reason? If so, why?
Thanks for your thoughts,
Hello all!
my last month's phone bill sky rocketed after i setup asterisk with softphones
all over the house!
could someone help me set up a limitation for my wife and kids not to be able
to talk for more than 5 min at a time!
or like 20 min per week! or whtever limitation i could set for
For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP
range in the U.S. I'm particularly interested in the Gigaset S685 IP.
Since it's DECT 6.0, and there's an English (UK) version, I'm thinking
it should work just fine, after dealing with the walwart issue (and
maybe caller
hi all,
has anyone able to configure ultramonkey for sip (namely asterisk).
i tried from this tutorial:
http://blog.iclutton.com/2008/01/load-balancing-and-high-availablity.html
i have this on my ldirectord.cf:
virtual=123.45.67.155:5060
real=123.45.67.130:5060 gate
Paul Chambers wrote:
For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP
range in the U.S. I'm particularly interested in the Gigaset S685 IP.
Since it's DECT 6.0, and there's an English (UK) version, I'm thinking
it should work just fine, after dealing with the walwart
RoLaNd RoLaNd wrote:
Hello all!
my last month's phone bill sky rocketed after i setup asterisk with
softphones all over the house!
could someone help me set up a limitation for my wife and kids not to
be able to talk for more than 5 min at a time!
or like 20 min per week! or whtever
RoLaNd RoLaNd schrieb:
Hello all!
my last month's phone bill sky rocketed after i setup asterisk with
softphones all over the house!
could someone help me set up a limitation for my wife and kids not to be
able to talk for more than 5 min at a time!
or like 20 min per week! or whtever
On Thu, Aug 21, 2008 at 12:50 PM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
Hello all!
my last month's phone bill sky rocketed after i setup asterisk with
softphones all over the house!
could someone help me set up a limitation for my wife and kids not to be
able to talk for more than 5 min at
On Thu, Aug 21, 2008 at 09:40:04AM -0700, Michael Collins wrote:
To those running call centers I have a question: what kinds of soft phones,
if any, do you use? Iâm wondering what is out there that has some hooks
for
custom applications or host system integration, etc. OTOH, do
That's a good point. I don't know, honestly, if you can react to a SIP
register or an IAX login from within the dialplan. To anyone else:
Is there a way to act in the dialplan on a manager event?
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
Steve Totaro wrote:
On Thu, Aug 21, 2008 at 12:50 PM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
Hello all!
my last month's phone bill sky rocketed after i setup asterisk with
softphones all over the house!
could someone help me set up a limitation for my wife and kids not to be
able to talk
On 21 Aug 2008, at 18:44, Jay R. Ashworth wrote:
On Thu, Aug 21, 2008 at 09:40:04AM -0700, Michael Collins wrote:
To those running call centers I have a question: what kinds of
soft phones,
if any, do you use? I’m wondering what is out there that has
some hooks for
custom
This has got to be one of the funniest threads ever to grace this forum.
Sorry honey! ...CLICK.
In my house, this might require a more 'diplomatic' approach :-)
-Karl
On Thu, 21 Aug 2008 21:41:40 +0300, RoLaNd RoLaNd
[EMAIL PROTECTED] said:
i tried that before.. it didnt actually work! it
Phone Guy: NO PHONE FOR YOU!
Karl Fife wrote:
This has got to be one of the funniest threads ever to grace this forum.
Sorry honey! ...CLICK.
In my house, this might require a more 'diplomatic' approach :-)
-Karl
On Thu, 21 Aug 2008 21:41:40 +0300, RoLaNd RoLaNd
[EMAIL PROTECTED] said:
You're not kidding. I would have to investigate cheaper routing.
Truncating my wife's calls would be met with harsh reaction at best.
Maybe painful, too.
Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
FWD 54245
Original Message
Heck, I was going to say I probably be on the sofa that night and the
next...
[EMAIL PROTECTED] wrote:
You're not kidding. I would have to investigate cheaper routing.
Truncating my wife's calls would be met with harsh reaction at best.
Maybe painful, too.
Michael Graves
mgraves at mstvp.com
I'd run top on the server to see if the CPU utilization is going
through the roof. If you use AGI, make sure there aren't any orphaned
processes consuming resources.
If all else fails on the software side of things, I'd restart the
server.
_
Darren Sessions
Saul Bejarano wrote:
With SS7 They will have to define point code and stuff like that, it is
usually granted to carriers which are members of the SS7 network, I have
not seen a carrier offering SS7 as a home service.
Go for standard DSS1 which as somebody said will be the equivalent in
Thanks a lot for your kindly help and advise.
1) I did restart for the machine and it stayed the same.
2) If I call to the Asterisk via the PSTN, and the IVR answer and then I enter
the extension of the IP Phone which is in another country, the voice is nice
and no problem, but If I call from
Dear Darren;
I discovered that calling from the Asterisk to the IP Phone Extension (like
calling from mobile to digium and then enter the IP Phone extension, or calling
from fxs to the IP Phone extension), it goes very good without any problem.
But calling from the same IP Phone to another IP
Dear All;
I start beleive that if I did asterisk compilation again, then the problem
might be removed as I start beleive it is related to corrupt happened in some
files.
The questions here are:
1) Which could these files?
2) How can I know the reason for the corrupt?
3) I have another
Hi,
I have several asterisk pstn gateways running, each with at least 2 e1
pri circuits connected. I wonder if there's a way to block incoming
calls on the pri's, in such way that my telco sends the call to one of
the other pri's (all the pri's are together in a 'hunt group', calls get
evenly
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
- Egbert [EMAIL PROTECTED] wrote:
I have several asterisk pstn gateways running, each with at least 2 e1
pri circuits connected. I wonder if there's a way to block incoming
calls on the pri's,
I have been working on this functionality and have development branches that
are ready for
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
Ring and Call Waiting Tone Spec
Hello all,
I have a system at a motel that is mostly analog phones with 2 32 port
astribanks.
I am having problems getting a modem data call to connect.
There are many travelling salesmen that require this functionality to work
to dial direct into their company systems.
I am using Asterisk
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